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[asterisk-bristuff.git] / ChangeLog

2008-09-09  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.22-rc5 released.

2008-09-09 15:40 +0000 [r142063]  Russell Bryant <russell@digium.com>

        * res/res_features.c: Ensure that the stored CDR reference is still
          valid after the bridge before poking at it. Also, keep the
          channel locked while messing with this CDR. (fixes crashes
          reported in issue #13409)

2008-09-08 21:10 +0000 [r141809]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: Fix pedantic mode of chan_sip to only check
          the remote tag of an endpoint once a dialog has been confirmed.
          Up until that point, it is possible and legal for the far-end to
          send provisional responses with a different To: tag each time.
          With this patch applied, these provisional messages will not
          cause a matching problem. (closes issue #11536) Reported by: ibc
          Patches: 11536v2.patch uploaded by putnopvut (license 60)

2008-09-08 21:02 +0000 [r141806]  Russell Bryant <russell@digium.com>

        * main/pbx.c: When doing an async goto, detect if the channel is
          already in the middle of a masquerade. This can happen when
          chan_local is trying to optimize itself out. If this happens,
          fail the async goto instead of bursting into flames. (closes
          issue #13435) Reported by: geoff2010

2008-09-08 20:15 +0000 [r141741]  Jason Parker <jparker@digium.com>

        * Makefile, redhat (removed): Remove RPM package targets from
          Makefile (and all associated parts). This has never worked in
          1.4, and we decided that it makes no sense to be done here. There
          are many distros out there that already have "proper" spec files
          that can be (re)used. Closes issue #13113 Closes issue #10950
          Closes issue #10952

2008-09-08  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.22-rc4 released.

2008-09-08 20:15 +0000 [r141741]  Jason Parker <jparker@digium.com>

        * Makefile, redhat (removed): Remove RPM package targets from
          Makefile (and all associated parts). This has never worked in
          1.4, and we decided that it makes no sense to be done here. There
          are many distros out there that already have "proper" spec files
          that can be (re)used. Closes issue #13113 Closes issue #10950
          Closes issue #10952

2008-09-08 16:26 +0000 [r141678]  Russell Bryant <russell@digium.com>

        * configure, configure.ac: Actually use Zaptel CFLAGS if using
          Zaptel instead of DAHDI This fixes building against Zaptel when
          using a custom path

2008-09-06 20:13 +0000 [r141565]  Steve Murphy <murf@digium.com>

        * channels/chan_sip.c: This fix comes from Joshua Colp The
          Brilliant, who, given the trace, came up with a solution. This
          will most likely will close 13235 and 13409. I'll wait till
          Monday to verify, and then close these bugs.

2008-09-06 15:23 +0000 [r141503]  Tilghman Lesher <tlesher@digium.com>

        * res/res_agi.c: Reverting behavior change (AGI should not exit
          non-zero on SUCCESS) (closes issue #13434) Reported by:
          francesco_r

2008-09-05 21:10 +0000 [r141217-141366]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_agent.c: Agent's should not try to call a channel's
          indicate callback if the channel has been hung up. It will likely
          crash otherwise ABE-1159

        * apps/app_voicemail.c: Since greetings are not stored in IMAP, we
          should not be DISPOSE'ing of them the same way we do with other
          messages. (closes issue #13414) Reported by: mthomasslo Patches:
          13414v2.patch uploaded by putnopvut (license 60) Tested by:
          mthomasslo

        * channels/chan_sip.c: Commit 140417 had a logic flaw in it which
          caused port 5060 to always be used when dialing a peer if no
          explicit port was specified. This broke the behavior of
          implicitly using the port from which the peer registered if no
          port is specified. This commit fixes the logic flaw. (closes
          issue #13424) Reported by: mdu113 Patches: 13424.patch uploaded
          by putnopvut (license 60) Tested by: mdu113

2008-09-05 14:15 +0000 [r141094-141156]  Steve Murphy <murf@digium.com>

        * main/channel.c: A small change to prevent double-posting of
          CDR's; thanks to Daniel Ferrer for bringing it to our attention

        * pbx/ael/ael-test/ref.ael-vtest25 (added),
          pbx/ael/ael-test/ael-vtest25/extensions.ael (added),
          pbx/ael/ael-test/ael-vtest25 (added), pbx/ael/ael_lex.c,
          pbx/ael/ael-test/ref.ael-test6, pbx/ael/ael.flex: (closes issue
          #13357) Reported by: pj Tested by: murf (closes issue #13416)
          Reported by: yarns Tested by: murf If you find this message
          overly verbose, relax, it's probably not meant for you. This
          message is meant for probably only two people in the whole world:
          me, or the poor schnook that has to maintain this code because
          I'm either dead or unavailable at the moment. This fix solves two
          reports, both having to do with embedding a function call in a
          ${} construct. It was tricky because the funccall syntax has
          parenthesis () in it. And up till now, the 'word' token in the
          flex stuff didn't allow that, because it would tend to steal the
          LP and RP tokens. To be truthful, the "word" token was the
          trickiest, most unstable thing in the whole lexer. I was lucky it
          made this long without complaints. I had to choose every
          character in the pattern with extreme care, and I knew that
          someday I'd have to revisit it. Well, the day has come. So, my
          brilliant idea (and I'm being modest), was to use the surrounding
          ${} construct to make a state machine and capture everything in
          it, no matter what it contains. But, I have to now treat the word
          token like I did with comments, in that I turn the whole thing
          into a state-machine sort of spec, with new contexts
          "curlystate", "wordstate", and "brackstate". Wait a minute,
          "brackstate"? Yes, well, it didn't take very many regression
          tests to point out if I do this for ${} constructs, I also have
          to do it with the $[] constructs, too. I had to create a separate
          pcbstack2 and pcbstack3 because these constructs can occur inside
          macro argument lists, and when we have two state machines
          operating on the same structures we'd get problems otherwise. I
          guess I could have stopped at pcbstack2 and had the brackstate
          stuff share it, but it doesn't hurt to be safe. So, the pcbpush
          and pcbpop routines also now have versions for "2" and "3". I had
          to add the {KEYWORD} construct to the initial pattern for "word",
          because previously word would match stuff like "default7",
          because it was a longer match than the keyword "default". But,
          not any more, because the word pattern only matches only one or
          two characters now, and it will always lose. So, I made it the
          winner again by making an optional match on any of the keywords
          before it's normal pattern. I added another regression test to
          make sure we don't lose this in future edits, and had to fix just
          one regression, where it no longer reports a 'cascaded' error,
          which I guess is a plus. I've given some thought as to whether to
          apply these fixes to 1.4 and the 1.6.x releases, vs trunk; I
          decided to put it in 1.4 because one of the bug reports was
          against 1.4; and it is unexpected that AEL cannot handle this
          situation. It actually reduced the amount of useless "cascade"
          error messages that appeared in the regressions (by one line,
          ehhem). There is a possible side-effect in that it does now do
          more careful checking of what's in those ${} constructs, as far
          as matching parens, and brackets are concerned. Some users may
          find a an insidious problem and correct it this way. This should
          be exceedingly rare, I hope.

2008-09-04 17:00 +0000 [r141028]  Jeff Peeler <jpeeler@digium.com>

        * res/res_features.c, res/res_agi.c: (closes issue #11979) Fixes
          multiple parking problems: Crash when executing a park on an
          extension dialed by AGI due to not returning the proper return
          code. Crash when using a builtin feature that was a subset of a
          enabled dynamic feature. Crash due to always hanging up the peer
          despite the fact that the peer was supposed to be parked.

2008-09-03  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.22-rc3 released.

2008-09-03 14:29 +0000 [r140850]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Fix voicemail forwarding when using ODBC
          storage. (closes issue #13387) Reported by: moliveras Patches:
          13387.patch uploaded by putnopvut (license 60) Tested by:
          putnopvut, moliveras

2008-09-03 13:24 +0000 [r140816]  Russell Bryant <russell@digium.com>

        * main/poll.c: Don't freak out if the poll emulation receives NULL
          for the pollfds array (closes issue #13307) Reported by: jcovert

2008-09-02 23:47 +0000 [r140751]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: After adding the context checking to
          app_voicemail for IMAP storage, I left out a crucial place to
          copy the context to the vm_state structure. This is the
          correction.

2008-09-02 23:36 +0000 [r140670-140747]  Steve Murphy <murf@digium.com>

        * main/cdr.c: I am turning the warnings generated in ast_cdr_free
          and post_cdr into verbose level 2 messages. Really, they matter
          little to end users. You either get the CDR's you wanted, or you
          don't, and it is a bug.

        * main/channel.c: After reconsidering, with respect to 13409,
          ast_cdr_detach should be OK, better in fact, than ast_cdr_free,
          which generates lots of useless warnings that will undoubtably
          generate complaints.

        * main/channel.c, main/pbx.c: (closes issue #13409) Reported by:
          tomaso Patches: asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by
          tomaso (license 564) I basically spent the day, verifying that
          this patch solves the problem, and doesn't hurt in non-problem
          cases. Why valgrind did not plainly reveal this leak absolutely
          mystifies and stuns me. Many, many thanks to tomaso for finding
          and providing the fix.

2008-09-02 18:14 +0000 [r140605]  Sean Bright <sean.bright@gmail.com>

        * channels/chan_iax2.c: Make sure to use the correct length of the
          mohinterpret and mohsuggest buffers when copying configuration
          values. (closes issue #13336) Reported by:
          decryptus_proformatique Patches:
          chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded
          by decryptus (license 555)

2008-08-29 17:34 +0000 [r140417-140488]  Mark Michelson <mmichelson@digium.com>

        * main/manager.c, apps/app_queue.c, channels/chan_iax2.c: After
          working on the ao2_containers branch, I noticed something a bit
          strange. In all cases where we provide a callback function to
          ao2_container_alloc, the callback function would only return 0 or
          CMP_MATCH. After inspecting the ao2_callback() code carefully, I
          found that if you're only looking for one specific item, then you
          should return CMP_MATCH | CMP_STOP. Otherwise, astobj2 will
          continue traversing the current bucket until the end searching
          for more matches. In cases like chan_iax2 where in 1.4, all the
          peers are shoved into a single bucket, this makes for potentially
          terrible performance since the entire bucket will be traversed
          even if the peer is one of the first ones come across in the
          bucket. All the changes I have made were for cases where the
          callback function defined was passed to ao2_container_alloc so
          that calls to ao2_find could find a unique instance of whatever
          object was being stored in the container.

        * apps/app_voicemail.c: Add context checking when retrieving a
          vm_state. This was causing a problem for people who had
          identically named mailboxes in separate voicemail contexts. This
          commit affects IMAP storage only. (closes issue #13194) Reported
          by: moliveras Patches: 13194.patch uploaded by putnopvut (license
          60) Tested by: putnopvut, moliveras

        * channels/chan_sip.c: Fix SIP's parsing so that if a port is
          specified in a string to Dial(), it is not ignored. (closes issue
          #13355) Reported by: acunningham Patches: 13355v2.patch uploaded
          by putnopvut (license 60) Tested by: acunningham

2008-08-27 19:49 +0000 [r140299]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: Fix tag checking in get_sip_pvt_byid_locked
          when in pedantic mode. The problem was that the wrong tags would
          be compared depending on the direction of the call. (closes issue
          #13353) Reported by: flefoll Patches:
          chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll
          (license 244)

2008-08-26 16:49 +0000 [r140115]  Jeff Peeler <jpeeler@digium.com>

        * channels/chan_dahdi.c: add HAVE_PRI if define around
          dahdi_close_pri_fd

2008-08-26 16:07 +0000 [r140060]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: Fix some bogus scheduler usage in chan_sip.
          This code used the return value of a completely unrelated
          function to determine whether the scheduler should be run or not.
          This would have caused the scheduler to not run in cases where it
          should have. Also, leave a note about another scheduler issue
          that needs to be addressed at some point.

2008-08-26 15:57 +0000 [r140056]  Jeff Peeler <jpeeler@digium.com>

        * channels/chan_dahdi.c: (closes issue #12071) Reported by: tzafrir
          Patches: dahdi_close.diff uploaded by tzafrir (license 46) Tested
          by: tzafrir, jpeeler This patch fixes closing open file
          descriptors in the case of an error.

2008-08-26 15:27 +0000 [r140051]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix a race condition with the IAX scheduler
          thread. A lock and condition are used here to allow newly
          scheduled tasks to wake up the scheduler just in case the new
          task needs to run sooner than the current wakeup time when the
          thread is sleeping. However, there was a race condition such that
          a newly scheduled task would not properly wake up the scheduler
          or affect the wake up period. The order of execution would have
          been: 1) Scheduler thread determines wake up time of N ms. 2)
          Another thread schedules a task and signals the condition, with
          an execution time of < N ms. 3) Scheduler thread locks and goes
          to sleep for N ms. By moving the sleep time determination to
          inside the critical section, this possibility is avoided.

2008-08-26 15:22 +0000 [r140050]  Terry Wilson <twilson@digium.com>

        * Makefile: sounds/Makefile installs sounds using the "new"
          language directory structure, but languageprefix needs to be set
          = yes for sounds in subdirectories (digits/1, etc.) to play as
          the correct language. Fix the generation of asterisk.conf to
          include languageprefix=yes

2008-08-26 14:09 +0000 [r140029]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_dahdi.c: correct a file location in an error
          message

2008-08-25 21:47 +0000 [r139927]  Jeff Peeler <jpeeler@digium.com>

        * main/manager.c: Fix a typo I made. Lesson learned, apply the
          patch if one exists.

2008-08-25 21:31 +0000 [r139909]  Sean Bright <sean.bright@gmail.com>

        * build_tools/get_moduleinfo, build_tools/get_makeopts: Some
          versions of awk (nawk, for example) don't like empty regular
          expressions so be slightly more verbose. (closes issue #13374)
          Reported by: dougm Patches: 13374.diff uploaded by seanbright
          (license 71) Tested by: dougm

2008-08-25 20:46 +0000 [r139869]  Terry Wilson <twilson@digium.com>

        * channels/chan_sip.c: Make SIPADDHEADER() propagate indefinitely

2008-08-25 15:52 +0000 [r139769]  Mark Michelson <mmichelson@digium.com>

        * main/config.c: Fix the logic in config_text_file_save so that if
          an UpdateConfig manager action is issued and the file specified
          in DstFileName does not yet exist, an error is not returned.
          (closes issue #13341) Reported by: vadim Patches: 13341.patch
          uploaded by putnopvut (license 60) (with small modification from
          seanbright)

2008-08-25 15:33 +0000 [r139764]  Steve Murphy <murf@digium.com>

        * main/pbx.c, res/res_features.c: This patch reverts the changes
          made via 139347, and 139635, as users are seeing adverse
          difference. I will un-close 13251. Back to the drawing board/
          concept/ beginning/ whatever!

2008-08-22 22:24 +0000 [r139635]  Steve Murphy <murf@digium.com>

        * res/res_features.c: I found some problems with the code I
          committed earlier, when I merged them into trunk, so I'm coming
          back to clean up. And, in the process, I found an error in the
          code I added to trunk and 1.6.x, that I'll fix using this patch
          also.

2008-08-22 21:36 +0000 [r139621]  Jeff Peeler <jpeeler@digium.com>

        * main/manager.c: (closes issue #13359) Reported by: Laureano
          Patches: originate_channel_check.patch uploaded by Laureano
          (license 265)

2008-08-22 19:45 +0000 [r139456-139553]  Mark Michelson <mmichelson@digium.com>

        * include/asterisk/threadstorage.h: Fix compilation when
          DEBUG_THREAD_LOCALS is selected (closes issue #13298) Reported
          by: snuffy Patches: bug13298_20080822.diff uploaded by snuffy
          (license 35)

        * main/frame.c: Remove show_frame_stats_deprecated since it is not
          used anywhere and causes build errors if building under dev-mode
          with TRACE_FRAMES selected in menuselect. (closes issue #13362)
          Reported by: snuffy

        * channels/chan_iax2.c: Fix the build. Thanks, mvanbaak!

        * channels/chan_iax2.c: Prevent a deadlock in chan_iax2 resulting
          from incorrect locking order between iax2_pvt and ast_channel
          structures. AST-13

2008-08-21 23:39 +0000 [r139387]  Jeff Peeler <jpeeler@digium.com>

        * channels/chan_dahdi.c: Fixes loop that could possibly never exit
          in the event of a channel never being able to be opened or
          specify after a restart. (closes issue #11017)

2008-08-21 23:03 +0000 [r139347]  Steve Murphy <murf@digium.com>

        * main/pbx.c, res/res_features.c: (closes issue #13251) Reported
          by: sergee Tested by: murf THis is a bold move for a static
          release fix, but I wouldn't have made it if I didn't feel
          confident (at least a *bit* confident) that it wouldn't mess
          everyone up. The reasoning goes something like this: 1. We simply
          cannot do anything with CDR's at the current point (in pbx.c,
          after the __ast_pbx_run loop). It's way too late to have any
          affect on the CDRs. The CDR is already posted and gone, and the
          remnants have been cleared. 2. I was very much afraid that moving
          the running of the 'h' extension down into the bridge code (where
          it would be now practical to do it), would result in a lot more
          calls to the 'h' exten, so I implemented it as another exten
          under another name, but found, to my pleasant surprise, that
          there was a 1:1 correspondence to the running of the 'h' exten in
          the pbx_run loop, and the new spot at the end of the bridge. So,
          I ifdef'd out the current 'h' loop, and moved it into the bridge
          code. The only difference I can see is the stuff about the
          AST_PBX_KEEPALIVE, and hopefully, if this is still an important
          decision point, I can replicate it if there are complaints. To be
          perfectly honest, the KEEPALIVE situation is not totally clear to
          me, and how it relates to a post-bridge situation is less clear.
          I suspect the users will point out everything in total clarity if
          this steps on anyone's toes! 3. I temporarily swap the bridge_cdr
          into the channel before running the 'h' exten, which makes it
          possible for users to edit the cdr before it goes out the door.
          And, of course, with the endbeforehexten config var set, the
          users can also get at the billsec/duration vals. After the h
          exten finishes, the cdr is swapped back and processing continues
          as normal. Please, all who deal with CDR's, please test this
          version of Asterisk, and file bug reports as appropriate!

2008-08-21 10:11 +0000 [r139283]  Philippe Sultan <philippe.sultan@gmail.com>

        * channels/chan_gtalk.c: Apply fix for issue #13310 to branch 1.4,
          too.

2008-08-20 22:14 +0000 [r139213]  Russell Bryant <russell@digium.com>

        * apps/app_chanspy.c: Fix a crash in the ChanSpy application. The
          issue here is that if you call ChanSpy and specify a spy group,
          and sit in the application long enough looping through the
          channel list, you will eventually run out of stack space and the
          application with exit with a seg fault. The backtrace was always
          inside of a harmless snprintf() call, so it was tricky to track
          down. However, it turned out that the call to snprintf() was just
          the biggest stack consumer in this code path, so it would always
          be the first one to hit the boundary. (closes issue #13338)
          Reported by: ruddy

2008-08-20 19:52 +0000 [r139151]  Shaun Ruffell <sruffell@digium.com>

        * codecs/codec_dahdi.c: Fix bug where the samples were not accurate
          when in G723 mode, which would cause the timestamp field of the
          RTP header to be invalid.

2008-08-20 19:35 +0000 [r139145]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_dahdi.c, configure,
          include/asterisk/autoconfig.h.in, configure.ac: Backport support
          for Zaptel/DAHDI channel-level alarms from trunk/1.6, because not
          doing so just makes it difficult for people with channels that
          are in alarm when Asterisk starts up to get them going once the
          alarm is cleared (closes issue #12160) Reported by: tzafrir
          Patches: asterisk-chanalarms_14.patch uploaded by tzafrir
          (license 46) Tested by: tzafrir

2008-08-20 17:14 +0000 [r139074]  Steve Murphy <murf@digium.com>

        * main/cdr.c: (closes issue #13263) Reported by: brainy Tested by:
          murf The specialized reset routine is tromping on the flags field
          of the CDR. I made a change to not reset the DISABLED bit. This
          should get rid of this problem.

2008-08-20 15:37 +0000 [r139015]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: sip_read should properly handle a NULL
          return from sip_rtp_read. (closes issue #13257) Reported by:
          travishein

2008-08-19 23:22 +0000 [r138949]  Jeff Peeler <jpeeler@digium.com>

        * include/asterisk/dahdi_compat.h: add DAHDI_POLICY_WHEN_FULL
          compatability define for Zaptel

2008-08-19 23:17 +0000 [r138942]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_agent.c: Reset agent_pvt variables back to the
          values in agents.conf (from what the corresponding channel
          variables were set to) when the agent logs out. (closes issue
          #13098) Reported by: davidw Patches:
          20080731__issue13098_agent_ackcall_not_reset.diff uploaded by
          bbryant (license 36) Tested by: davidw

2008-08-19 22:56 +0000 [r138938]  Jeff Peeler <jpeeler@digium.com>

        * channels/chan_dahdi.c: Add configuration option to
          chan_dahdi.conf to allow buffering policy and number of buffers
          to be configured per channel. Syntax: buffers=<num of
          buffers>,<policy> Where the number of buffers is some
          non-negative integer and the policy is either "full", "half", or
          "immediate".

2008-08-19 18:50 +0000 [r138685-138886]  Mark Michelson <mmichelson@digium.com>

        * apps/app_chanspy.c: Add a lock and unlock prior to the
          destruction of the chanspy_ds lock to ensure that no other
          threads still have it locked. While this should not happen under
          normal circumstances, it appears that if the spyer and spyee hang
          up at nearly the same time, the following may occur. 1.
          ast_channel_free is called on the spyee's channel. 2. The chanspy
          datastore is removed from the spyee's channel in
          ast_channel_free. 3. In the spyer's thread, the spyer attempts to
          remove and destroy the datastore from the spyee channel, but the
          datastore has already been removed in step 2, so the spyer
          continues in the code. 4. The spyee's thread continues and calls
          the datastore's destroy callback, chanspy_ds_destroy. This
          involves locking the chanspy_ds. 5. Now the spyer attempts to
          destroy the chanspy_ds lock. The problem is that in step 4, the
          spyee has locked this lock, meaning that the spyer is attempting
          to destroy a lock which is currently locked by another thread.
          The backtrace provided in issue #12969 supports the idea that
          this is possible (and has even occurred). This commit does not
          close the issue, but should help in preventing one type of crash
          associated with the use of app_chanspy.

        * apps/app_queue.c: Change the inequalities used in app_queue with
          regards to timeouts from being strict to non-strict for more
          accuracy. (closes issue #13239) Reported by: atis Patches:
          app_queue_timeouts_v2.patch uploaded by atis (license 242)

2008-08-18 16:57 +0000 [r138663]  Kevin P. Fleming <kpfleming@digium.com>

        * codecs/codec_dahdi.c: look for transcoder in proper place based
          on build against Zaptel or DAHDI

2008-08-18 11:57 +0000 [r138569]  Sean Bright <sean.bright@gmail.com>

        * channels/chan_dahdi.c: You know what's awesome? Code that
          compiles... ;)

2008-08-18 02:05 +0000 [r138516]  Jeff Peeler <jpeeler@digium.com>

        * channels/chan_dahdi.c: fix compilation warnings

2008-08-16 01:12 +0000 [r138309-138360]  Jeff Peeler <jpeeler@digium.com>

        * channels/chan_dahdi.c: fixes use count to properly decrement if
          an active dahdi channel is destroyed allowing module to be
          unloaded

        * channels/chan_dahdi.c: add forgotten locks around ss_thread_count
          in ss_thread for dahdi restart

2008-08-15 22:33 +0000 [r138258]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_sip.c, configs/sip.conf.sample: More fixes for
          realtime peers. (closes issue #12921) Reported by: Nuitari
          Patches: 20080804__bug12921.diff.txt uploaded by Corydon76
          (license 14) 20080815__bug12921.diff.txt uploaded by Corydon76
          (license 14) Tested by: Corydon76

2008-08-15 21:28 +0000 [r138119-138238]  Jeff Peeler <jpeeler@digium.com>

        * channels/chan_dahdi.c: initialize condition variable
          ss_thread_complete using ast_cond_init

        * channels/chan_dahdi.c: declared static mutexes using
          AST_MUTEX_DEFINE_STATIC macro

        * channels/chan_dahdi.c: Fixes the dahdi restart functionality.
          Dahdi restart allows one to restart all DAHDI channels, even if
          they are currently in use. This is different from unloading and
          then loading the module since unloading requires the use count to
          be zero. Reloading the module is different in that the signalling
          is not changed from what it was originally configured. Also, this
          fixes not closing all the file descriptors for D-channels upon
          module unload (which would prevent loading the module
          afterwards). (closes issue #11017)

2008-08-15 15:07 +0000 [r138027]  Russell Bryant <russell@digium.com>

        * main/autoservice.c: Ensure that when a hangup occurs in
          autoservice, that a hangup frame gets properly deferred to be
          read from the channel owner when it gets taken out of
          autoservice. (closes issue #12874) Reported by: dimas Patches:
          v1-12874.patch uploaded by dimas (license 88)

2008-08-15 14:51 +0000 [r137847-138023]  Tilghman Lesher <tlesher@digium.com>

        * funcs/func_strings.c: Additional check for more string specifiers
          than arguments. (closes issue #13299) Reported by: adomjan
          Patches: 20080813__bug13299.diff.txt uploaded by Corydon76
          (license 14) func_strings.c-sprintf.patch uploaded by adomjan
          (license 487) Tested by: adomjan

        * channels/chan_dahdi.c: Oops, wrong direction

        * channels/chan_dahdi.c: When creating the secondary subchannel
          name, it is necessary to compare to the existing channel name
          without the "Zap/" or "DAHDI/" prefix, since our test string is
          also without that prefix. (closes issue #13027) Reported by:
          dferrer Patches: chan_zap-1.4.21.1_fix2.patch uploaded by dferrer
          (license 525) (Slightly modified by me, to compensate for both
          names)

2008-08-14 14:05 +0000 [r137731]  Russell Bryant <russell@digium.com>

        * configs/sip.conf.sample: Comments in this config file were
          aligned only if your tab size was set to 8. So, convert tabs to
          spaces so that things should be aligned regardless of what tab
          size you use in your editor.

2008-08-14 02:03 +0000 [r137677-137679]  Kevin P. Fleming <kpfleming@digium.com>

        * Zaptel-to-DAHDI.txt: forgot one module name that changed

        * include/asterisk/dahdi_compat.h, channels/chan_dahdi.c,
          build_tools/menuselect-deps.in, configure, configure.ac,
          codecs/codec_dahdi.c: add support for Zaptel versions that
          contain the new transcoder interface

2008-08-13 21:35 +0000 [r137580]  Jeff Peeler <jpeeler@digium.com>

        * channels/chan_dahdi.c: Register DAHDISendKeypadFacility
          application if dahdi_chan_mode is set to DAHDI + Zap. Mark
          ZapSendKeypadFacility application as deprecated on usage.

2008-08-13 20:46 +0000 [r137527-137530]  Kevin P. Fleming <kpfleming@digium.com>

        * Zaptel-to-DAHDI.txt (added): add document describing what users
          will need to be aware of when upgrading to this version and using
          DAHDI

        * apps/app_meetme.c: remove some more chan_zap references

        * doc/asterisk-conf.txt, channels/chan_dahdi.c: document
          dahdichanname option in doc/asterisk-conf.txt make chan_dahdi
          read its configuration from zapata.conf if dahdichanname has been
          set to 'no'

2008-08-13 14:33 +0000 [r137348-137405]  Sean Bright <sean.bright@gmail.com>

        * doc/cdrdriver.txt: Update docs to reflect the change to cdr_tds

        * cdr/cdr_tds.c: Bring cdr_tds in line with the other CDR backends
          and have it try to store CDR(userfield) if it is set. The new
          behavior is to check for the userfield column on module load, and
          if it exists, we will store CDR(userfield) when CDRs are written.
          A similar patch already went into trunk and 1.6.0. (closes issue
          #13290) Reported by: falves11

2008-08-11 13:33 +0000 [r137188]  Kevin P. Fleming <kpfleming@digium.com>

        * apps/app_meetme.c: convert this module to be able to handle DAHDI
          or Zaptel (reported on asterisk-users, don't know how this got
          missed before)

2008-08-11 00:20 +0000 [r137138]  Tilghman Lesher <tlesher@digium.com>

        * res/res_odbc.c: Deallocate database connection handle on
          disconnect, as we allocate another one on connect. (closes issue
          #13271) Reported by: dveiga

2008-08-09 17:11 +0000 [r136999]  Russell Bryant <russell@digium.com>

        * configure, configure.ac: Ensure PBX_DAHDI_TRANSCODE will evaluate
          to 0 if not found instead of empty. pointed out by tzafrir on
          #asterisk-dev

2008-08-09 15:25 +0000 [r136946]  Tilghman Lesher <tlesher@digium.com>

        * /, include/asterisk/compat.h, include/asterisk/astobj2.h: Merged
          revisions 136945 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008)
          | 2 lines Regression fixes for Solaris ........

2008-08-08 00:15 +0000 [r136726]  Steve Murphy <murf@digium.com>

        * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
          pbx/ael/ael-test/ref.ael-vtest13,
          pbx/ael/ael-test/ref.ael-ntest10, pbx/pbx_ael.c,
          include/asterisk/ael_structs.h: (closes issue #13236) Reported
          by: korihor Wow, this one was a challenge! I regrouped and ran a
          new strategy for setting the ~~MACRO~~ value; I set it once per
          extension, up near the top. It is only set if there is a switch
          in the extension. So, I had to put in a chunk of code to detect a
          switch in the pval tree. I moved the code to insert the set of
          ~~exten~~ up to the beginning of the gen_prios routine, instead
          of down in the switch code. I learned that I have to push the
          detection of the switches down into the code, so everywhere I
          create a new exten in gen_prios, I make sure to pass onto it the
          values of the mother_exten first, and the exten next. I had to
          add a couple fields to the exten struct to accomplish this, in
          the ael_structs.h file. The checked field makes it so we don't
          repeat the switch search if it's been done. I also updated the
          regressions.

2008-08-07 18:25 +0000 [r136560]  Kevin P. Fleming <kpfleming@digium.com>

        * build_tools/menuselect-deps.in, configure, configure.ac: change
          the required dependency for codec_dahdi to only be satisfied by
          DAHDI and not Zaptel, as the new transcoder interface is only in
          DAHDI

2008-08-07 18:14 +0000 [r136544]  Shaun Ruffell <sruffell@digium.com>

        * codecs/codec_dahdi.c: Updated codec_dahdi to use the new
          transcoder interface in the first DAHDI release. Codec dahdi no
          longer functions with the transcoder interface in zaptel at this
          time (which the last zaptel release was 1.4.11). NOTE: Still
          needs an update to the configure script to make sure that
          codec_dahdi is only built if the new transcoder interface is
          present in the drivers. (Issue: DAHDI-42)

2008-08-07 16:50 +0000 [r136488]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_queue.c: Update persistent state on all exit conditions.
          (closes issue #12916) Reported by: sgenyuk Patches:
          app_queue.patch.txt uploaded by neutrino88 (license 297) Tested
          by: sgenyuk, aragon

2008-08-07 16:30 +0000 [r136404-136484]  Kevin P. Fleming <kpfleming@digium.com>

        * include/asterisk/doxyref.h: add a raw list of all libraries that
          any part of Asterisk links directly to

        * apps/app_voicemail.c: work around a bug in gcc-4.2.3 that
          incorrectly ignores the casting away of 'const' for pointers when
          the developer knows it is safe to do so

        * Makefile: remove config.cache during distclean, in case the user
          is using autoconf caching

2008-08-07 01:31 +0000 [r136304-136348]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_dahdi.c: Also, parse
          useincomingcalleridonzaptransfer (and add appropriate deprecation
          warnings).

        * channels/chan_dahdi.c: For backwards compatibility with previous
          1.4 versions which used "zapchan" in users.conf, ensure that we
          still support it.

2008-08-06 21:18 +0000 [r136241]  Richard Mudgett <rmudgett@digium.com>

        * channels/misdn_config.c, channels/chan_misdn.c,
          configs/misdn.conf.sample: * The allowed_bearers setting in
          misdn.conf misspelled one of its options: digital_restricted. *
          Fixed some other spelling errors and typos.

2008-08-06 20:42 +0000 [r136238]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: We only need to unregister the QueueStatus
          manager command once on an unload

2008-08-06 20:14 +0000 [r136190]  Tilghman Lesher <tlesher@digium.com>

        * contrib/init.d/rc.redhat.asterisk: -C option takes a filename,
          not a directory path. (closes issue #13007) Reported by:
          klaus3000

2008-08-06 18:58 +0000 [r136168]  Russell Bryant <russell@digium.com>

        * Makefile: Remove the use of --no-print-directory when compiling
          subdirectories. This allows vim :make functionality to work
          properly when errors have occurred in the build. Without printing
          the directories, vim did not know how to find the file that the
          error occurred in. If the extra bit of build noise annoys anyone,
          just let me know, and I'll make this optional.

2008-08-06 15:58 +0000 [r136062]  Mark Michelson <mmichelson@digium.com>

        * main/rtp.c, channels/chan_skinny.c: Since adding the
          AST_CONTROL_SRCUPDATE frame type, there are places where
          ast_rtp_new_source may be called where the tech_pvt of a channel
          may not yet have an rtp structure allocated. This caused a crash
          in chan_skinny, which was fixed earlier, but now the same crash
          has been reported against chan_h323 as well. It seems that the
          best solution is to modify ast_rtp_new_source to not attempt to
          set the marker bit if the rtp structure passed in is NULL. This
          change to ast_rtp_new_source also allows the removal of what is
          now a redundant pointer check from chan_skinny. (closes issue
          #13247) Reported by: pj

2008-08-06 03:53 +0000 [r135899-135949]  Tilghman Lesher <tlesher@digium.com>

        * main/channel.c: Fix a longstanding bug in channel walking logic,
          and fix the explanation to make sense. (Closes issue #13124)

        * main/translate.c: Since powerof() can return an error condition,
          it's foolhardy not to detect and deal with that condition.
          (Related to issue #13240)

        * include/asterisk/threadstorage.h, include/asterisk/utils.h: 1)
          Bugfix for debugging code 2) Reduce compiler warnings for another
          section of debugging code (Closes issue #13237)

2008-08-06 00:29 +0000 [r135841-135850]  Mark Michelson <mmichelson@digium.com>

        * /: Remove properties that should not be here

        * apps/app_skel.c: Revert inadvertent changes to app_skel that
          occurred when I was testing for a memory leak

        * include/asterisk/abstract_jb.h, main/channel.c, /,
          apps/app_skel.c, main/abstract_jb.c, main/fixedjitterbuf.h:
          Merging the issue11259 branch. The purpose of this branch was to
          take into account "burps" which could cause jitterbuffers to
          misbehave. One such example is if the L option to Dial() were
          used to inject audio into a bridged conversation at regular
          intervals. Since the audio here was not passed through the
          jitterbuffer, it would cause a gap in the jitterbuffer's
          timestamps which would cause a frames to be dropped for a brief
          period. Now ast_generic_bridge will empty and reset the
          jitterbuffer each time it is called. This causes injected audio
          to be handled properly. ast_generic_bridge also will empty and
          reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE
          frame since the change in audio source could negatively affect
          the jitterbuffer. All of this was made possible by adding a new
          public API call to the abstract_jb called ast_jb_empty_and_reset.
          (closes issue #11259) Reported by: plack Tested by: putnopvut

2008-08-05 23:13 +0000 [r135799]  Steve Murphy <murf@digium.com>

        * apps/app_dial.c, main/cdr.c, main/channel.c, res/res_features.c,
          include/asterisk/cdr.h: (closes issue #12982) Reported by: bcnit
          Tested by: murf I discovered that also, in the previous bug fixes
          and changes, the cdr.conf 'unanswered' option is not being
          obeyed, so I fixed this. And, yes, there are two 'answer' times
          involved in this scenario, and I would agree with you, that the
          first answer time is the time that should appear in the CDR. (the
          second 'answer' time is the time that the bridge was begun). I
          made the necessary adjustments, recording the first answer time
          into the peer cdr, and then using that to override the bridge
          cdr's value. To get the 'unanswered' CDRs to appear, I purposely
          output them, using the dial cmd to mark them as DIALED (with a
          new flag), and outputting them if they bear that flag, and you
          are in the right mode. I also corrected one small mention of the
          Zap device to equally consider the dahdi device. I heavily tested
          10-sec-wait macros in dial, and without the macro call; I tested
          hangups while the macro was running vs. letting the macro
          complete and the bridge form. Looks OK. Removed all the
          instrumentation and debug.

2008-08-05 21:34 +0000 [r135747]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_iax2.c: In a conversion to use ast_strlen_zero, the
          meaning of the flag IAX_HASCALLERID was perverted. This change
          reverts IAX2 to the original meaning, which was, that the
          callerid set on the client should be overridden on the server,
          even if that means the resulting callerid is blank. In other
          words, if you set "callerid=" in the IAX config, then the
          callerid should be overridden to blank, even if set on the
          client. Note that there's a distinction, even on realtime,
          between the field not existing (NULL in databases) and the field
          existing, but set to blank (override callerid to blank).

2008-08-05 13:25 +0000 [r135597]  Sean Bright <sean.bright@gmail.com>

        * main/cli.c: Use PATH_MAX for filenames

2008-08-04 20:15 +0000 [r135536]  Russell Bryant <russell@digium.com>

        * configs/chan_dahdi.conf.sample: fix a config sample typo

2008-08-04 17:07 +0000 [r135479-135482]  Tilghman Lesher <tlesher@digium.com>

        * contrib/init.d/rc.mandrake.asterisk: Define ASTSBINDIR for script

        * apps/app_voicemail.c: Memory leak on unload (closes issue #13231)
          Reported by: eliel Patches: app_voicemail.leak.patch uploaded by
          eliel (license 64)

2008-08-04 16:26 +0000 [r135473]  Russell Bryant <russell@digium.com>

        * configs/chan_dahdi.conf.sample: Add a minor clarification to the
          documentation of mohinterpret and mohsuggest

2008-08-01 11:43 +0000 [r135055-135058]  Michiel van Baak <michiel@vanbaak.info>

        * apps/app_ices.c: make app_ices compile on OpenBSD.

        * channels/chan_skinny.c: fix some potential deadlocks in
          chan_skinny (closes issue #13215) Reported by: qwell Patches:
          2008080100_bug13215.diff.txt uploaded by mvanbaak (license 7)
          Tested by: mvanbaak

2008-07-31 22:18 +0000 [r134983]  Kevin P. Fleming <kpfleming@digium.com>

        * main/http.c: accomodate users who seem to lack a sense of humor
          :-)

2008-07-31 21:53 +0000 [r134976]  Tilghman Lesher <tlesher@digium.com>

        * sample.call, main/manager.c, pbx/pbx_spool.c: Specify codecs in
          callfiles and manager, to allow video calls to be set up from
          callfiles and AMI. (closes issue #9531) Reported by: Geisj
          Patches: 20080715__bug9531__1.4.diff.txt uploaded by Corydon76
          (license 14) 20080715__bug9531__1.6.0.diff.txt uploaded by
          Corydon76 (license 14) Tested by: Corydon76

2008-07-31 19:37 +0000 [r134915]  Russell Bryant <russell@digium.com>

        * apps/app_ices.c: Get app_ices working again (closes issue #12981)
          Reported by: dlogan Patches:
          20080709__app_ices_v2_update_trunk.diff uploaded by bbryant
          (license 36) 20080709__app_ices_v2_update_14.diff uploaded by
          bbryant (license 36) Tested by: bbryant

2008-07-31 19:23 +0000 [r134883]  Steve Murphy <murf@digium.com>

        * res/res_features.c: (closes issue #11849) Reported by: greyvoip
          Tested by: murf OK, a few days of debugging, a bunch of
          instrumentation in chan_sip, main/channel.c, main/pbx.c, etc. and
          5 solid notebook pages of notes later, I have made the small
          tweek necc. to get the start time right on the second CDR when: A
          Calls B B answ. A hits Xfer button on sip phone, A dials C and
          hits the OK button, A hangs up C answers ringing phone B and C
          converse B and/or C hangs up But does not harm the scenario
          where: A Calls B B answ. B hits xfer button on sip phone, B dials
          C and hits the OK button, B hangs up C answers ringing phone A
          and C converse A and/or C hangs up The difference in start times
          on the second CDR is because of a Masquerade on the B channel
          when the xfer number is sent. It ends up replacing the CDR on the
          B channel with a duplicate, which ends up getting tossed out. We
          keep a pointer to the first CDR, and update *that* after the
          bridge closes. But, only if the CDR has changed. I hope this
          change is specific enough not to muck up any current CDR-based
          apps. In my defence, I assert that the previous information was
          wrong, and this change fixes it, and possibly other similar
          scenarios. I wonder if I should be doing the same thing for the
          channel, as I did for the peer, but I can't think of a scenario
          this might affect. I leave it, then, as an exersize for the
          users, to find the scenario where the chan's CDR changes and
          loses the proper start time.

2008-07-31 16:45 +0000 [r134814]  Russell Bryant <russell@digium.com>

        * channels/iax2-parser.c: In case we have some processing threads
          that free more frames than they allocate, do not let the frame
          cache grow forever. (closes issue #13160) Reported by: tavius
          Tested by: tavius, russell

2008-07-31 15:56 +0000 [r134758]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Add more timeout checks into app_queue,
          specifically targeting areas where an unknown and potentially
          long time has just elapsed. Also added a check to try_calling()
          to return early if the timeout has elapsed instead of potentially
          setting a negative timeout for the call (thus making it have *no*
          timeout at all). (closes issue #13186) Reported by:
          miquel_cabrespina Patches: 13186.diff uploaded by putnopvut
          (license 60) Tested by: miquel_cabrespina

2008-07-30 22:39 +0000 [r134704]  Tilghman Lesher <tlesher@digium.com>

        * main/sched.c, include/asterisk/sched.h: Oops, wrong define

2008-07-30 22:02 +0000 [r134652]  Steve Murphy <murf@digium.com>

        * pbx/pbx_ael.c: (closes issue #13197) Reported by: pj (closes
          issue #13051) Reported by: pj This patch substitutes commas in
          the expr supplied to the if () statement, as in if ( expr ) ...
          This solves both the bugs above, and makes the source symmetric
          with switch statements, which were earlier reported to need this
          sort of treatment. I tested this using the examples, both for the
          compiler and at run time. Looks good.

2008-07-30 21:38 +0000 [r134649]  Tilghman Lesher <tlesher@digium.com>

        * configure, configure.ac: Qwell pointed out, via IRC, that the
          previous fix only worked when explicitly set. When nothing is
          set, and the option is implied, it breaks, because configure sets
          the prefix to 'NONE'. Fixing.

2008-07-30 20:37 +0000 [r134540-134595]  Russell Bryant <russell@digium.com>

        * pbx/pbx_dundi.c: Reduce stack consumption by 12.5% of the max
          stack size to fix a crash when compiled with LOW_MEMORY. (closes
          issue #13154) Reported by: edantie

        * funcs/func_curl.c: Fix a memory leak in func_curl. Every thread
          that used this function leaked an allocation the size of a
          pointer. (reported by jmls in #asterisk-dev)

2008-07-30 19:47 +0000 [r134480-134536]  Tilghman Lesher <tlesher@digium.com>

        * configure, configure.ac: Only override sysconfdir and mandir when
          prefix=/usr (closes issue #13093) Reported by: pabelanger

        * res/res_agi.c: launch_netscript sometimes returns -1, which fails
          to set AGISTATUS. Map failure to -1, so that AGISTATUS is always
          set. (closes issue #13199) Reported by: smw1218

2008-07-30 18:31 +0000 [r134475]  Mark Michelson <mmichelson@digium.com>

        * main/app.c: Fix a spot where a function could return without
          bringing a channel out of autoservice.

2008-07-30 15:29 +0000 [r134254-134352]  Kevin P. Fleming <kpfleming@digium.com>

        * Makefile: use the proper method for building version.h

        * include/asterisk/dahdi_compat.h, apps/app_dahdibarge.c,
          channels/chan_dahdi.c, apps/app_meetme.c, apps/app_flash.c,
          apps/app_dahdiscan.c, apps/app_dahdiras.c, codecs/codec_dahdi.c:
          build against the now-typedef-free dahdi/user.h

2008-07-29 15:54 +0000 [r134223]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Merging the imap_consistency branch. The
          main aim of this branch was to make the IMAP code function in the
          same manner as the ODBC code does, eliminating the need for so
          many IMAP-specific code chunks. The focal point of all of this
          work was to make the various macros (e.g. RETRIEVE, DISPOSE)
          functionally equivalent. While doing the above work, I also fixed
          a few bugs that I came across in my testing. Among these were 1.
          Fixed message forwarding. This was completely broken when using
          IMAP. 2. Fixed the inability to save new messages as old and vice
          versa. 3. Fixed the "delete" options in voicemail.conf when using
          IMAP storage. Even though a few bugs were fixed and the code is a
          lot more consistent, the one thing that was *not* improved in
          this branch was performance. The merge of this to trunk may not
          come immediately due to the amount of work it will probably
          involve. (closes issue #12764) Reported by: balsamcn

2008-07-28 21:50 +0000 [r134161]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_voicemail.c: Detect when sox fails to raise the volume,
          because sox can't read the file. (closes issue #12939) Reported
          by: rickbradley Patches: 20080728__bug12939.diff.txt uploaded by
          Corydon76 (license 14) Tested by: rickbradley

2008-07-26 15:31 +0000 [r133980]  Russell Bryant <russell@digium.com>

        * main/asterisk.c, include/asterisk/doxyref.h: Add the licensing
          section to the docs in 1.4, as well, so that we can work on
          having an accurate list for each version of Asterisk that is
          supported

2008-07-25 18:00 +0000 [r133649-133709]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_voicemail.c: Remove unnecessary mmap flag (Closes issue
          #13161)

        * main/channel.c, channels/chan_agent.c, main/devicestate.c: Fix
          some errant device states by making the devicestate API more
          strict in terms of the device argument (only without the unique
          identifier appended). (closes issue #12771) Reported by: davidw
          Patches: 20080717__bug12771.diff.txt uploaded by Corydon76
          (license 14) Tested by: davidw, jvandal, murf

2008-07-25 15:00 +0000 [r133578]  Russell Bryant <russell@digium.com>

        * /, LICENSE: Merged revisions 133577 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r133577 | russell | 2008-07-25 10:00:13 -0500 (Fri, 25 Jul 2008)
          | 2 lines Fix the IAX2 URI for calling Digium ........

2008-07-25 14:40 +0000 [r133572]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: We need to make sure to null-terminate the
          "name" portion of SIP URI parameters so that there are no bogus
          comparisons. Thanks to bbryant for pointing this out.

2008-07-24 21:17 +0000 [r133361-133488]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_sip.c: Fix rtautoclear and rtcachefriends (Closes
          issue #12707)

        * /: Blocked revisions 133360 via svnmerge ........ r133360 |
          tilghman | 2008-07-23 22:46:01 -0500 (Wed, 23 Jul 2008) | 2 lines
          This part was not correctly patched for AST-2008-010. ........

2008-07-23 21:49 +0000 [r133295]  Jason Parker <jparker@digium.com>

        * channels/chan_dahdi.c: inbandrelease is gone - it's now
          inbanddisconnect

2008-07-23 21:05 +0000 [r133226-133237]  Kevin P. Fleming <kpfleming@digium.com>

        * include/asterisk/stringfields.h, main/utils.c: revert an
          optimization that broke ABI... thanks russell!

        * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c,
          apps/app_dahdibarge.c, channels/chan_dahdi.c,
          apps/app_dahdiras.c: make some more changes to the dahdi/zap
          channel name support stuff to ensure allthe globals are 'const',
          and clean up mmichelson's changes to app_chanspy to simplify the
          code

2008-07-23 19:39 +0000 [r132974-133169]  Mark Michelson <mmichelson@digium.com>

        * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c,
          channels/chan_dahdi.c: As suggested by seanbright, the
          PSEUDO_CHAN_LEN in app_chanspy should be set at load time, not at
          compile time, since dahdi_chan_name is determined at load time.
          Also changed the next_unique_id_to_use to have the static
          qualifier. Also added the dahdi_chan_name_len variable so that
          strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for
          the suggestion.

        * apps/app_chanspy.c: Zap/pseudo is ten characters, but
          DAHDI/pseudo is twelve. The strncmp call in next_channel should
          account for this.

        * apps/app_chanspy.c: Update the "last" channel in next_channel in
          app_chanspy so that the same pseudo channel isn't constantly
          returned. related to issue #13124

        * channels/chan_dahdi.c: Small cleanup. Move the declaration of the
          DAHDI_SPANINFO variable to the block where it is used. This
          allows one less #ifdef HAVE_PRI to clutter things up. Thanks to
          Tzafrir for pointing this out on #asterisk-dev

        * channels/chan_dahdi.c: Fix building of chan_dahdi when HAVE_PRI
          is not defined.

2008-07-23 15:52 +0000 [r132872-132942]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_dahdi.c: ensure that after a channel is created, if
          it happened to be in 'channel alarm' state, when that alarm
          clears we won't generate a spurious 'alarm cleared' message
          (closes issue #12160) Reported by: tzafrir

        * include/asterisk/stringfields.h, main/utils.c: minor optimization
          for stringfields: when a field is being set to a larger value
          than it currently contains and it happens to be the most recent
          field allocated from the currentl pool, it is possible to 'grow'
          it without having to waste the space it is currently using (or
          potentially even allocate a new pool)

2008-07-23 11:37 +0000 [r132826]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c: another Fix because of r119585, this
          commit has broken high frequented BRI Ports, there was a
          possibility that a channel, that was marked as in_use would be
          reused later, the corresponding port could got stuck then. So it
          is recommended to upgrade for chan_misdn users.

2008-07-22 22:14 +0000 [r132790]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: Allow Spiraled INVITEs to work correctly
          within Asterisk. Prior to this change, a spiraled INVITE would
          cause a 482 Loop Detected to be sent to the caller. With this
          change, if a potential loop is detected, the Request-URI is
          inspected to see if it has changed from what was originally
          received. If pedantic mode is on, then this inspection is fully
          RFC 3261 compliant. If pedantic mode is not on, then a string
          comparison is used to test the equality of the two R-URIs. This
          has been tested by using OpenSER to rewrite the R-URI and send
          the INVITE back to Asterisk. (closes issue #7403) Reported by:
          stephen_dredge

2008-07-22 22:11 +0000 [r132784-132787]  Kevin P. Fleming <kpfleming@digium.com>

        * include/asterisk/options.h, main/asterisk.c,
          apps/app_dahdibarge.c, channels/chan_dahdi.c, apps/app_flash.c,
          apps/app_dahdiras.c: fix up namespace pollution for
          dahdi_chan_mode enum correct registration of AMI actions in
          chan_dahdi; in zap-only mode, only register the Zap flavors of
          the actions (and use Zap prefixes for headers and acks), but in
          dahdi+zap mode, register both Zap and DAHDI flavors of actions

        * Makefile.rules: add rules to create preprocessor output... useful
          for debugging macros

2008-07-22 21:19 +0000 [r132713]  Tilghman Lesher <tlesher@digium.com>

        * configs/iax.conf.sample, /, channels/chan_iax2.c: Merged
          revisions 132711 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008)
          | 2 lines Fixes for AST-2008-010 and AST-2008-011 ........

2008-07-22 21:17 +0000 [r132704-132712]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_dahdi.c: ensure that if any alarms exist at channel
          creation time, they are handled identically to if they occurred
          later, so that later alarm clearing will work properly and 'make
          sense' (closes issue #12160) Reported by: tzafrir

        * configure, configure.ac, acinclude.m4: make AST_C_COMPILE_CHECK
          able to print a 'pretty' description of what it is doing

2008-07-22 20:10 +0000 [r132645]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c, doc/sip-retransmit.txt (added): The most
          common question on the #asterisk iRC channel and on mailing lists
          seems to be in regards to an error message when retransmit fails.
          This is frequently misunderstood as a failure of Asterisk, not a
          failure of the network to reach the other party. This document
          tries to assist the Asterisk user in sorting out these issues by
          explaining the logic and pointing at some possible causes.
          Hopefully, we will get other questions now :-)

2008-07-22 19:57 +0000 [r132571-132642]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_dahdi.c: correct wording in comment

        * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, configure,
          include/asterisk/autoconfig.h.in, configure.ac: use renamed
          libpri API call for controlling this feature (was improperly
          named before)

        * channels/chan_dahdi.c: teach chan_dahdi how to find the D-channel
          on BRI spans, and don't attempt to use channel 24 as a D-channel
          on spans of unexpected sizes

2008-07-21 20:51 +0000 [r132506-132507]  Brett Bryant <bbryant@digium.com>

        * apps/app_sendtext.c: Fix a bug where SENDTEXTSTATUS isn't set
          properly when it isn't supported on a channel (yet _another_
          useful patch by eliel). (issue #13081) Reported by: eliel
          Patches: app_sendtext1.4.c uploaded by eliel (license 64) Tested
          by: eliel

        * channels/chan_iax2.c: Fix a bug in 1.4 branch with iax2 channels
          not being removed when a call was rejected (from the calling box,
          not the box that denied the registration). Related to revisions
          132466 in trunk, and 132467 in 1.6.0. Earlier I had accidently
          tested 1.4 with a backport from those revisions, so I didn't see
          this problem (oops).

2008-07-19 16:45 +0000 [r132311]  Kevin P. Fleming <kpfleming@digium.com>

        * LICENSE: grant a license exception to allow distribution of
          Asterisk binaries that use the UW IMAP Toolkit (which is licensed
          under a non-GPL-compatible license)

2008-07-18 19:06 +0000 [r131970-132112]  Tilghman Lesher <tlesher@digium.com>

        * main/say.c: Fix for Taiwanese number syntax (closes issue #12319)
          Reported by: CharlesWang Patches: saynumber-tw-1.4.18.1.patch
          uploaded by CharlesWang (license 444)

        * main/config.c: Textual clarification (closes issue #13106)
          Reported by: flefoll Patches:
          config.c.br14.120173.patch-unknown-directive uploaded by flefoll
          (license 244)

        * include/asterisk/sched.h, channels/chan_iax2.c: Spinlock within
          the destroy, to allow a scheduled job to continue, if it's
          waiting on the mutex which the destroy thread has.

        * main/sched.c: Oops

        * main/sched.c, include/asterisk/sched.h: Preserve ABI
          compatibility with last change

        * main/sched.c, include/asterisk/sched.h, channels/chan_iax2.c:
          Make the ast_assert call within ast_sched_del report something
          useful.

2008-07-18 16:15 +0000 [r131921]  Kevin P. Fleming <kpfleming@digium.com>

        * main/dlfcn.c (removed), main/loader.c, main/Makefile,
          include/asterisk/dlfcn-compat.h (removed): remove the dlfcn
          compatibility stuff, because no platforms that Asterisk currently
          runs on it use it, and it doesn't build anyway

2008-07-18 15:34 +0000 [r131915]  Brett Bryant <bbryant@digium.com>

        * res/res_features.c: Fix a bug in blind transfers where the
          BLINDTRANSFER variable isn't always set to the other end of the
          blind transfer. (closes issue #12586)

2008-07-17 20:35 +0000 [r131790]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_dahdi.c: Revert part of issue #5620 (revision 6965)
          as it appears that it was in error. This should fix talk call
          progress on analog lines. (closes issue #12178) Reported by:
          michael-fig Patches: 20080717__bug12178.diff.txt uploaded by
          Corydon76 (license 14)

2008-07-16 22:17 +0000 [r131491]  Brett Bryant <bbryant@digium.com>

        * channels/chan_iax2.c: Fix a bug in iax2 registration that allowed
          peers to register with case-insensitive names (user_cmp_cb and
          peer_cmp_cb are now both case-sensitive). (closes issue #13091)

2008-07-16 21:46 +0000 [r131480]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_iax2.c: Apparently, in certain cases, a callno is
          already destroyed when iax2_destroy is called.

2008-07-16 20:47 +0000 [r131421]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Always ensure that the channel's tech_pvt
          reference is NULL after calling the destroy callback. (closes
          issue #13060) Reported by: jpgrayson Patches:
          chan_iax2_tech_pvt_crash.patch uploaded by jpgrayson (license
          492)

2008-07-16 20:23 +0000 [r131299-131369]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Move the init_queue call back to where it used
          to be (changed Sept 12 last year). It was moved then to prevent a
          memory leak. Since then, the same memory leak recurred and was
          fixed in a better way. Now it has been found that the placement
          of this init_queue call can cause problems if a realtime queue
          has values changed to an empty string. The problem is that the
          default value for that queue parameter would not be set. (closes
          issue #13084) Reported by: elbriga

        * apps/app_queue.c: Apparently, "thread safety" is important,
          whatever that means. :P (Thanks Russell!)

        * apps/app_queue.c: Make absolutely certain that the transfer
          datastore is removed from the calling channel once the caller is
          finished in the queue. This could have weird con- sequences when
          dialing local queue members when multiple transfers occur on a
          single call. Also fixed a memory leak that would occur when an
          attended transfer occurred from a queue member. (closes issue
          #13047) Reported by: festr

2008-07-16 17:53 +0000 [r131242]  Steve Murphy <murf@digium.com>

        * pbx/pbx_ael.c: (closes issue #13090) Reported by: murf The
          problem was that, esoteric as it is, because the hangerupper
          context immediately preceded the std-priv-extent macro, that the
          checking code accidentally would fall from traversing hangerupper
          into the std-priv-exten macro, where it would hit the hangerupper
          in the 'includes', and proceed into an infinite recursion. A
          small fix to traverse into the statements of the context instead
          of the context solves this issue. I also added some commented out
          printfs for debug, which were pretty handy in the face of a dorky
          gdb. This was a problem around since the package was first
          written; but evidently pretty rare in turning up in the field.

2008-07-15 17:47 +0000 [r131012]  Michiel van Baak <michiel@vanbaak.info>

        * main/cdr.c: remove 4 lines of redundant code. (closes issue
          #13080) Reported by: gknispel_proformatique Patches:
          trunk_ast_cdr_setapp.patch uploaded by gknispel (license 261)

2008-07-15 17:19 +0000 [r130889-130959]  Tilghman Lesher <tlesher@digium.com>

        * main/manager.c, channels/chan_sip.c: astman_send_error does not
          need a newline appended -- the API takes care of that for us.
          (closes issue #13068) Reported by: gknispel_proformatique
          Patches: asterisk_1_4_astman_send.patch uploaded by gknispel
          (license 261) asterisk_trunk_astman_send.patch uploaded by
          gknispel (license 261)

        * channels/chan_iax2.c: Override the callerid in all cases when the
          callerid is set in the user, not just when a remote callerid is
          set. Also, if not set in the user, allow the remote CallerID to
          pass through. (closes issue #12875) Reported by: dimas Patches:
          20080714__bug12875.diff.txt uploaded by Corydon76 (license 14)

2008-07-14 17:50 +0000 [r130792]  Mark Michelson <mmichelson@digium.com>

        * apps/app_dial.c: Add a check to the CAN_EARLY_BRIDGE macro in
          app_dial to be sure there are no audiohooks present on the
          channels involved. This fixed a one-way audio situation I had in
          my test setup. I couldn't find any open issues that suggested
          one-way audio with regards to mixmonitor (or other audiohook)
          usage, though.

2008-07-14 17:10 +0000 [r130735]  Michiel van Baak <michiel@vanbaak.info>

        * main/dnsmgr.c: notify the user that dnsmgr refresh wont work when
          dnsmgr is not enabled. Previously this command would
          automagically appear and disappear. This was confusing. (closes
          issue #12796) Reported by: chappell Patches:
          dnsmgr_refresh_3.diff uploaded by chappell (license 8) Tested by:
          russell, chappell, mvanbaak

2008-07-14 10:38 +0000 [r130634]  Russell Bryant <russell@digium.com>

        * main/audiohook.c: Bump up the debug level for a message.

2008-07-13 22:48 +0000 [r130573]  Michiel van Baak <michiel@vanbaak.info>

        * main/manager.c: fix memory leak when originate from manager
          cannot create a thread (closes issue #13069) Reported by:
          gknispel_proformatique Patches:
          asterisk_trunk_action_originate.patch uploaded by gknispel
          (license 261) Tested by: gknispel_proformatique, mvanbaak

2008-07-13 17:56 +0000 [r130514]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_iax2.c: Reverting 2 changesets, as it breaks
          incoming IAX2 calls (Related to issue #12963) Reported by:
          mvanbaak

2008-07-12 10:25 +0000 [r130373]  Michiel van Baak <michiel@vanbaak.info>

        * pbx/pbx_ael.c: in 1.4 the functions still have | as argument
          seperator. This commit fixes the use of RAND in the ael random
          function. (closes issue #13061) Reported by: danpwi

2008-07-11 22:23 +0000 [r130298-130317]  Kevin P. Fleming <kpfleming@digium.com>

        * Makefile: forcibly remove the modules that are changing names

        * include/asterisk/options.h, main/asterisk.c, cdr/cdr_csv.c,
          Makefile, main/channel.c, apps/app_dahdibarge.c,
          channels/chan_dahdi.c, doc/hardware.txt, apps/app_flash.c,
          apps/app_dahdiras.c, main/file.c,
          contrib/utils/zones2indications.c, include/asterisk/channel.h,
          channels/chan_iax2.c: a whole pile of Zaptel/DAHDI compatibility
          work, with lots more to come... this tree is not yet ready for
          users to be easily upgrading or switching, but it needs to be :-)

2008-07-11 20:03 +0000 [r130173-130236]  Mark Michelson <mmichelson@digium.com>

        * main/audiohook.c: Remove redundant logic

        * main/audiohook.c: Fix a typo in audiohook_read_frame_both. While
          this change has not been proven to fix any specific issue, it is
          incorrect and could cause unforeseen problems.

2008-07-11 18:51 +0000 [r130102-130169]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_iax2.c: Ensure that a destination callno of 0 will
          not match for frames that do not start a dialog (new, lagrq, and
          ping). (closes issue #12963) Reported by: russellb Patches:
          chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492)

        * channels/chan_agent.c: Pass the devicestate from an underlying
          channel up through the Agent channel. This should make the Agent
          always report the correct device state, even when the underlying
          channel is used for other purposes. (closes issue #12773)
          Reported by: davidw Patches: 20080710__bug12773.diff.txt uploaded
          by Corydon76 (license 14) Tested by: davidw

2008-07-11 16:08 +0000 [r130039-130042]  Kevin P. Fleming <kpfleming@digium.com>

        * doc/configuration.txt, configs/extensions.conf.sample,
          configs/sla.conf.sample, configs/zapata.conf.sample (removed),
          contrib/scripts/autosupport, README,
          configs/chan_dahdi.conf.sample (added), channels/chan_dahdi.c,
          include/asterisk/doxyref.h, doc/sla.tex, doc/ael.txt,
          configs/extensions.ael.sample, configs/smdi.conf.sample: new
          installations should be using DAHDI instead of Zaptel, so the
          sample config file is now chan_dahdi.conf instead of zapata.conf
          also, convert remaining references to zapata.conf in various
          places

        * configs/zapata.conf.sample, channels/chan_dahdi.c, configure,
          include/asterisk/autoconfig.h.in, configure.ac: add support for a
          configuration parameter for 'inband audio during RELEASE', which
          is currently mandatory in libpri-1.4.4 but will become
          configurable in libpri-1.4.5 later today (related to issue
          #13042)

2008-07-11 14:18 +0000 [r129970]  Russell Bryant <russell@digium.com>

        * include/asterisk/astobj.h: add a simple ASTOBJ_TRYWRLOCK macro
          ...

2008-07-11 14:14 +0000 [r129907-129967]  Kevin P. Fleming <kpfleming@digium.com>

        * main/astmm.c: simplify calculation

        * main/astmm.c: fix a flaw found while experimenting with structure
          alignment and padding; low-fence checking would not work properly
          on 64-bit platforms, because the compiler was putting 4 bytes of
          padding between the fence field and the allocation memory block
          added a very obvious runtime warning if this condition reoccurs,
          so the developer who broke it can be chastised into fixing it :-)

        * sounds/Makefile: don't attempt to set user/group ownership of
          extracted sound files (reported on asterisk-users) (closes issue
          #13059)

2008-07-10 21:57 +0000 [r129741-129803]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_iax2.c: Correctly deal with duplicate NEW frames
          (due to retransmission). Also, fixup the destination call number
          matching to be more strict and reliable. (closes issue #12963)
          Reported by: jpgrayson Patches: chan_iax2_dup_new_fix3.patch
          uploaded by jpgrayson (license 492) Tested by: jpgrayson,
          Corydon76

        * res/res_config_odbc.c: Oops

2008-07-10 16:03 +0000 [r129567]  Russell Bryant <russell@digium.com>

        * sample.call: Note that pbx_spool.so is the module used for call
          files (inspired by a question in #asterisk)

2008-07-10 13:57 +0000 [r129505]  Sean Bright <sean.bright@gmail.com>

        * main/editline: Update svn:ignore

2008-07-09 19:32 +0000 [r129436]  Mark Michelson <mmichelson@digium.com>

        * main/rtp.c: Fix a problem where inbound rfc2833 audio would be
          sent to the core instead of being P2P bridged. When the core
          regenerated the rfc2833 packet for the outbound leg, the SSRC
          would be different than the RTP audio on the call leg causing
          DTMF detection issues on the far end. (closes issue #12955)
          Reported by: tonyredstone Patches: dynamic_rtp.patch uploaded by
          tsearle (license 373) Tested by: tonyredstone

2008-07-09 13:41 +0000 [r129343]  Sean Bright <sean.bright@gmail.com>

        * main/editline/makelist (removed), main/editline/makelist.in
          (added), main/editline/configure, main/editline/Makefile.in,
          main/editline/configure.in: Look for the system installed awk
          instead of assuming it's at /usr/bin/awk. Pointed out by jmls via
          #asterisk-dev.

2008-07-08 21:31 +0000 [r129158-129208]  Mark Michelson <mmichelson@digium.com>

        * doc/imapstorage.txt: Update documentation to have the correct
          option name

        * apps/app_voicemail.c, doc/imapstorage.txt: Backport TCP-related
          timeouts to IMAP voicemail in 1.4 since it should solve bugs
          people are experiencing. Specifically, there are times where
          communication with the IMAP server causes system calls to block
          forever. If this should happen when querying the mailbox so that
          chan_sip's do_monitor thread can send MWI to a phone, it means
          that SIP calls cannot be processed any more. The timeout options
          are outlined in doc/imapstorage.txt. Defaults for the timeouts
          are sixty seconds. (closes issue #12987) Reported by: mthomasslo

2008-07-08 20:27 +0000 [r129047-129149]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_dial.c, channels/chan_sip.c, include/asterisk/causes.h:
          Cause SIP to return a 480 instead of a 404 when a sip peer
          exists, but is not registered. (closes issue #12885) Reported by:
          ibc Patches: 20080701__bug12885__2.diff.txt uploaded by Corydon76
          (license 14) Tested by: ibc

        * channels/chan_iax2.c: Timestamp decoding for video mini-frames is
          bogus, because the timestamp only includes 15 bits, unlike voice
          frames, which contain a 16-bit timestamp. (closes issue #13013)
          Reported by: jpgrayson Patches: chan_iax2_unwrap_ts.patch
          uploaded by jpgrayson (license 492)

2008-07-08 09:52 +0000 [r128912-128950]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Don't hangup the call if we can't resolve
          the Contact if there's a proxy route set for the call. ---- This
          comment was added a while ago and today it hit me badly. /* OEJ:
          Possible issue that may need a check: If we have a proxy route
          between us and the device, should we care about resolving the
          contact or should we just send it? */

        * channels/chan_sip.c: Fix issues where repeated messages where
          ignored, but retransmitted reliably instead of unreliably.
          Reported by: johan Patches: 12746.txt uploaded by oej (license
          306) Tested by: johan (issue #12746)

2008-07-08 00:01 +0000 [r128812-128856]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_voicemail.c: Check for non-NULL before stripping
          characters. (closes issue #12954) Reported by: bfsworks Patches:
          20080701__bug12954.diff.txt uploaded by Corydon76 (license 14)
          Tested by: deti

        * apps/app_voicemail.c: Stop using deprecated method, as requested
          by Kevin.

2008-07-07 22:41 +0000 [r128795]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix handling of when a pvt disappears.
          Properly return the pvt locked and don't hold the pvt lock while
          destroying the ast_channel. (closes issue #13014) Reported by:
          jpgrayson Patches: chan_iax2_ast_iax2_new2.patch uploaded by
          jpgrayson (license 492)

2008-07-07 20:47 +0000 [r128737]  Sean Bright <sean.bright@gmail.com>

        * channels/chan_iax2.c: Remove spurious trailing whitespace from
          log messages and fix a spelling error in a log message. (closes
          issue #13017) Reported by: jpgrayson Patches:
          chan_iax2_space_after_newline.patch uploaded by jpgrayson
          (license 492) chan_iax2_spelling.patch uploaded by jpgrayson
          (license 492)

2008-07-07 17:02 +0000 [r128639]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_iax2.c: By using the iaxdynamicthreadcount to
          identify a thread, it was possible for thread identifiers to be
          duplicated. By using a globally-unique monotonically- increasing
          integer, this is now avoided. (closes issue #13009) Reported by:
          jpgrayson Patches: chan_iax2_dyn_threadnum.patch uploaded by
          jpgrayson (license 492)

2008-07-07 16:51 +0000 [r128637]  Kevin P. Fleming <kpfleming@digium.com>

        * configure, configure.ac: use tzafrir's patch to fix this problem
          properly... i made the previous set of changes without thoroughly
          testing them, doh! (closes issue #12911) Reported by: tzafrir
          Patches: custum_dahdi_configure_2.diff uploaded by tzafrir
          (license 46) Tested by: tzafrir

2008-07-04 16:11 +0000 [r127973-128029]  Tilghman Lesher <tlesher@digium.com>

        * pbx/pbx_config.c: Move the free down one

        * main/pbx.c, include/asterisk/pbx.h, pbx/pbx_config.c: Fix the
          'dialplan remove extension' logic, so that it a) works with
          cidmatch, and b) completes contexts correctly when the extension
          is ambiguous. (closes issue #12980) Reported by: licedey Patches:
          20080703__bug12980.diff.txt uploaded by Corydon76 (license 14)
          Tested by: Corydon76

2008-07-03 22:20 +0000 [r127754-127895]  Kevin P. Fleming <kpfleming@digium.com>

        * apps/Makefile: remove this, it has been moved to the main
          Makefile

        * Makefile, main/editline/np/vis.c: a couple of small
          Solaris-related fixes (closes issue #11885) Reported by: snuffy,
          asgaroth

        * configure, main/Makefile, configure.ac, acinclude.m4: ensure that
          DAHDI_INCLUDE and ZAPTEL_INCLUDE are added in all the places
          needed improve AST_EXT_LIB_CHECK to accept (and remember)
          additional CFLAGS data like it does in trunk already (closes
          issue #12911) Reported by: tzafrir

2008-07-03 00:16 +0000 [r127663]  Steve Murphy <murf@digium.com>

        * main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c,
          channels/chan_sip.c, res/res_features.c, include/asterisk/cdr.h:
          The CDRfix4/5/6 omnibus cdr fixes. (closes issue #10927) Reported
          by: murf Tested by: murf, deeperror (closes issue #12907)
          Reported by: falves11 Tested by: murf, falves11 (closes issue
          #11849) Reported by: greyvoip As to 11849, I think these changes
          fix the core problems brought up in that bug, but perhaps not the
          more global problems created by the limitations of CDR's
          themselves not being oriented around transfers. Reopen if necc,
          but bug reports are not the best medium for enhancement
          discussions. We need to start a second-generation CDR
          standardization effort to cover transfers. (closes issue #11093)
          Reported by: rossbeer Tested by: greyvoip, murf

2008-07-02 20:47 +0000 [r127560]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_agent.c: Fix thread-safety of some of the
          pbx_builtin_getvar_helper calls

2008-07-02 19:47 +0000 [r127501]  Tilghman Lesher <tlesher@digium.com>

        * main/acl.c: Merged revisions 127466 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/trunk ........ r127466 |
          tilghman | 2008-07-02 13:31:11 -0500 (Wed, 02 Jul 2008) | 6 lines
          Solaris fix (closes issue #12949) Reported by: snuffy Patches:
          bug_12949.diff uploaded by snuffy (license 35) ........

2008-07-01 23:36 +0000 [r127244]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Add error message to failed open(2) calls
          inside the copy() function of app_voicemail. This idea came as
          part of my work in helping to resolve issue #12764.

2008-07-01 20:25 +0000 [r126999-127133]  Tilghman Lesher <tlesher@digium.com>

        * build_tools/cflags.xml, channels/chan_iax2.c: Disable the old,
          slow search for matching callno in chan_iax2 (but allow it to be
          reenabled for debugging)

        * channels/chan_iax2.c: Oops

        * channels/chan_iax2.c: Change around how we schedule pings and
          lagrqs, and fix a reason why the jobs were not getting properly
          cancelled. (closes issue #12903) Reported by: stevedavies
          Patches: 20080620__bug12903__2.diff.txt uploaded by Corydon76
          (license 14) Tested by: stevedavies

        * channels/chan_iax2.c: Suppress annoying warning by finding the
          remaining cases where the callno is not in the hash.

2008-07-01 14:59 +0000 [r126735-126902]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Use domain part of SIP uri in register=
          configuration as fromdomain. Reported by: one47 Patches:
          sip-reg-fromdom2.dpatch uploaded by one47 (license 23) (closes
          issue #12474)

        * channels/chan_sip.c: Handle escaped URI's in call pickups. Patch
          by oej and IgorG. Reported by: IgorG Patches:
          bug12299-11062-v2.patch uploaded by IgorG (license 20) Tested by:
          IgorG, oej (closes issue #12299)

        * configs/sip.conf.sample: Clear up documentation on "domain="
          setting in sip.conf Reported by: davidw (closes issue #12413)

        * channels/chan_sip.c: Report 200 OK to all in-dialog OPTIONs
          requests (to confirm that the dialog exist). Don't bother
          checking the request URI. (closes issue #11264) Reported by: ibc

        * channels/chan_sip.c: Fix bad XML for hold notification. Reported
          by: gowen72 Patches: hold.patch uploaded by gowen72 (license 432)
          (closes issue #12942)

2008-06-30 23:11 +0000 [r126680]  Jeff Peeler <jpeeler@digium.com>

        * channels/chan_dahdi.c: Load the proper channel configuration file
          based on which driver was detected.

2008-06-30 22:30 +0000 [r126674]  Tilghman Lesher <tlesher@digium.com>

        * configs/zapata.conf.sample: Add note about other names for
          EuroISDN

2008-06-30 16:05 +0000 [r126573]  Russell Bryant <russell@digium.com>

        * include/asterisk/lock.h: Fix a typo in the non-DEBUG_THREADS
          version of the recently added DEADLOCK_AVOIDANCE() macro. This
          caused the lock to not actually be released, and as a result, not
          avoid deadlocks at all. This resolves the issues reported in the
          last while about Asterisk locking up all over the place (and most
          commonly, in chan_iax2). (closes issue #12927) (closes issue
          #12940) (closes issue #12925) (potentially closes others ...)

2008-06-30 12:50 +0000 [r126516]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Send all responses to an INVITE reliably, so
          that we retransmit if we don't get an ACK and also fail if we
          don't get the very same precious ACK. Based on patch by tsearle,
          with my own additions. (closes issue #12951) Reported by: tsearle
          Patches: busy_retransmit.patch uploaded by tsearle (license 373)

2008-06-29 18:05 +0000 [r126395]  Kevin P. Fleming <kpfleming@digium.com>

        * pbx/Makefile: ignore warnings for prototypes in GTK headers

2008-06-27 22:01 +0000 [r125740-126056]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_sip.c: When we get a 408 Timeout, don't stop trying
          to re-register. (closes issue #12863) Reported by: ricvil

        * include/asterisk/tonezone_compat.h: Since HAVE_DAHDI is defined
          to HAVE_ZAPTEL in dahdi_compat.h, we must first check for
          HAVE_ZAPTEL. (closes issue #12938) Reported by: opticron Patches:
          tonezone_compat.diff uploaded by opticron (license 267)

        * main/utils.c, include/asterisk/lock.h: In this debugging
          function, copy to a buffer instead of using potentially unsafe
          pointers.

        * channels/chan_local.c: Add proper deadlock avoidance. (closes
          issue #12914) Reported by: ozan Patches:
          20080625__bug12914.diff.txt uploaded by Corydon76 (license 14)
          Tested by: ozan

2008-06-26 23:03 +0000 [r125587]  Jason Parker <jparker@digium.com>

        * main/utils.c: Make sure to unlock the lock_info lock (huh?).
          Possible deadlock?

2008-06-26 22:52 +0000 [r125476-125585]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Add the interface of a queue member to the
          output of the "queue show" command so that it can easily be
          associated with a queue member's name. This helps so that the
          appropriate queue member can be removed or paused since the
          interface is required, not the member's name. (closes issue
          #12783) Reported by: davevg Patches: app_queue.diff uploaded by
          davevg (license 209) with small mod from me

        * apps/app_queue.c: Backport of attended transfer queue_log patch
          from trunk. This patch allows for attended transfers to be logged
          in the queue_log the same way that blind transfers have always
          been. It was decided by popular opinion on the asterisk-dev
          mailing list that this should be backported to 1.4. Thanks to
          everyone who gave an opinion.

        * apps/app_queue.c: Prior to this patch, the "queue show" command
          used cached information for realtime queues instead of giving
          up-to-date info. Now realtime is queried for the latest and
          greatest in queue info. (closes issue #12858) Reported by: bcnit
          Patches: queue_show.patch uploaded by putnopvut (license 60)

2008-06-26 16:32 +0000 [r125384]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Add support for peer realm based auth (a few
          missing lines, the rest is well documented but never worked)

2008-06-26 15:30 +0000 [r125327]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_dahdi.c: ensure that (whenever possible) if we
          generate a log message because an ioctl() call to DAHDI/Zaptel
          failed, that we include the reason it failed by including the
          stringified error number (issue AST-80)

2008-06-26 11:01 +0000 [r125218-125276]  Tilghman Lesher <tlesher@digium.com>

        * main/rtp.c: Check for rtcp structure before trying to delete
          schedule. (closes issue #12872) Reported by: destiny6628 Patches:
          20080621__bug12872.diff.txt uploaded by Corydon76 (license 14)
          Tested by: destiny6628

        * configs/agents.conf.sample: Document ackcall=always. (closes
          issue #12852) Reported by: davidw

2008-06-25 22:21 +0000 [r125132]  Kevin P. Fleming <kpfleming@digium.com>

        * apps/app_rpt.c, include/asterisk/dahdi_compat.h,
          channels/chan_dahdi.c, configure,
          include/asterisk/tonezone_compat.h (added), configure.ac: allow
          tonezone to live in a different place than DAHDI/Zaptel, since
          dahdi-tools and dahdi-linux are now separate packages and can be
          installed in different places don't include tonezone.h in
          dahdi_compat.h, because only a couple of modules need it get
          app_rpt building again after the DAHDI changes (closes issue
          #12911) Reported by: tzafrir

2008-06-25 00:46 +0000 [r124908-124965]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_dahdi.c: Pvt deadlock causes some channels to get
          stuck in Reserved status. (closes issue #12621) Reported by:
          fabianoheringer Patches: 20080612__bug12621.diff.txt uploaded by
          Corydon76 (license 14) Tested by: fabianoheringer

        * apps/app_voicemail.c: Occasionally control characters find their
          way into CallerID. These need to be stripped prior to placing
          CallerID in the headers of an email. (closes issue #12759)
          Reported by: RobH Patches: 20080602__bug12759__2.diff.txt
          uploaded by Corydon76 (license 14) Tested by: RobH

        * channels/chan_sip.c: Don't access the pvt structure if unable to
          acquire the lock. (closes issue #12162) Reported by: norman
          Patches: 12162-lockfail.diff uploaded by qwell (license 4)

2008-06-23 21:22 +0000 [r124743]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_iax2.c: emit a warning if the old IAX2 call
          searching code finds a call when the new code did not... so that
          we can get rid of the old code in 2-3 months

2008-06-22 02:54 +0000 [r124540]  Steve Murphy <murf@digium.com>

        * apps/app_forkcdr.c: (closes issue #12910) Reported by: chris-mac
          Sorry, my testing did not contain the simple case of forkCDR(v),
          I am much embarrassed to admit. If I had, I would have more
          solidly initialized the opts element for varset.

2008-06-20 23:12 +0000 [r124395-124450]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_rpt.c: usleep with a value over 1,000,000 is
          nonportable. Changing to use sleep() instead. (closes issue
          #12814) Reported by: pputman Patches: app_rtp_sleep.patch
          uploaded by pputman (license 81)

        * main/app.c: If the last character in a string to be parsed is the
          delimiter, then we should count that final empty string as an
          additional argument.

2008-06-20 21:14 +0000 [r124372]  Jeff Gehlbach <jeffg@opennms.org>

        * doc/asterisk-mib.txt, doc/digium-mib.txt: Fix issues in
          digium-mib.txt and asterisk-mib.txt to placate smilint - bug
          12905

2008-06-20 20:16 +0000 [r124182-124315]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_local.c: When using a Local channel, started by a
          call file, with a destination of an AGI script, the AGI script
          does not always get notified of a hangup if the underlying
          channel hangs up early. (closes issue #11833) Reported by: IgorG
          Patches: local_hangup-v1.diff uploaded by IgorG (license 20)

        * channels/chan_dahdi.c: It's possible for a hangup to be received,
          even just after the initial cid spill. (closes issue #12453)
          Reported by: Alex728 Patches: 20080604__bug12453.diff.txt
          uploaded by Corydon76 (license 14)

2008-06-19 20:28 +0000 [r124112]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Fix IMAP forwarding so that messages are
          sent to the proper mailbox. (closes issue #12897) Reported by:
          jaroth Patches: destination_forward.patch uploaded by jaroth
          (license 50)

2008-06-19 19:55 +0000 [r124066]  Brett Bryant <bbryant@digium.com>

        * main/utils.c: Merge revision 124064 from trunk. Add errors that
          report any locks held by threads when they are being closed.

2008-06-19 16:58 +0000 [r123710-123930]  Tilghman Lesher <tlesher@digium.com>

        * main/channel.c: Change informative messages to use the _multiple
          variant when multiple formats are possible. (Closes issue #12848)
          Reported by klaus3000

        * build_tools/strip_nonapi: Only process 40 arguments (20 files) at
          once with xargs, because some older shells may force xargs to
          separate on an odd boundary. (Closes issue #12883) Reported by
          Nik Soggia

        * configs/sip.conf.sample: Correct description of notifyringing
          option. (Closes issue #12890) Reported by gminet

        * main/asterisk.c: The RDTSC instruction was introduced on the
          Pentium line of microprocessors, and is not compatible with
          certain 586 clones, like Cyrix. Hence, asking for i386
          compatibility was always incorrect. See
          http://en.wikipedia.org/wiki/RDTSC (Closes issue #12886) Reported
          by tecnoxarxa

        * main/say.c, doc/lang (added), doc/lang/hebrew.ods (added): Add
          support for saying numbers in Hebrew. (closes issue #11662)
          Reported by: greenfieldtech Patches: say.c.patch-12042008
          uploaded by greenfieldtech (license 369) Hebrew-Sounds.ods
          uploaded by greenfieldtech (with signficant changes to the
          spreadsheet by me)

        * pbx/pbx_spool.c: Set the variables top-down, so that if a script
          sets a variable more than once, the last one will take
          precedence. (closes issue #12673) Reported by: phber Patches:
          20080519__bug12673.diff.txt uploaded by Corydon76 (license 14)

2008-06-17 20:26 +0000 [r123485]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: Make chan_sip build under dev mode with
          compilers >= GCC 4.2 Thanks to jpeeler for alerting me of this

2008-06-17 18:56 +0000 [r123391]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_iax2.c: Fix 3 more places where not locking the
          structure could cause the wrong lock to be unlocked. (Closes
          issue #12795)

2008-06-17 18:09 +0000 [r123274-123333]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: Cisco BTS sends SIP responses with a tab
          between the Cseq number and SIP request method in the Cseq:
          header. Asterisk did not handle this properly, but with this
          patch, all is well. (closes issue #12834) Reported by: tobias_e
          Patches: 12834.patch uploaded by putnopvut (license 60) Tested
          by: tobias_e

        * apps/app_queue.c: davidw pointed out that the holdtime
          calculation used by app_queue does not use "boxcar" filtering as
          the comments say. The term "boxcar" means that the number of
          samples used to calculate stays constant, with new samples
          replacing the oldest ones. The queue holdtime calculation uses
          all holdtime samples collected since the queue was loaded, so the
          comment has been changed to be accurate. (closes issue #12781)
          Reported by: davidw

2008-06-17 15:48 +0000 [r123271]  Russell Bryant <russell@digium.com>

        * main/astobj2.c: Fix a memory leak in astobj2 that was pointed out
          by seanbright. When a container got destroyed, the underlying
          bucket list entry for each object that was in the container at
          that time did not get free'd.

2008-06-16 19:50 +0000 [r123110-123113]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_mgcp.c, channels/chan_dahdi.c,
          channels/chan_skinny.c, channels/chan_h323.c,
          channels/chan_iax2.c: Port "hasvoicemail" change from SIP to
          other channel drivers

        * channels/chan_sip.c: People expect that if "hasvoicemail" is set
          in users.conf, even if "mailbox" isn't set, that SIP will detect
          a mailbox. (closes issue #12855) Reported by: PLL Patches:
          20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14)
          Tested by: PLL

2008-06-16 12:31 +0000 [r122869-122919]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Only compare the first 15 characters so that
          even if the charset is specified we still accept it as SDP.
          (closes issue #12803) Reported by: lanzaandrea Patches:
          chan_sip.c.diff uploaded by lanzaandrea (license 496)

        * channels/chan_sip.c: Don't send a BYE on a dialog that is already
          gone during a REFER. (closes issue #12865) Reported by: flefoll
          Patches: chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by
          flefoll (license 244)

2008-06-13 21:44 +0000 [r122713]  Mark Michelson <mmichelson@digium.com>

        * main/autoservice.c: Short circuit the loop in autoservice_run if
          there are no channels to poll. If we continued, then the result
          would be calling poll() with a NULL pollfd array. While this is
          fine with POSIX's poll(2) system call, those who use Asterisk's
          internal poll mechanism (Darwin systems) would have a failed
          assertion occur when poll is called. (related to issue #10342)

2008-06-13 18:57 +0000 [r122663]  Jeff Peeler <jpeeler@digium.com>

        * include/asterisk/dahdi_compat.h, res/res_musiconhold.c: fixed
          dahdi compatability header from assuming either dahdi or zaptel
          is installed (may not have either)

2008-06-13 17:45 +0000 [r122617]  Terry Wilson <twilson@digium.com>

        * apps/app_dial.c: Remove extra option from previous solution
          attempt

2008-06-13 17:36 +0000 [r122613]  Jeff Peeler <jpeeler@digium.com>

        * configure, configure.ac: (closes issue #12846) Reported by:
          Netview Tested by: jpeeler Use correct location to search for
          tonezone.

2008-06-13 16:29 +0000 [r122589]  Terry Wilson <twilson@digium.com>

        * apps/app_dial.c, res/res_features.c: This should fix the behavior
          of the 'T' dial feature being passed incorrectly to the
          transferee when builtin_atxfers are used. Also, doing a
          builtin_atxfer to parking was broken and is fixed here as well.
          (closes issue #11898) Reported by: sergee Tested by: otherwiseguy

2008-06-12 19:08 +0000 [r122314]  Jeff Peeler <jpeeler@digium.com>

        * main/indications.c, include/asterisk/dahdi_compat.h (added),
          main/loader.c, main/channel.c, channels/chan_dahdi.c (added),
          configure, apps/app_zapscan.c (removed), apps/app_zapras.c
          (removed), main/app.c, include/asterisk/options.h,
          apps/app_rpt.c, channels/chan_mgcp.c, apps/app_read.c,
          channels/chan_zap.c (removed), apps/app_page.c,
          include/asterisk/indications.h, apps/app_dahdiras.c (added),
          configure.ac, apps/app_disa.c, include/asterisk/channel.h,
          apps/app_getcpeid.c, apps/app_queue.c, apps/app_zapbarge.c
          (removed), channels/chan_misdn.c, apps/app_flash.c,
          build_tools/menuselect-deps.in, funcs/func_channel.c,
          main/file.c, res/snmp/agent.c, contrib/utils/zones2indications.c,
          codecs/codec_dahdi.c (added), res/res_indications.c,
          pbx/pbx_config.c, makeopts.in, apps/app_chanspy.c,
          main/asterisk.c, apps/app_dahdibarge.c (added),
          apps/app_meetme.c, include/asterisk/autoconfig.h.in,
          apps/app_dahdiscan.c (added), acinclude.m4,
          res/res_musiconhold.c, codecs/codec_zap.c (removed),
          channels/chan_iax2.c: Adds DAHDI support alongside Zaptel. DAHDI
          usage favored, but all Zap stuff should continue working. Release
          announcement to follow.

2008-06-12 18:50 +0000 [r122311]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Properly play a holdtime message if the
          announce-holdtime option is set to "once." (closes issue #12842)
          Reported by: ramonpeek Patches: patch001.diff uploaded by
          ramonpeek (license 266)

2008-06-12 18:22 +0000 [r122259]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix some race conditions that cause
          ast_assert() to report that chan_iax2 tried to remove an entry
          that wasn't in the scheduler

2008-06-12 15:46 +0000 [r122208]  Jeff Peeler <jpeeler@digium.com>

        * apps/app_parkandannounce.c, res/res_features.c: (closes issue
          #12193) Reported by: davidw Patch by: Corydon76, modified by me
          to work properly with ParkAndAnnounce app

2008-06-12 15:18 +0000 [r122130-122137]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_meetme.c: Flipflop the sections for two options, since
          the section for 'X' (exit context) may otherwise absorb
          keypresses meant for 's' (admin/user menu). (closes issue #12836)
          Reported by: blitzrage Patches: 20080611__bug12836.diff.txt
          uploaded by Corydon76 (license 14) Tested by: blitzrage

        * main/channel.c: Occasionally, the alertpipe loses its nonblocking
          status, so detect and correct that situation before it causes a
          deadlock. (Reported and tested by ctooley via #asterisk-dev)

2008-06-12 14:51 +0000 [r122127]  Steve Murphy <murf@digium.com>

        * main/cdr.c, apps/app_forkcdr.c: Arkadia tried to warn me, but the
          code added to ast_cdr_busy, _failed, and _noanswer was redundant.
          Didn't spot it until I was resolving conflicts in trunk. Ugh.
          Redundant code removed. It wasn't harmful. Just dumb.

2008-06-12  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.21 released.

2008-06-06  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.21-rc2 released.

2008-06-05 18:03 +0000 [r120731-120735]  Russell Bryant <russell@digium.com>

        * UPGRADE-1.2.txt: fix filename

        * UPGRADE-1.2.txt (added), UPGRADE.txt: Add the UPGRADE.txt file
          from Asterisk 1.2, for handy reference.

2008-06-05 16:56 +0000 [r120675]  Philippe Sultan <philippe.sultan@gmail.com>

        * res/res_jabber.c: Ignore appended resource when comparing JIDs.

2008-06-05 16:38 +0000 [r120671]  Russell Bryant <russell@digium.com>

        * doc/smdi.txt, res/res_smdi.c: It turns out that searching on the
          forwarding station isn't very useful for most people, so pull in
          the changes that allow searching for SMDI messages based on other
          components of the SMDI message. Also, update the SMDI
          documentation.

2008-06-04 22:05 +0000 [r120513]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Make sure that the string we set will survive
          the unref of the queue member. Thanks to Russell, who pointed
          this out.

2008-06-04 18:35 +0000 [r120425]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_zap.c: If we fail to setup the PRI request channel,
          don't continue, exit with an error. (closes issue #11989)
          Reported by: Corydon76 Patches: 20080213__zap_memleak.diff.txt
          uploaded by Corydon76 (license 14)

2008-06-04 16:26 +0000 [r120371]  Russell Bryant <russell@digium.com>

        * pbx/pbx_config.c: Make the "dialplan remove include" CLI command
          actually work. Also, tweak some formatting, and make the success
          message a little bit more clear. (closes AST-52)

2008-06-04 14:11 +0000 [r120285]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Tab completion when removing a member should
          give the member's interface, not the name, since the interface is
          what is expected for the command. (closes issue #12783) Reported
          by: davevg

2008-06-04 13:31 +0000 [r120282]  Joshua Colp <jcolp@digium.com>

        * main/pbx.c, pbx/pbx_config.c: Fix a log message and add a message
          for when the dialplan is done reloading. (closes issue #12716)
          Reported by: chappell Patches: dialplan_reload_2.diff uploaded by
          chappell (license 8)

2008-06-03 22:41 +0000 [r120226]  Tilghman Lesher <tlesher@digium.com>

        * pbx/pbx_loopback.c: Due to incorrect use of the
          AST_LIST_INSERT_HEAD() macro the loopback switch cannot perform
          any translation on the extension number before searching for it
          in the target context. (closes issue #12473) Reported by:
          chappell Patches: pbx_loopback.c.diff uploaded by chappell
          (license 8)

2008-06-03 22:15 +0000 [r120173]  Jeff Peeler <jpeeler@digium.com>

        * main/config.c: (closes issue #11594) Reported by: yem Tested by:
          yem This change decreases the buffer size allocated on the stack
          substantially in config_text_file_load when LOW_MEMORY is turned
          on. This change combined with the fix from revision 117462
          (making mkintf not copy the zt_chan_conf structure) was enough to
          prevent the crash.

2008-06-03 21:34 +0000 [r120168]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix another place where peer->callno could
          change at a very bad time, and also fix a place where a peer was
          used after the reference was released. (inspired by rev 120001)

2008-06-03  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.21-rc1 released.

2008-06-03 18:23 +0000 [r120001-120061]  Tilghman Lesher <tlesher@digium.com>

        * main/manager.c: When listing the manager users, managers in
          users.conf are not shown, even though they are allowed to
          connect. (closes issue #12594) Reported by: bkruse Patches:
          12594-managerusers-2.diff uploaded by qwell (license 4) Tested
          by: bkruse

        * channels/chan_iax2.c: Save the callno when we're poking, because
          our peer structure could change during deadlock avoidance (and
          thus we unlock the wrong callno, causing a cascade failure).
          (closes issue #12717) Reported by: gewfie Patches:
          20080525__bug12717.diff.txt uploaded by Corydon76 (license 14)
          Tested by: gewfie

2008-06-03 15:26 +0000 [r119929-119966]  Steve Murphy <murf@digium.com>

        * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
          pbx/ael/ael-test/ref.ael-vtest13,
          pbx/ael/ael-test/ref.ael-vtest17,
          pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
          pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5,
          pbx/ael/ael-test/ref.ael-test15: Updated the regressions on AEL.
          Hadn't updated this for the changes I made to preserve ${EXTEN}
          in switches, which affected several tests because it adds extra
          priorities, and at least one needed to be updated because of the
          removal of the empty extension warning message.

        * pbx/pbx_ael.c: as per
          http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html,
          which is a message from Philipp Kempgen, requesting that the
          WARNING that an extension is empty be reduced to a NOTICE or
          less, as empty extensions are syntactically possible, and no big
          deal. With which I agree, and have removed that WARNING message
          entirely. I think it is not necessary to see this message. It
          didn't state that a NoOp() was inserted automatically on your
          behalf, and really, as users, who cares? Why freak out dialplan
          writers with unnecessary warnings? The details of the
          machinations a compiler goes thru to produce working assembly
          code is of little interest to most programmers-- we will follow
          the unix principal of doing our work silently.

2008-06-03 14:46 +0000 [r119926]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Treat ECONNREFUSED as an error that will
          stop further retransmissions. (issue #AST-58, patch from
          Switchvox)

2008-06-02 20:08 +0000 [r119742-119838]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Revert a change made for issue #12479. This
          change caused a regression such that a dial string such as
          (IAX2/foo) did not automatically fall back to dialing the 's'
          extension anymore. (closes issue #12770) Reported by: dagmoller

        * main/manager.c: Improve CLI command blacklist checking for the
          command manager action. Previously, it did not handle case or
          whitespace properly. This made it possible for blacklisted
          commands to get executed anyway. (closes issue #12765)

2008-06-02 14:32 +0000 [r119740]  Philippe Sultan <philippe.sultan@gmail.com>

        * channels/chan_gtalk.c, res/res_jabber.c: Do not link the guest
          account with any configured XMPP client (in jabber.conf). The
          actual connection is made when a call comes in Asterisk. Fix the
          ast_aji_get_client function that was not able to retrieve an XMPP
          client from its JID. (closes issue #12085) Reported by: junky
          Tested by: phsultan

2008-06-02 12:30 +0000 [r119687]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Even of the first PING or LAGRQ doesn't get
          sent because it comes up too soon, make sure to reschedule so it
          gets sent later.

2008-06-02 09:29 +0000 [r119585-119636]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c: fixed compile issue when dev-mode is
          enabled

        * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h: Added
          counter for unhandled_bmsg Print, this prevents the logs to be
          flooded to fast and save CPU in this error scenario. Added
          'last_used' element to bc structure, when a bchannel changes from
          used to free this exact time will be marked in last_used. When a
          new channel is requested the find_free_chan function will check
          if the new empty channel was used within the last second, if yes
          it will search for the next channel, if no it will return this
          channel. This simple mechanism has prooven to prevent race
          conditions where the NT and TE tried to allocate the exact same
          channel at the same time (RELEASE cause: 44).

2008-06-02 01:06 +0000 [r119530-119533]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Change a debug message to an actual debug
          message

        * apps/app_dial.c: Fix another typo in documentation

2008-06-01 20:47 +0000 [r119478]  Michiel van Baak <michiel@vanbaak.info>

        * apps/app_dial.c: small typo fix 'retires' => 'retries'

2008-05-30 21:17 +0000 [r119404]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_queue.c: When joinempty=strict, it only failed on join
          if there were busy members. If all members were logged out OR
          paused, then it (incorrectly) let callers join the queue. (closes
          issue #12451) Reported by: davidw

2008-05-30 19:46 +0000 [r119354]  Joshua Colp <jcolp@digium.com>

        * main/autoservice.c: Fix a bug I found while testing for another
          issue.

2008-05-30 16:44 +0000 [r119301]  Michiel van Baak <michiel@vanbaak.info>

        * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
          contrib/init.d/rc.debian.asterisk,
          contrib/init.d/rc.mandrake.asterisk,
          contrib/init.d/rc.redhat.asterisk,
          contrib/init.d/rc.gentoo.asterisk,
          contrib/init.d/rc.slackware.asterisk: dont use a bashism way to
          check the $VERSION variable. The rc/init.d scripts, and
          safe_asterisk work on normal sh now again. Tested on: OpenBSD 4.2
          (me) Debian etch (me) Ubuntu Hardy (me and loloski) FC9 (loloski)
          (closes issue #12687) Reported by: loloski Patches:
          20080529-12687-safe_asterisk-fixversion.diff.txt uploaded by
          mvanbaak (license 7) Tested by: loloski, mvanbaak

2008-05-30 12:55 +0000 [r119076-119238]  Russell Bryant <russell@digium.com>

        * /, channels/chan_iax2.c: Merged revisions 119237 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30
          May 2008) | 7 lines - Instead of only enforcing destination call
          number checking on an ACK, check all full frames except for PING
          and LAGRQ, which may be sent by older versions too quickly to
          contain the destination call number. (As suggested by Tim Panton
          on the asterisk-dev list) - Merge changes from
          team/russell/iax2-frame-race, which prevents PING and LAGRQ from
          being sent before the destination call number is known. ........

        * main/autoservice.c: Fix a race condition in channel autoservice.
          There was still a small window of opportunity for a DTMF frame,
          or some other deferred frame type, to come in and get dropped.
          (closes issue #12656) (closes issue #12656) Reported by: dimas
          Patches: v3-12656.patch uploaded by dimas (license 88) -- with
          some modifications by me

        * include/asterisk/audiohook.h: Oddly enough, all of the contents
          of audiohook.h were in there twice. I have removed the second
          copy.

2008-05-29 20:24 +0000 [r119071]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_zap.c: Call waiting tone occurs too often, because
          it's getting serviced by both subchannels. (closes issue #11354)
          Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded
          by Corydon76 (license 14)

2008-05-29 19:04 +0000 [r118956-119012]  Russell Bryant <russell@digium.com>

        * apps/app_milliwatt.c: - Fix a typo in the argument to Playtones -
          use ast_safe_sleep() instead of calling the wait application
          (thanks to tilghman for pointing these out!)

        * /, channels/chan_iax2.c: Merged revisions 119008 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29
          May 2008) | 7 lines Merge changes from
          team/russell/iax2-another-fix-to-the-fix As described in the
          following post to the asterisk-dev mailing list, only enforce
          destination call numbers when processing an ACK.
          http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html
          (closes issue #12631) ........

        * apps/app_milliwatt.c: - Mark app_milliwatt dependent on
          res_indications (thanks to jsmith) - fix a typo in a log message
          (thanks to qwell)

        * apps/app_milliwatt.c: Change milliwatt to use the proper tone by
          default (1004 Hz) instead of 1000 Hz. An option is there to use
          1000 Hz for anyone that might want it.

2008-05-29 17:33 +0000 [r118953-118954]  Tilghman Lesher <tlesher@digium.com>

        * include/asterisk/lock.h: Define also when not DEBUG_THREADS

        * channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c,
          channels/chan_agent.c, channels/chan_alsa.c, main/utils.c,
          include/asterisk/lock.h, channels/chan_iax2.c: Add some debugging
          code that ensures that when we do deadlock avoidance, we don't
          lose the information about how a lock was originally acquired.

2008-05-29 00:25 +0000 [r118858]  Steve Murphy <murf@digium.com>

        * main/cdr.c, apps/app_forkcdr.c: (closes issue #10668) (closes
          issue #11721) (closes issue #12726) Reported by: arkadia Tested
          by: murf These changes: 1. revert the changes made via bug 10668;
          I should have known that such changes, even tho they made sense
          at the time, seemed like an omission, etc, were actually integral
          to the CDR system via forkCDR. It makes sense to me now that
          forkCDR didn't natively end any CDR's, but rather depended on
          natively closing them all at hangup time via traversing and
          closing them all, whether locked or not. I still don't completely
          understand the benefits of setvar and answer operating on locked
          cdrs, but I've seen enough to revert those changes also, and stop
          messing up users who depended on that behavior. bug 12726 found
          reverting the changes fixed his changes, and after a long review
          and working on forkCDR, I can see why. 2. Apply the suggested
          enhancements proposed in 10668, but in a completely compatible
          way. ForkCDR will behave exactly as before, but now has new
          options that will allow some actions to be taken that will
          slightly modify the outcome and side-effects of forkCDR. Based on
          conversations I've had with various people, these small tweaks
          will allow some users to get the behavior they need. For
          instance, users executing forkCDR in an AGI script will find the
          answer time set, and DISPOSITION set, a situation not covered
          when the routines were first written. 3. A small problem in the
          cdr serializer would output answer and end times even when they
          were not set. This is now fixed.

2008-05-28 16:10 +0000 [r118716]  Brett Bryant <bbryant@digium.com>

        * channels/chan_iax2.c: merge revision 118702 from trunk to 1.4 --
          Fixes a bug in chan_iax that uses send_command to poke a peer
          while a channel is unlocked in some cases, and because it can
          cause seemingly random failures could be related to some bugs in
          the tracker...

2008-05-28 14:23 +0000 [r118558-118646]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add an
          option to use the source IP address of RTP as the destination IP
          address of UDPTL when a specific option is enabled. If the remote
          side is properly configured (ports forwarded) then UDPTL will
          flow. (closes issue #10417) Reported by: cstadlmann

        * channels/chan_sip.c: Fix an issue where codec preferences were
          not set on dialogs that were not authenticated via a user or peer
          and allow framing to work without rtpmap in the SDP. (closes
          issue #12501) Reported by: slimey

2008-05-27 19:15 +0000 [r118551]  Tilghman Lesher <tlesher@digium.com>

        * main/cli.c: When showing an error message for a command, don't
          shorten the command output, as it tends to confuse the user (it's
          fine for suggesting other commands, however). Reported by:
          seanbright (on #asterisk-dev) Fixed by: me

2008-05-27 19:07 +0000 [r118509]  Mark Michelson <mmichelson@digium.com>

        * apps/app_chanspy.c: Russell noted to me that in the case that
          separate threads use their own addressing system, the fix I made
          for issue 12376 does not guarantee uniqueness to the datastores'
          uids. Though I know of no system that works this way, I am going
          to change this right now to prevent trying to track down some
          future bug that may occur and cause untold hours of debugging
          time to track down. The change involves using a global counter
          which increases with each new chanspy_ds which is created. This
          guarantees uniqueness.

2008-05-27 18:58 +0000 [r118465]  Tilghman Lesher <tlesher@digium.com>

        * main/asterisk.c: NULL character should terminate only commands
          back to the core, not log messages to the console. (closes issue
          #12731) Reported by: seanbright Patches:
          20080527__bug12731.diff.txt uploaded by Corydon76 (license 14)
          Tested by: seanbright

2008-05-27 17:17 +0000 [r118416]  Michiel van Baak <michiel@vanbaak.info>

        * apps/app_voicemail.c: small update to the g() option of
          app_voicemail to note that gain changes only work on zap channels
          right now. issue #12578 shows it's not clear right now.

2008-05-27 16:38 +0000 [r118365]  Mark Michelson <mmichelson@digium.com>

        * apps/app_chanspy.c: Add a unique id to the datastore allocated in
          app_chanspy since it is possible that multiple spies may be
          listening to the same channel. (closes issue #12376) Reported by:
          DougUDI Patches: 12376_chanspy_uid.diff uploaded by putnopvut
          (license 60) Tested by: destiny6628 (closes issue #12243)
          Reported by: atis

2008-05-27 15:45 +0000 [r118358]  Tilghman Lesher <tlesher@digium.com>

        * configs/queues.conf.sample: Add a note that pbx_config.so is
          needed for Local channels. (Closes issue #12671)

2008-05-25 16:02 +0000 [r118251]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_sip.c: Realtime flag affects construction in
          multiple ways, so consulting whether rtcachefriends was set was
          done too soon (needed to be done inside build_peer, not just as a
          flag to build_peer). Also, fullcontact needed to be
          reconstructed, because realtime separates the embedded ';' into
          multiple fields. (closes issue #12722) Reported by: barthpbx
          Patches: 20080525__bug12722.diff.txt uploaded by Corydon76
          (license 14) Tested by: barthpbx (Much of the discussion happened
          on #asterisk-dev for diagnosing this issue)

2008-05-23 21:21 +0000 [r118163]  Jeff Peeler <jpeeler@digium.com>

        * channels/chan_zap.c: Fix a few things I missed to ensure
          zt_chan_conf structure is not modified in mkintf

2008-05-23 13:18 +0000 [r118052-118055]  Tilghman Lesher <tlesher@digium.com>

        * include/asterisk/utils.h: Add format type checking for recently
          de-inlined function

        * doc/cli.txt (added), doc/00README.1st: Add information on using
          the Asterisk console, including tab command line completion.
          (Closes issue #12681)

2008-05-23 12:30 +0000 [r118048]  Russell Bryant <russell@digium.com>

        * include/asterisk/utils.h, main/utils.c: Don't declare a function
          that takes variable arguments as inline, because it's not valid,
          and on some compilers, will emit a warning.
          http://gcc.gnu.org/onlinedocs/gcc/Inline.html#Inline (closes
          issue #12289) Reported by: francesco_r Patches by Tilghman, final
          patch by me

2008-05-22 18:53 +0000 [r117809-117899]  Tilghman Lesher <tlesher@digium.com>

        * main/asterisk.c: Also remove preamble from asynchronous events
          (reported by jsmith on #asterisk-dev)

        * funcs/func_realtime.c: Take into account the length of delimiters
          when calculating result string length. (closes issue #12696)
          Reported by: adomjan Patches: func_realtime.c-longdelimiter.patch
          uploaded by adomjan (license 487)

2008-05-21 20:11 +0000 [r117582]  Jeff Peeler <jpeeler@digium.com>

        * channels/chan_zap.c: Ensure that passed in zt_chan_conf structure
          is not modified in mkintf.

2008-05-21 19:38 +0000 [r117574]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Apply the autoframing setting to dialogs
          that do not get matched against a user or peer.

2008-05-21 18:44 +0000 [r117519-117523]  Tilghman Lesher <tlesher@digium.com>

        * pbx/pbx_spool.c: Revert accidental commit of the last change

        * main/asterisk.c, pbx/pbx_spool.c: Strip the preamble from the
          output also when -rx is not being used (Related to issue #12702)

2008-05-21 18:28 +0000 [r117479-117514]  Russell Bryant <russell@digium.com>

        * main/asterisk.c: Don't filter the magic character in the network
          verboser. It gets filtered once it reaches the client. (related
          to issue #12702, pointed out by tilghman)

        * main/asterisk.c, pbx/pbx_gtkconsole.c: 1) Don't print the verbose
          marker in front of every message from ast_verbose() being sent to
          remote consoles. 2) Fix pbx_gtkconsole to filter out the verbose
          marker. (related to issue #12702)

        * main/asterisk.c: Don't display the verbose marker for calls to
          ast_verbose() that do not include a VERBOSE_PREFIX in front of
          the message. (closes issue #12702) Reported by: johnlange Patched
          by me

2008-05-21 16:58 +0000 [r117462]  Jeff Peeler <jpeeler@digium.com>

        * channels/chan_zap.c: Pass a pointer for the conf parameter to the
          function mkintf rather than the whole zt_chan_conf structure.

2008-05-20  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.20 released.

2008-05-14  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.20-rc3 released.

2008-05-14 12:51 +0000 [r116230]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Accept text messages even with Content-Type:
          text/plain;charset=Södermanländska

2008-05-13 23:47 +0000 [r116088]  Mark Michelson <mmichelson@digium.com>

        * main/channel.c, include/asterisk/lock.h: A change to the way
          channel locks are handled when DEBUG_CHANNEL_LOCKS is defined.
          After debugging a deadlock, it was noticed that when
          DEBUG_CHANNEL_LOCKS is enabled in menuselect, the actual origin
          of channel locks is obscured by the fact that all channel locks
          appear to happen in the function ast_channel_lock(). This code
          change redefines ast_channel_lock to be a macro which maps to
          __ast_channel_lock(), which then relays the proper file name,
          line number, and function name information to the core lock
          functions so that this information will be displayed in the case
          that there is some sort of locking error or core show locks is
          issued.

2008-05-13 21:17 +0000 [r115990-116038]  Russell Bryant <russell@digium.com>

        * channels/chan_local.c: Fix a deadlock involving channel
          autoservice and chan_local that was debugged and fixed by
          mmichelson and me. We observed a system that had a bunch of
          threads stuck in ast_autoservice_stop(). The reason these threads
          were waiting around is because this function waits to ensure that
          the channel list in the autoservice thread gets rebuilt before
          the stop() function returns. However, the autoservice thread was
          also locked, so the autoservice channel list was never getting
          rebuilt. The autoservice thread was stuck waiting for the channel
          lock on a local channel. However, the local channel was locked by
          a thread that was stuck in the autoservice stop function. It
          turned out that the issue came down to the local_queue_frame()
          function in chan_local. This function assumed that one of the
          channels passed in as an argument was locked when called.
          However, that was not always the case. There were multiple cases
          in which this channel was not locked when the function was
          called. We fixed up chan_local to indicate to this function
          whether this channel was locked or not. The previous assumption
          had caused local_queue_frame() to improperly return with the
          channel locked, where it would then never get unlocked. (closes
          issue #12584) (related to issue #12603)

        * main/autoservice.c: Fix an issue that I noticed in autoservice
          while mmichelson and I were debugging a different problem. I
          noticed that it was theoretically possible for two threads to
          attempt to start the autoservice thread at the same time. This
          change makes the process of starting the autoservice thread,
          thread-safe.

2008-05-13 20:28 +0000 [r115944]  Joshua Colp <jcolp@digium.com>

        * channels/chan_alsa.c: Use the right flag to open the audio in
          non-blocking. (closes issue #12616) Reported by:
          nicklewisdigiumuser

2008-05-13 18:36 +0000 [r115884]  Tilghman Lesher <tlesher@digium.com>

        * main/asterisk.c: If the socket dies (read returns 0=EOF), return
          immediately. (Closes issue #12637)

2008-05-12 17:51 +0000 [r115735]  Mark Michelson <mmichelson@digium.com>

        * main/utils.c: If a thread holds no locks, do not print any
          information on the thread when issuing a core show locks command.
          This will help to de-clutter output somewhat. Russell said it
          would be fine to place this improvement in the 1.4 branch, so
          that's why it's going here too.

2008-05-09 16:34 +0000 [r115579]  Joshua Colp <jcolp@digium.com>

        * configure, include/asterisk/autoconfig.h.in, configure.ac:
          Improve res_ninit and res_ndestroy autoconf logic on the Darwin
          platform.

2008-05-08 19:19 +0000 [r115545-115568]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Remove debug output.

        * /, channels/chan_iax2.c: Merged revisions 115564 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08
          May 2008) | 25 lines Fix a race condition that bbryant just found
          while doing some IAX2 testing. He was running Asterisk trunk
          running IAX2 calls through a few Asterisk boxes, however, the
          audio was extremely choppy. We looked at a packet trace and saw a
          storm of INVAL and VNAK frames being sent from one box to
          another. It turned out that what had happened was that one box
          tried to send a CONTROL frame before the 3 way handshake had
          completed. So, that frame did not include the destination call
          number, because it didn't have it yet. Part of our recent work
          for security issues included an additional check to ensure that
          frames that are supposed to include the destination call number
          have the correct one. This caused the frame to be rejected with
          an INVAL. The frame would get retransmitted for forever, rejected
          every time ... This race condition exists in all versions that
          got the security changes, in theory. However, it is really only
          likely that this would cause a problem in Asterisk trunk. There
          was a control frame being sent (SRCUPDATE) at the _very_
          beginning of the call, which does not exist in 1.2 or 1.4.
          However, I am fixing all versions that could potentially be
          affected by the introduced race condition. These changes are what
          bbryant and I came up with to fix the issue. Instead of simply
          dropping control frames that get sent before the handshake is
          complete, the code attempts to wait a little while, since in most
          cases, the handshake will complete very quickly. If it doesn't
          complete after yielding for a little while, then the frame gets
          dropped. ........

        * channels/chan_sip.c: Don't give up on attempting an outbound
          registration if we receive a 408 Timeout. (closes issue #12323)

        * contrib/scripts/postgres_cdr.sql (removed): remove
          postgres_cdr.sql, as the CDR schema is in realtime_pgsql.sql, as
          well (closes issue #9676)

        * contrib/init.d/rc.debian.asterisk: Don't exit the script if
          Asterisk is not running. (closes issue #12611)

        * main/pbx.c: Don't use a channel before checking for channel
          allocation failure. (closes issue #12609) Reported by: edantie

        * contrib/init.d/rc.debian.asterisk: Use the same method for
          executing Asterisk as the rest of the script. (closes issue
          #12611) Reported by: b_plessis

2008-05-07  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.20-rc2 released.

2008-05-07 18:17 +0000 [r115512-115517]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: Track peer references when stored in the
          sip_pvt struct as the peer related to a qualify ping or a
          subscription. This fixes some realtime related crashes. (closes
          issue #12588) (closes issue #12555)

2008-05-06 19:55 +0000 [r115418-115422]  Jason Parker <jparker@digium.com>

        * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 115421
          via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) |
          7 lines read requires an argument on some non-bash shells (closes
          issue #12593) Reported by: bkruse Patches:
          getilbc.sh_12593_v1.diff uploaded by bkruse (license 132)
          ........

        * res/res_musiconhold.c: Switch to using ast_random() rather than
          just rand(). This does not fix the bug reported, but I believe it
          is correct. (from issue #12446) Patches: bug_12446.diff uploaded
          by snuffy (license 35)

2008-05-06 19:31 +0000 [r115415]  Tilghman Lesher <tlesher@digium.com>

        * main/asterisk.c: Don't print the terminating NUL. (Closes issue
          #12589)

2008-05-06 13:54 +0000 [r115341]  Joshua Colp <jcolp@digium.com>

        * configure, configure.ac: Add in missing argument.

2008-05-05 22:50 +0000 [r115333]  Tilghman Lesher <tlesher@digium.com>

        * main/asterisk.c, main/logger.c: Separate verbose output from CLI
          output, by using a preamble. (closes issue #12402) Reported by:
          Corydon76 Patches: 20080410__no_verbose_in_rx_output.diff.txt
          uploaded by Corydon76 (license 14)
          20080501__no_verbose_in_rx_output__1.4.diff.txt uploaded by
          Corydon76 (license 14)

2008-05-05 22:10 +0000 [r115327]  Joshua Colp <jcolp@digium.com>

        * build_tools/menuselect-deps.in, configure,
          include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
          configure.ac: Make sure that either the main speex library
          contains preprocess functions or that speexdsp does. If both fail
          then speex stuff can not be built.

2008-05-05 21:41 +0000 [r115320]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Don't consider a caller "handled" until the
          caller is bridged with a queue member. There was too much of an
          opportunity for the member to hang up (either during a delay,
          announcement, or overly long agi) between the time that he
          answered the phone and the time when he actually was bridged with
          the caller. The consequence of this was that if the member hung
          up in that interval, then proper abandonment details would not be
          noted in the queue log if the caller were to hang up at any point
          after the member hangup. (closes issue #12561) Reported by:
          ablackthorn

2008-05-05 20:17 +0000 [r115308-115312]  Tilghman Lesher <tlesher@digium.com>

        * Makefile: Reverse order, such that user configs override default
          selections

        * include/asterisk/res_odbc.h: Err, the documentation on the return
          value of ast_odbc_backslash_is_escape is exactly backwards.

2008-05-05 19:49 +0000 [r115297-115304]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: Avoid putting opaque="" in Digest
          authentication. This patch came from switchvox. It fixes
          authentication with Primus in Canada, and has been in use for a
          very long time without causing problems with any other providers.
          (closes issue AST-36)

2008-05-05 03:22 +0000 [r115285]  Tilghman Lesher <tlesher@digium.com>

        * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
          contrib/init.d/rc.debian.asterisk,
          contrib/init.d/rc.mandrake.asterisk,
          contrib/init.d/rc.redhat.asterisk,
          contrib/init.d/rc.gentoo.asterisk,
          contrib/init.d/rc.slackware.asterisk: When starting Asterisk, bug
          out if Asterisk is already running. (closes issue #12525)
          Reported by: explidous Patches: 20080428__bug12525.diff.txt
          uploaded by Corydon76 (license 14) Tested by: mvanbaak

2008-05-04 02:09 +0000 [r115276-115282]  Joshua Colp <jcolp@digium.com>

        * configure, acinclude.m4: Expand the test function for GCC
          attributes so that more complex attributes are properly
          recognized.

        * include/asterisk/compiler.h: For my next trick I will make these
          work with what our autoconf header file gives us.

        * configure, acinclude.m4: Treat warnings as errors when checking
          if a GCC attribute exists. We have to do this as GCC will just
          ignore the attribute and pop up a warning, it won't actually fail
          to compile.

2008-05-02 20:25 +0000 [r115257]  Brett Bryant <bbryant@digium.com>

        * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in,
          configure.ac, CHANGES: Add new "pri show version" command to show
          the libpri version for support reasons.

2008-05-02 14:28 +0000 [r115196]  Mark Michelson <mmichelson@digium.com>

        * include/asterisk/sched.h: Clarify a comment that was, well, just
          wrong. It turns out that ignoring the way that macros expand.
          Instead, I have clarified in the comment why the macro will work
          even if the scheduler id for the task to be deleted changes
          during the execution of the macro.

2008-05-01 23:20 +0000 [r115017-115102]  Tilghman Lesher <tlesher@digium.com>

        * include/asterisk/res_odbc.h: Change the comment of deprecated to
          an actual compiler deprecation

        * main/utils.c: '#' is another reserved character for URIs that
          also needs to be escaped. (closes issue #10543) Reported by:
          blitzrage Patches: 20080418__bug10543.diff.txt uploaded by
          Corydon76 (license 14)

2008-05-01  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.20-rc1 released.

2008-04-30 16:30 +0000 [r114891]  Russell Bryant <russell@digium.com>

        * include/asterisk/dlinkedlists.h (added), channels/chan_iax2.c:
          Merge changes from team/russell/iax2_find_callno and
          iax2_find_callno_1.4 These changes address a critical performance
          issue introduced in the latest release. The fix for the latest
          security issue included a change that made Asterisk randomly
          choose call numbers to make them more difficult to guess by
          attackers. However, due to some inefficient (this is by far, an
          understatement) code, when Asterisk chose high call numbers,
          chan_iax2 became unusable after just a small number of calls. On
          a small embedded platform, it would not be able to handle a
          single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run
          more than about 16 IAX2 channels. Ouch. These changes address
          some performance issues of the find_callno() function that have
          bothered me for a very long time. On every incoming media frame,
          it iterated through every possible call number trying to find a
          matching active call. This involved a mutex lock and unlock for
          each call number checked. So, if the random call number chosen
          was 20000, then every media frame would cause 20000 locks and
          unlocks. Previously, this problem was not as obvious since
          Asterisk always chose the lowest call number it could. A second
          container for IAX2 pvt structs has been added. It is an astobj2
          hash table. When we know the remote side's call number, the pvt
          goes into the hash table with a hash value of the remote side's
          call number. Then, lookups for incoming media frames are a very
          fast hash lookup instead of an absolutely insane array traversal.
          In a quick test, I was able to get more than 3600% more IAX2
          channels on my machine with these changes.

2008-04-30 16:23 +0000 [r114890]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Don't crash on bad SIP replys. Fix created
          in Huntsville together with Mark M (putnopvut) (closes issue
          #12363) Reported by: jvandal Tested by: putnopvut, oej

2008-04-30 14:46 +0000 [r114875-114880]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/iax2.h, channels/chan_iax2.c: use the ARRAY_LEN macro
          for indexing through the iaxs/iaxsl arrays so that the size of
          the arrays can be adjusted in one place, and change the size of
          the arrays from 32768 calls to 2048 calls when LOW_MEMORY is
          defined

        * Makefile.rules: pay attention to *all* header files for
          dependency tracking, not just the local ones (inspired by r578 of
          asterisk-addons by tilghman)

2008-04-29 19:40 +0000 [r114848]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Use the MACRO_CONTEXT and MACRO_EXTEN channel
          variables instead of the channel's macrocontext and macroexten
          fields. This is needed because if macros are daisy-chained, the
          incorrect context and extension are placed on the new channel. I
          also added locking to the channel prior to accessing these
          variables as noted in trunk's janitor project file. (closes issue
          #12549) Reported by: darren1713 Patches:
          app_queue.c.macroextenpatch uploaded by darren1713 (license 116)
          (with modifications from me) Tested by: putnopvut

2008-04-29 17:08 +0000 [r114829]  Jason Parker <jparker@digium.com>

        * res/res_config_pgsql.c: Change warning message to debug, since
          there are cases where 0 results is perfectly fine.

2008-04-29 12:53 +0000 [r114823]  Kevin P. Fleming <kpfleming@digium.com>

        * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114822
          via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr
          2008) | 2 lines stop script from appending source code if run
          multiple times ........

2008-04-28 04:47 +0000 [r114708]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_voicemail.c, channels/chan_gtalk.c: When modules are
          embedded, they take on a different name, without the ".so"
          extension. Specifically check for this name, when we're checking
          if a module is loaded. (Closes issue #12534)

2008-04-27 01:26 +0000 [r114695]  Sean Bright <sean.bright@gmail.com>

        * configure, configure.ac: When we don't explicitly pass a path to
          the --with-tds configure option, we may end up finding tds.h in
          /usr/local/include instead of /usr/include. If this happens, the
          grep that looks for the version (from tdsver.h) will fail and
          we'll have some problems during the build.

2008-04-26 13:15 +0000 [r114689]  Tilghman Lesher <tlesher@digium.com>

        * contrib/scripts/vmail.cgi: Clicking forward without selecting a
          message leaves an errant .lock file. (closes issue #12528)
          Reported by: pukepail Patches: patch.diff uploaded by pukepail
          (license 431)

2008-04-25 21:54 +0000 [r114673]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Use consistent logic for checking to see if
          a call number has been chosen yet. Also, remove some redundant
          logic I recently added in a fix.

2008-04-25 19:32 +0000 [r114662]  Mark Michelson <mmichelson@digium.com>

        * apps/app_chanspy.c: Move the unlock of the spyee channel to
          outside the start_spying() function so that the channel is not
          unlocked twice when using whisper mode.

2008-04-25 15:53 +0000 [r114649]  Tilghman Lesher <tlesher@digium.com>

        * configs/zapata.conf.sample, configs/iax.conf.sample,
          configs/iaxprov.conf.sample, configs/sip.conf.sample: Reference
          documentation files that actually exist. (closes issue #12516)
          Reported by: linuxmaniac Patches: diff_rev114611.patch uploaded
          by linuxmaniac (license 472)

2008-04-24 21:35 +0000 [r114624-114632]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: Re-invite RTP during a masquerade so that,
          for instance, an AMI redirect of two channels which are natively
          bridged will preserve audio on both channels. This prevents a
          problem with Asterisk not re-inviting due to one of the channels
          having being a zombie. (closes issue #12513) Reported by:
          mneuhauser Patches:
          asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by
          mneuhauser (license 425)

        * apps/app_queue.c: Output of channel variables when
          eventwhencalled=vars was set was being truncated two characters.
          This patch corrects the problem. (closes issue #12493) Reported
          by: davidw

        * channels/chan_local.c: Resolve a deadlock in chan_local by
          releasing the channel lock temporarily. (closes issue #11712)
          Reported by: callguy Patches: 11712.patch uploaded by putnopvut
          (license 60) Tested by: acunningham

2008-04-24 19:53 +0000 [r114621]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_local.c: Ensure that when we set the accountcode,
          it actually shows up in the CDR. (Fix for AMI Originate) (Closes
          issue #12007)

2008-04-24 15:55 +0000 [r114608]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix a silly mistake in a change I made
          yesterday that caused chan_iax2 to blow up very quickly. (issue
          #12515)

2008-04-24 14:55 +0000 [r114603]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Only have one max-forwards header in
          outbound REFERs. Discovered in the Asterisk SIP Masterclass in
          Orlando. Thanks Joe!

2008-04-23 22:18 +0000 [r114597-114600]  Russell Bryant <russell@digium.com>

        * main/http.c: Improve some broken cookie parsing code. Previously,
          manager login over HTTP would only work if the mansession_id
          cookie was first. Now, the code builds a list of all of the
          cookies in the Cookie header. This fixes a problem observed by
          users of the Asterisk GUI. (closes AST-20)

        * apps/app_chanspy.c, main/http.c: Fix an issue that caused getting
          the correct next channel to not always work. Also, remove setting
          the amount of time to wait for a digit from 5 seconds back down
          to 1/10 of a second. I believe this was so the beep didn't get
          played over and over really fast, but a while back I put in
          another fix for that issue. (closes issue #12498) Reported by:
          jsmith Patches: app_chanspy_channel_walk.trunk.patch uploaded by
          jsmith (license 15)

2008-04-23 18:28 +0000 [r114594]  Jason Parker <jparker@digium.com>

        * res/res_musiconhold.c: Fix reload/unload for res_musiconhold
          module. (closes issue #11575) Reported by: sunder Patches:
          M11575_14_rev3.diff uploaded by junky (license 177)
          bug11575_trunk.diff.txt uploaded by jamesgolovich (license 176)

2008-04-23 17:55 +0000 [r114587-114591]  Russell Bryant <russell@digium.com>

        * main/manager.c, include/asterisk/manager.h: Store the manager
          session ID explicitly as 4 byte ID instead of a ulong. The
          mansession_id cookie is coded to be limited to 8 characters of
          hex, and this could break logins from 64-bit machines in some
          cases. (inspired by AST-20)

        * channels/chan_iax2.c: Fix find_callno_locked() to actually return
          the callno locked in some more cases.

2008-04-23 16:51 +0000 [r114584]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Add 502 support for both directions, not
          only one... (see r114571)

2008-04-23 14:54 +0000 [r114579]  Joshua Colp <jcolp@digium.com>

        * main/pbx.c: Instead of stopping dialplan execution when SayNumber
          attempts to say a large number that it can not print out a
          message informing the user and continue on. (closes issue #12502)
          Reported by: bcnit

2008-04-22 23:51 +0000 [r114571]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_sip.c: Treat a 502 just like a 503, when it comes
          to processing a response code

2008-04-22 22:15 +0000 [r114522-114558]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: When we receive a full frame that is
          supposed to contain our call number, ensure that it has the
          correct one. (closes issue #10078) (AST-2008-006)

        * main/rtp.c, main/channel.c, formats/format_pcm.c, main/file.c: I
          thought I was going to be able to leave 1.4 alone, but that was
          not the case. I ran into some problems with G.722 in 1.4, so I
          have merged in all of the fixes in this area that I have made in
          trunk/1.6.0, and things are happy again.

        * res/res_musiconhold.c: Trivial change to read the number of
          samples from a frame before calling ast_write()

        * res/res_features.c: After a parked call times out, allow the call
          back to the parker to time out. (closes issue #10890)

        * channels/chan_iax2.c: If the dial string passed to the call
          channel callback does not indicate an extension, then consider
          the extension on the channel before falling back to the default.
          (closes issue #12479) Reported by: darren1713 Patches:
          exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license
          116)

        * channels/chan_sip.c, include/asterisk/sched.h: Merge changes from
          team/russell/issue_9520 These changes make sure that the
          reference count for sip_peer objects properly reflects the fact
          that the peer is sitting in the scheduler for a scheduled
          callback for qualifying peers or for expiring registrations.
          Without this, it was possible for these callbacks to happen at
          the same time that the peer was being destroyed. This was
          especially likely to happen with realtime peers, and for people
          making use of the realtime prune CLI command. (closes issue
          #9520) Reported by: kryptolus Committed patch by me

2008-04-21 14:39 +0000 [r114322]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Only drop audio if we receive it without a
          progress indication. We allow other frames through such as DTMF
          because they may be needed to complete the call. (closes issue
          #12440) Reported by: aragon

2008-04-19 13:57 +0000 [r114297-114299]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_playback.c: Ensure that help text terminates with a
          newline

        * res/res_musiconhold.c: MOH usage information needs a terminating
          newline, or else "asterisk -rx 'help moh reload'" will hang.
          Reported via -dev list, fixed by me.

2008-04-18 21:48 +0000 [r114275-114284]  Russell Bryant <russell@digium.com>

        * main/manager.c: Don't destroy a manager session if poll() returns
          an error of EAGAIN.

        * Makefile: ensure directories are created before we try to install
          stuff into them

        * Makefile: SUBDIRS_INSTALL is already listed as a subtarget for
          bininstall

2008-04-18 17:44 +0000 [r114257]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_zap.c, main/callerid.c: Clearing up error messages
          so they make a bit more sense. Also removing a redundant error
          message. Issue AST-15

2008-04-18 15:24 +0000 [r114248]  Russell Bryant <russell@digium.com>

        * channels/chan_agent.c: Ensure that we don't ast_strdupa(NULL)
          (closes issue #12476) Reported by: davidw Patch by me

2008-04-18 13:33 +0000 [r114245]  Sean Bright <sean.bright@gmail.com>

        * channels/chan_sip.c: Only complete the SIP channel name once for
          'sip show channel <channel>'

2008-04-18 06:49 +0000 [r114242]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_setcallerid.c: For consistency sake, ensure that the
          values that ${CALLINGPRES} returns are valid as an input to
          SetCallingPres. (Closes issue #12472)

2008-04-17 22:15 +0000 [r114230]  Russell Bryant <russell@digium.com>

        * main/autoservice.c: Remove redundant safety net. The check for
          the autoservice channel list state accomplishes the same goal in
          a better way. (issue #12470) Reported By: atis

2008-04-17 21:03 +0000 [r114207-114226]  Mark Michelson <mmichelson@digium.com>

        * apps/app_chanspy.c: Declaration of the peer channel in this scope
          was making it so the peer variable defined in the outer scope was
          never set properly, therefore making iterating through the
          channel list always restart from the beginning. This bug would
          have affected anyone who called chanspy without specifying a
          first argument. (closes issue #12461) Reported by: stever28

        * main/frame.c, include/asterisk/dsp.h: Add prototype for
          ast_dsp_frame_freed. I'm not sure how this was compiling
          before...

        * main/dsp.c, main/frame.c, include/asterisk/frame.h: It was
          possible for a reference to a frame which was part of a freed DSP
          to still be referenced, leading to memory corruption and eventual
          crashes. This code change ensures that the dsp is freed when we
          are finished with the frame. This change is very similar to a
          change Russell made with translators back a month or so ago.
          (closes issue #11999) Reported by: destiny6628 Patches:
          11999.patch uploaded by putnopvut (license 60) Tested by:
          destiny6628, victoryure

2008-04-17 16:23 +0000 [r114204]  Russell Bryant <russell@digium.com>

        * Makefile: Fix the bininstall target to install from subdirs, as
          well. (closes issue AST-8, patch from bmd at switchvox)

2008-04-17 13:42 +0000 [r114198]  Philippe Sultan <philippe.sultan@gmail.com>

        * res/res_jabber.c: Use keepalives effectively in order diagnose
          bug #12432.

2008-04-17 12:56 +0000 [r114195]  Tilghman Lesher <tlesher@digium.com>

        * res/res_agi.c: Add special case for when the agi cannot be
          executed, to comply with the documentation that we return failure
          in that case. (closes issue #12462) Reported by: fmueller
          Patches: 20080416__bug12462.diff.txt uploaded by Corydon76
          (license 14) Tested by: fmueller

2008-04-17 10:51 +0000 [r114191]  Sean Bright <sean.bright@gmail.com>

        * apps/app_chanspy.c: Make sure we have enough room for the
          recording's filename.

2008-04-16 20:46 +0000 [r114184]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_zap.c: use the ZT_SET_DIALPARAMS ioctl properly by
          initializing the structure to all zeroes in case it contains
          fields that we don't write values into (which it does as of
          Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian

2008-04-16 19:59 +0000 [r114180]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_vpb.cc: Backport revisions for latest vpb drivers
          to 1.4 (Closes issue #12457)

2008-04-16 17:30 +0000 [r114173]  Jason Parker <jparker@digium.com>

        * channels/chan_zap.c: Fix "fallthrough" behavior here, so config
          options in a previously configured user don't override settings
          in general. (closes issue #12458) Reported by: tzafrir Patches:
          chanzap_users_sections.diff uploaded by tzafrir (license 46)

2008-04-16 14:10 +0000 [r114167]  Joshua Colp <jcolp@digium.com>

        * apps/app_meetme.c: Include the proper headers for using mkdir on
          FreeBSD. (closes issue #12430) Reported by: ys Patches:
          app_meetme.c.diff uploaded by ys (license 281)

2008-04-15 20:26 +0000 [r114148]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Handle subscribe queues in all situations...
          Thanks to festr_ on irc for telling me about this bug.

2008-04-15 17:17 +0000 [r114120-114138]  Jason Parker <jparker@digium.com>

        * contrib/scripts/autosupport: Update Digium autosupport script,
          for more useful information. (closes issue #12452) Reported by:
          angler Patches: autosupport.diff uploaded by angler (license 106)

        * apps/app_queue.c: Allow autofill to work in the general section
          of queues.conf. Additionally, don't try to (re)set options when
          they have empty values in realtime (all unset columns would have
          an empty value). (closes issue #12445) Reported by: atis Patches:
          12445-autofill.diff uploaded by qwell (license 4)

        * channels/chan_h323.c: The call_token on the pvt can occasionally
          be NULL, causing a crash. If it is NULL, we can skip this
          channel, since it can't the one we're looking for. (closes issue
          #9299) Reported by: vazir

2008-04-14 17:41 +0000 [r114106-114117]  Mark Michelson <mmichelson@digium.com>

        * main/channel.c: Increase the retry count when attempting to show
          channels. This apparently cleared an issue someone was seeing
          when attempting to show channels when the load was high. (closes
          issue #11667) Reported by: falves11 Patches: 11677.txt uploaded
          by russell (license 2) Tested by: falves11

        * apps/app_dial.c, apps/app_queue.c: If the datastore has been
          moved to another channel due to a masquerade, then freeing the
          datastore here causes an eventual double free when the new
          channel hangs up. We should only free the datastore if we were
          able to successfully remove it from the channel we are
          referencing (i.e. the datastore was not moved). (closes issue
          #12359) Reported by: pguido

        * main/channel.c: Save a local copy of the generate callback prior
          to unlocking the channel in case the generate callback goes NULL
          on us after the channel is unlocked. Thanks to Russell for
          pointing this need out to me.

2008-04-14 14:52 +0000 [r114100-114103]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: It is possible for the remote side to say
          they want T38 but not give any capabilities. (closes issue
          #12414) Reported by: MVF

        * main/rtp.c: Don't change the SSRC when a new source comes into
          play, this might happen quite often and depending on the remote
          side... they might not like this. (closes issue #12353) Reported
          by: dimas

2008-04-11 22:32 +0000 [r114083]  Terry Wilson <twilson@digium.com>

        * channels/chan_iax2.c: Several places in the code called
          find_callno() (which releases the lock on the pvt structure) and
          then immediately locked the call and did things with it.
          Unfortunately, the call can disappear between the find_callno and
          the lock, causing Bad Stuff(tm) to happen. Added
          find_callno_locked() function to return the callno withtout
          unlocking for instances that it is needed. (issue #12400)
          Reported by: ztel

2008-04-11 21:35 +0000 [r114072]  Jason Parker <jparker@digium.com>

        * main/pbx.c: It's possible that a channel can have an async goto
          on the successful execution of an application as well. Closes
          issue #12172.

2008-04-11 15:44 +0000 [r114045-114063]  Mark Michelson <mmichelson@digium.com>

        * res/res_features.c: Fix a race condition that may happen between
          a sip hangup and a "core show channel" command. This patch adds
          locking to prevent the resulting crash. (closes issue #12155)
          Reported by: tsearle Patches: show_channels_crash2.patch uploaded
          by tsearle (license 373) Tested by: tsearle

        * main/utils.c, include/asterisk/lock.h: Fix 1.4 build when
          LOW_MEMORY is enabled.

        * channels/chan_sip.c: Be sure that we're not about to set
          bridgepvt NULL prior to dereferencing it. (closes issue #11775)
          Reported by: fujin

2008-04-10 17:26 +0000 [r114035]  Jason Parker <jparker@digium.com>

        * main/file.c: Only try to prefix language if we are not using an
          absolute path (suffix it otherwise).
          en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes
          issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff
          uploaded by qwell (license 4) Tested by: kuj, qwell

2008-04-10 15:58 +0000 [r114021-114032]  Joshua Colp <jcolp@digium.com>

        * apps/app_voicemail.c: Forgot the 1.4 branch for russian language
          fix. (closes issue #12404) Reported by: IgorG Patches:
          voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20)

        * apps/app_meetme.c: Create the directory where name recordings
          will go if it does not exist. (closes issue #12311) Reported by:
          rkeene Patches: 12311-mkdir.diff uploaded by qwell (license 4)

        * channels/chan_sip.c: Don't add custom URI options if they don't
          exist OR they are empty. (closes issue #12407) Reported by:
          homesick Patches: uri_options-1.4.diff uploaded by homesick
          (license 91)

2008-04-09 20:54 +0000 [r113927]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: We need to set the persistant_route [sic]
          parameter for the sip_pvt during the initial INVITE, no matter if
          we're building the route set from an INVITE request or response.
          (closes issue #12391) Reported by: benjaminbohlmann Tested by:
          benjaminbohlmann

2008-04-09 18:57 +0000 [r113874]  Tilghman Lesher <tlesher@digium.com>

        * cdr/cdr_csv.c, configs/cdr.conf.sample: If the [csv] section does
          not exist in cdr.conf, then an unload/load sequence is needed to
          correct the problem. Track whether the load succeeded with a
          variable, so we can fix this with a simple reload event, instead.

2008-04-09 16:50 +0000 [r113784]  Joshua Colp <jcolp@digium.com>

        * channels/chan_iax2.c: If we receive an AUTHREQ from the remote
          server and we are unable to reply (for example they have a secret
          configured, but we do not) then queue a hangup frame on the
          Asterisk channel. This will cause the channel to hangup and a
          HANGUP to be sent via IAX2 to the remote side which is the proper
          thing to do in this scenario. (closes issue #12385) Reported by:
          viraptor

2008-04-09 14:40 +0000 [r113681]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: If Asterisk receives a 488 on an INVITE (not
          a reinvite), then we should not send a BYE. (closes issue #12392)
          Reported by: fnordian Patches: chan_sip.patch uploaded by
          fnordian (license 110) with small modification from me

2008-04-09 01:34 +0000 [r113596]  Terry Wilson <twilson@digium.com>

        * channels/chan_iax2.c: Initialize fr->cacheable to make valgrind
          happy

2008-04-08 19:07 +0000 [r113507]  Mark Michelson <mmichelson@digium.com>

        * apps/app_parkandannounce.c: Fix potential buffer overflow that
          could happen if more than 100 announce files were specified when
          calling ParkAndAnnounce. This overflow is not exploitable
          remotely and so there is no need for a security advisory. (closes
          issue #12386) Reported by: davidw

2008-04-08 18:48 +0000 [r113402-113504]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Add a little more that is required for
          previously added devices.

        * channels/chan_skinny.c: Add support for several new(ish) devices
          - most notably, 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver
          for providing me the required information.

        * main/asterisk.c: Work around some silliness caused by
          sys/capability.h - this should fix compile errors a number of
          users have been experiencing.

2008-04-08 16:51 +0000 [r113348-113399]  Tilghman Lesher <tlesher@digium.com>

        * contrib/scripts/astgenkey.8: Add security note on astgenkey's
          manpage. (closes issue #12373) Reported by: lmamane Patches:
          20080406__bug12373.diff.txt uploaded by Corydon76 (license 14)

        * channels/chan_sip.c: Move check for still-bridged channels out a
          little further, to avoid possible deadlocks. (Closes issue
          #12252) Reported by: callguy Patches: 20080319__bug12252.diff.txt
          uploaded by Corydon76 (license 14) Tested by: callguy

2008-04-08 15:03 +0000 [r113296]  Joshua Colp <jcolp@digium.com>

        * include/asterisk/slinfactory.h, main/slinfactory.c,
          main/audiohook.c: If audio suddenly gets fed into one side of a
          channel after a lapse of frames flush the other factory so that
          old audio does not remain in the factory causing the sync code to
          not execute. (closes issue #12296) Reported by: jvandal

2008-04-07 21:34 +0000 [r113240]  Jeff Peeler <jpeeler@digium.com>

        * channels/chan_sip.c: (closes issue #12362) [redo of 113012] This
          fixes a for loop (in realtime_peer) to check all the
          ast_variables the loop was intending to test rather than just the
          first one. The change exposed the problem of calling memcpy on a
          NULL pointer, in this case the passed in sockaddr_in struct which
          is now checked.

2008-04-07 18:00 +0000 [r113118]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c, configs/skinny.conf.sample: Allow
          playback with noanswer (and add earlyrtp option). (closes issue
          #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn
          (license 30) Tested by: pj, qwell, DEA, wedhorn

2008-04-07 17:51 +0000 [r113117]  Tilghman Lesher <tlesher@digium.com>

        * funcs/func_strings.c: Force ast_mktime() to check for DST, since
          strptime(3) does not. (Closes issue #12374)

2008-04-07 16:08 +0000 [r113065]  Mark Michelson <mmichelson@digium.com>

        * main/channel.c: This fix prevents a deadlock that was experienced
          in chan_local. There was deadlock prevention in place in
          chan_local, but it would not work in a specific case because the
          channel was recursively locked. By unlocking the channel prior to
          calling the generator's generate callback in
          ast_read_generator_actions(), we prevent the recursive locking,
          and therefore the deadlock. (closes issue #12307) Reported by:
          callguy Patches: 12307.patch uploaded by putnopvut (license 60)
          Tested by: callguy

2008-04-07 15:16 +0000 [r113012]  Jeff Peeler <jpeeler@digium.com>

        * channels/chan_sip.c: (closes issue #12362) (closes issue #12372)
          Reported by: vinsik Tested by: tecnoxarxa This one line change
          makes an if inside a for loop (in realtime_peer) check all the
          ast_variables the loop was intending to test rather than just the
          first one.

2008-04-04 19:26 +0000 [r112766-112820]  Philippe Sultan <philippe.sultan@gmail.com>

        * channels/chan_gtalk.c: Free newly allocated channel before
          returning

        * channels/chan_gtalk.c: Prevent call connections when codecs don't
          match. (closes issue #10604) Reported by: keepitcool Patches:
          branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested
          by: phsultan

2008-04-04 00:52 +0000 [r112709-112711]  Joshua Colp <jcolp@digium.com>

        * main/Makefile: Pass in the path to Zaptel for systems that
          install Zaptel headers in a separate location.

        * main/asterisk.c: One thing at a time... let's get 1.4 building.

2008-04-03 23:57 +0000 [r112689]  Dwayne M. Hubbard <dhubbard@digium.com>

        * main/asterisk.c: add a Zaptel timer check to verify the timer is
          responding when Zaptel support is compiled into Asterisk and
          Zaptel drivers are loaded. This will help people not waste their
          valuable time debugging side effects.

2008-04-03 14:32 +0000 [r112393-112599]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_zap.c: Fix the testing of the "res" variable so
          that it is more logically correct and makes the correct warning
          and debug messages print. (closes issue #12361) Reported by:
          one47 Patches: chan_zap_deferred_digit.patch uploaded by one47
          (license 23)

        * main/manager.c: Fix a race condition in the manager. It is
          possible that a new manager event could be appended during a
          brief time when the manager is not waiting for input. If an event
          comes during this period, we need to set an indicator that there
          is an event pending so that the manager doesn't attempt to wait
          forever for an event that already happened. (closes issue #12354)
          Reported by: bamby Patches: manager_race_condition.diff uploaded
          by bamby (license 430) (comments added by me)

        * apps/app_queue.c: Ensure that there is no timeout if none is
          specified. (closes issue #12349) Reported by: johnlange

2008-04-01  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.19 released.

2008-03-28  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.19-rc4 released.

2008-03-28 16:19 +0000 [r111658]  Jason Parker <jparker@digium.com>

        * formats/format_wav_gsm.c: The file size of WAV49 does not need to
          be an even number. (closes issue #12128) Reported by: mdu113
          Patches: 12128-noevenlength.diff uploaded by qwell (license 4)
          Tested by: qwell, mdu113

2008-03-28 14:35 +0000 [r111442-111605]  Tilghman Lesher <tlesher@digium.com>

        * doc/valgrind.txt: Update debugging text, since Valgrind
          eliminated the --log-file-exactly option. (Closes issue #12320)

        * main/acl.c: For FreeBSD, at least, the ifa_addr element could be
          NULL. (closes issue #12300) Reported by: festr Patches:
          acl.c.patch uploaded by festr (license 443)

2008-03-27 13:03 +0000 [r111341-111391]  Steve Murphy <murf@digium.com>

        * apps/app_playback.c, main/pbx.c: These small documentation
          updates made in response to a query in asterisk-users, where a
          user was using Playback, but needed the features of Background,
          and had no idea that Background existed, or that it might provide
          the features he needed. I thought the best way to avert these
          kinds of queries was to provide "See Also" references in all
          three of "Background", "Playback", "WaitExten". Perhaps a project
          to do this with all related apps is in order.

        * pbx/pbx_ael.c, include/asterisk/ael_structs.h: (closes issue
          #12302) Reported by: pj Tested by: murf These changes will set a
          channel variable ~~EXTEN~~ just before generating code for a
          switch, with the value of ${EXTEN}. The exten is marked as having
          a switch, and ever after that, till the end of the exten, we
          substitute any ${EXTEN} with ${~~EXTEN~~} instead in application
          arguments; (and the ${EXTEN: also). The reason for this, is that
          because switches are coded using separate extensions to provide
          pattern matching, and jumping to/from these switch extensions
          messes up the ${EXTEN} value, which blows the minds of users.

2008-03-27 00:25 +0000 [r111245-111280]  Jason Parker <jparker@digium.com>

        * main/frame.c: Put this flag back so we don't change the API.

        * main/frame.c: Remove excessive smoother optimization that was
          causing audio glitches (small "pops") after (about 200ms later)
          an "incorrectly" sized frame was received. While it would be very
          nice to keep this as optimized as possible, it makes no sense for
          the smoother to be dropping random bits of audio like this. Isn't
          that the whole point of a smoother? Closes issue #12093.

2008-03-26 19:55 +0000 [r111129]  Joshua Colp <jcolp@digium.com>

        * contrib/scripts/autosupport: Update autosupport script. (closes
          issue #12310) Reported by: angler Patches: autosupport.diff
          uploaded by angler (license 106)

2008-03-26 19:51 +0000 [r111126]  Kevin P. Fleming <kpfleming@digium.com>

        * /, UPGRADE.txt: Merged revisions 111125 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar
          2008) | 2 lines update UPGRADE notes to document usage of the
          script ........

2008-03-26 19:37 +0000 [r111049-111121]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: This code change is made just for
          clarification. It does exactly the same thing as before. It just
          doesn't look as wrong.

        * apps/app_voicemail.c: Add a lock to the vm_state structure and
          use the lock around mail_open calls to prevent concurrent access
          of the same mailstream. This, along with trunk's ability to
          configure TCP timeouts for IMAP storage will help to prevent
          crashes and hangs when using voicemail with IMAP storage. (closes
          issue #10487) Reported by: ewilhelmsen

2008-03-26 19:06 +0000 [r111024]  Kevin P. Fleming <kpfleming@digium.com>

        * codecs/ilbc, /, contrib/scripts/get_ilbc_source.sh (added):
          Merged revisions 111019 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar
          2008) | 2 lines add a script to make getting the iLBC source code
          simple for end users ........

2008-03-26 19:04 +0000 [r111014-111020]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: If we are requested to authenticate a
          reinvite make sure that it contains T38 SDP if need be. (closes
          issue #11995) Reported by: fall

        * channels/chan_iax2.c: Make sure that full video frames are sent
          whenever the 15 bit timestamp rolls over. (closes issue #11923)
          Reported by: mihai Patches: asterisk-fullvideo.patch uploaded by
          mihai (license 94)

2008-03-26 17:43 +0000 [r110880-110962]  Kevin P. Fleming <kpfleming@digium.com>

        * UPGRADE.txt: add note that the user will need to enable
          codec_ilbc to get it to build

        * codecs/ilbc/StateConstructW.h (removed),
          codecs/ilbc/libilbc.vcproj (removed), codecs/ilbc/packing.h
          (removed), codecs/ilbc/getCBvec.c (removed),
          codecs/ilbc/LPCdecode.c (removed), codecs/ilbc/enhancer.c
          (removed), codecs/ilbc/lsf.c (removed), codecs/ilbc/iLBC_encode.c
          (removed), codecs/ilbc/getCBvec.h (removed),
          codecs/ilbc/LPCdecode.h (removed), codecs/ilbc/enhancer.h
          (removed), codecs/ilbc/FrameClassify.c (removed),
          codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h (removed),
          codecs/ilbc/iLBC_encode.h (removed), codecs/ilbc/FrameClassify.h
          (removed), codecs/ilbc/helpfun.c (removed), codecs/ilbc/doCPLC.c
          (removed), codecs/ilbc/anaFilter.c (removed),
          codecs/ilbc/helpfun.h (removed), codecs/ilbc/createCB.c
          (removed), codecs/ilbc/doCPLC.h (removed),
          codecs/ilbc/anaFilter.h (removed), UPGRADE.txt,
          codecs/ilbc/iLBC_decode.c (removed), codecs/ilbc/constants.c
          (removed), codecs/ilbc/createCB.h (removed), CHANGES,
          codecs/ilbc/iLBC_decode.h (removed), codecs/ilbc/constants.h
          (removed), codecs/Makefile, codecs/ilbc/iCBSearch.c (removed),
          codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed),
          codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c
          (removed), codecs/ilbc/iCBSearch.h (removed),
          codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed),
          codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c
          (removed), codecs/ilbc/hpOutput.h (removed),
          codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c,
          codecs/ilbc/LPCencode.h (removed), codecs/ilbc/StateSearchW.h
          (removed), codecs/ilbc/iCBConstruct.c (removed),
          codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h
          (removed), codecs/ilbc/syntFilter.h (removed),
          codecs/ilbc/StateConstructW.c (removed), codecs/ilbc/packing.c
          (removed): Merged revisions 110869 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar
          2008) | 2 lines due to licensing restrictions, we cannot
          distribute the source code for iLBC encoding and decoding... so
          remove it, and add instructions on how the user can obtain it
          themselves ........

2008-03-25 22:51 +0000 [r110779]  Jason Parker <jparker@digium.com>

        * cdr/cdr_custom.c: Make file access in cdr_custom similar to
          cdr_csv. Fixes issue #12268. Patch borrowed from r82344

2008-03-25 20:03 +0000 [r110727]  Jeff Peeler <jpeeler@digium.com>

        * channels/chan_sip.c: This one line change makes an if inside a
          for loop (in realtime_peer) check all the ast_variables the loop
          was intending to test rather than just the first one.

2008-03-25 15:40 +0000 [r110635]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: When reverting a commit, I accidentally left
          in this bit which was an experiment to see what would happen. It
          passed the compile test, and I didn't notice I had left this
          change in too. So this is a revert of a revert...sort of.

2008-03-25 14:37 +0000 [r110628]  Joshua Colp <jcolp@digium.com>

        * include/asterisk/options.h, main/asterisk.c, Makefile,
          main/app.c: Add an option (transmit_silence) which transmits
          silence during both Record() and DTMF generation. The reason this
          is an option is that in order to transmit silence we have to
          setup a translation path. This may not be needed/wanted in all
          cases. (closes issue #10058) Reported by: tracinet

2008-03-24 19:17 +0000 [r110618]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: This is a revert for revision 108288. The
          reason is that that revision was not for an actual bug fix per
          se, and so it really should not have been in 1.4 in the first
          place. Plus, people who compile with DO_CRASH are more likely to
          encounter a crash due to this change. While I think the usage of
          DO_CRASH in ast_sched_del is a bit absurd, this sort of change is
          beyond the scope of 1.4 and should be done instead in a developer
          branch based on trunk so that all scheduler functions are fixed
          at once. I also am reverting the change to trunk and 1.6 since
          they also suffer from the DO_CRASH potential. (closes issue
          #12272) Reported by: qq12345

2008-03-24 17:34 +0000 [r110614]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Turn a NOTICE into a DEBUG message.

2008-03-21 14:32 +0000 [r110474]  Jason Parker <jparker@digium.com>

        * codecs/gsm/Makefile: Don't attempt to do optimizations of gsm on
          mips platforms either. (closes issue #12270) Reported by:
          zandbelt Patches: 026-gsm-mips.patch uploaded by zandbelt
          (license 33)

2008-03-20 23:13 +0000 [r110163-110395]  Russell Bryant <russell@digium.com>

        * main/autoservice.c: Shorten the ast_waitfor() timeout from 500 ms
          to 50 ms in the autoservice thread. This really should not make a
          difference except in very rare cases. That case would be that all
          of the channels in autoservice are not generating any frames. In
          that case, this change reduces the potential amount of time that
          a thread waits in ast_autoservice_stop() for the autoservice
          thread to wrap back around to the beginning of its loop. (closes
          issue #12266, reported by dimas)

        * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
          110335 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008)
          | 6 lines Fix some very broken code that was introduced in 1.2.26
          as a part of the security fix. The dnsmgr is not appropriate
          here. The dnsmgr takes a pointer to an address structure that a
          background thread continuously updates. However, in these cases,
          a stack variable was passed. That means that the dnsmgr thread
          would be continuously writing to bogus memory. ........

        * apps/app_meetme.c: Fix a bug where when calls on the trunk side
          hang up while on hold, the state is not properly reflected.
          (closes issue #11990, reported by anakaoka, patched by me)

2008-03-19 20:33 +0000 [r110083]  Mark Michelson <mmichelson@digium.com>

        * apps/app_chanspy.c: Add a missing unlock in the case that memory
          allocation fails in app_chanspy. Thanks to Russell for confirming
          that this was an issue.

2008-03-19 19:11 +0000 [r110019-110035]  Joshua Colp <jcolp@digium.com>

        * res/res_musiconhold.c: Add sanity checking for position resuming.
          We *have* to make sure that the position does not exceed the
          total number of files present, and we have to make sure that the
          position's filename is the same as previous. These values can
          change if a music class is reloaded and give unpredictable
          behavior. (closes issue #11663) Reported by: junky

        * main/rtp.c: Make sure that the mark bit does not incorrectly
          cause video frame timestamps to be calculated as if they are
          audio frames. (closes issue #11429) Reported by: sperreault
          Patches: 11429-frametype.diff uploaded by qwell (license 4)

2008-03-19 17:12 +0000 [r109973]  Jason Parker <jparker@digium.com>

        * Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml
          (added): People report bugs about Asterisk crashing with DO_CRASH
          enabled was getting a little silly... Now we only show certain
          cflags when you run configure with --enable-dev-mode
          (corresponding menuselect change to follow)

2008-03-19 15:41 +0000 [r109908]  Steve Murphy <murf@digium.com>

        * main/config.c: (closes issue #11442) Reported by: tzafrir
          Patches: 11442.patch uploaded by murf (license 17) Tested by:
          murf I didn't give tzafrir very much time to test this, but if he
          does still have remaining issues, he is welcome to re-open this
          bug, and we'll do what is called for. I reproduced the problem,
          and tested the fix, so I hope I am not jumping by just going
          ahead and committing the fix. The problem was with what file_save
          does with templates; firstly, it tended to print out multiple
          options: [my_category](!)(templateref) instead of
          [my_category](!,templateref) which is fixed by this patch.
          Nextly, the code to suppress output of duplicate declarations
          that would occur because the reader copies inherited declarations
          down the hierarchy, was not working. Thus: [master-template](!)
          mastervar = bar [template](!,master-template) tvar = value
          [cat](template) catvar = val would be rewritten as: ;! ;!
          Automatically generated configuration file ;! Filename:
          experiment.conf (/etc/asterisk/experiment.conf) ;! Generator:
          Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;!
          [master-template](!) mastervar = bar
          [template](!,master-template) mastervar = bar tvar = value
          [cat](template) mastervar = bar tvar = value catvar = val This
          has been fixed. Since the config reader 'explodes' inherited vars
          into the category, users may, in certain circumstances, see
          output different from what they originally entered, but it should
          be both correct and equivalent.

2008-03-19 04:06 +0000 [r109763-109838]  Russell Bryant <russell@digium.com>

        * main/utils.c: Tweak spacing in a recent change because I'm very
          picky.

        * apps/app_chanspy.c: Fix one place where the chanspy datastore
          isn't removed from a channel. (issue #12243, reported by atis,
          patch by me)

2008-03-18 20:52 +0000 [r109713]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: This patch makes it so that all queue member
          status changes are handled through device state code. This
          removes several problems people were seeing where their queue
          members would get into an "unknown" state. Huge props go to atis
          on this one since he was the one who found the code section that
          was causing the problem and proposed the solution. I just wrote
          what he suggested :) (closes issue #12127) Reported by: atis
          Patches: 12127v3.patch uploaded by putnopvut (license 60) Tested
          by: atis, jvandal

2008-03-18 19:23 +0000 [r109648]  Jason Parker <jparker@digium.com>

        * codecs/log2comp.h: Allow codecs that use log2comp (g726) to
          compile correctly on x86 with gcc4 optimizations. (closes issue
          #12253) Reported by: fossil Patches: log2comp.patch uploaded by
          fossil (license 140)

2008-03-18 17:58 +0000 [r109575]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_agent.c: Make sure an agent doesn't try to send
          dtmf to a NULL channel closes issue #12242 Reported by Yourname

2008-03-18  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.19-rc3 released.

2008-03-18 16:25 +0000 [r109482]  Terry Wilson <twilson@digium.com>

        * include/asterisk/astobj.h: Fix character string being treated ad
          format string

2008-03-18 15:10 +0000 [r109393]  Jason Parker <jparker@digium.com>

        * /, channels/chan_sip.c: Merged revisions 109391 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r109391 | qwell | 2008-03-18 10:08:41 -0500 (Tue, 18 Mar 2008) |
          3 lines Do not return with a successful authentication if the
          From header ends up empty. (AST-2008-003) ........

2008-03-18 14:58 +0000 [r109386]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c, channels/chan_sip.c: Put a maximum limit on the
          number of payloads accepted, and also make sure a given payload
          does not exceed our maximum value. (AST-2008-002)

2008-03-18 06:37 +0000 [r109309]  Steve Murphy <murf@digium.com>

        * pbx/ael/ael-test/ael-ntest23 (added),
          pbx/ael/ael-test/ael-ntest23/t1/a.ael (added),
          pbx/ael/ael-test/ael-ntest23/t1/b.ael (added),
          pbx/ael/ael-test/ael-ntest23/t1/c.ael (added),
          pbx/ael/ael-test/ael-ntest23/t2/d.ael (added),
          pbx/ael/ael-test/ael-ntest23/t2/e.ael (added),
          pbx/ael/ael-test/ael-ntest23/t2/f.ael (added),
          pbx/ael/ael-test/ref.ael-ntest23 (added), pbx/ael/ael_lex.c,
          pbx/ael/ael-test/ael-ntest23/t3/g.ael (added),
          pbx/ael/ael-test/ael-ntest23/t3/h.ael (added),
          pbx/ael/ael-test/ael-ntest23/t3/i.ael (added), pbx/ael/ael.flex,
          pbx/ael/ael-test/ael-ntest23/t3/j.ael (added),
          pbx/ael/ael-test/ael-ntest23/qq.ael (added),
          pbx/ael/ael-test/ael-ntest23/t1 (added),
          pbx/ael/ael-test/ael-ntest23/t2 (added),
          pbx/ael/ael-test/ael-ntest23/t3 (added),
          pbx/ael/ael-test/ael-ntest23/extensions.ael (added): (closes
          issue #11903) Reported by: atis Many thanks to atis for spotting
          this problem and reporting it. The fix was to straighten out how
          items are placed on and removed from the file stack. Regressions
          as well as the provided test case helped to straighten out all
          code paths. valgrind was used to make sure all memory allocated
          was freed. Sorry for not solving this earlier. I got distracted.
          Added the ntest23 regression test, which is mainly a copy of
          ntest22, but with a few juicy errors thrown in, to replicate the
          kind of error that atis spotted.

2008-03-17 22:05 +0000 [r109226]  Mark Michelson <mmichelson@digium.com>

        * main/utils.c: Fix a logic flaw in the code that stores lock info
          which is displayed via the "core show locks" command. The idea
          behind this section of code was to remove the previous lock from
          the list if it was a trylock that had failed. Unfortunately,
          instead of checking the status of the previous lock, we were
          referencing the index immediately following the previous lock in
          the lock_info->locks array. The result of this problem, under the
          right circumstances, was that the lock which we currently in the
          process of attempting to acquire could "overwrite" the previous
          lock which was acquired. While this does not in any way affect
          typical operation, it *could* lead to misleading "core show
          locks" output.

2008-03-17 17:55 +0000 [r109171]  Michiel van Baak <michiel@vanbaak.info>

        * channels/chan_skinny.c: Update the directory of placed calls on
          skinny phones when dialing a channel that does not provide
          progress (analog ZAP lines) The phone does handle the double
          update on calls to channels that do provide progress and wont
          insert duplicate items (closes issue #12239) Reported by: DEA
          Patches: chan_skinny-call-log.txt uploaded by DEA (license 3)

2008-03-17 16:24 +0000 [r109107]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: 200 OKs in response to a reinvite need to be
          sent reliably. If the remote side does not receive one the dialog
          will be torn down. (closes issue #12208) Reported by: atrash

2008-03-17 15:15 +0000 [r109057]  Jason Parker <jparker@digium.com>

        * main/file.c: Backport revision 106439 from trunk. I didn't
          realize this was broken in 1.4 as well. Closes issue #12222.

2008-03-17 14:18 +0000 [r109012]  Mark Michelson <mmichelson@digium.com>

        * apps/app_chanspy.c: Make sure that we release the lock on the
          spyee channel if the spyee or spy has hung up (closes issue
          #12232) Reported by: atis

2008-03-16 21:47 +0000 [r108961]  Michiel van Baak <michiel@vanbaak.info>

        * main/dial.c: add missing break to case AST_CONTROL_SRCUPDATE
          (closes issue #12228) Reported by: andrew Patches: SRC.patch
          uploaded by andrew (license 240)

2008-03-14 20:09 +0000 [r108792-108796]  Russell Bryant <russell@digium.com>

        * channels/chan_oss.c: Fix a channel name issue. chan_oss registers
          the "Console" channel type, but it created channels with an "OSS"
          prefix. (closes issue #12194, reported by davidw, patched by me)

        * contrib/init.d/rc.suse.asterisk: Update the SuSE init script to
          start networking before asterisk, as well. (closes issue #12200,
          reported by and change suggested by reinerotto)

2008-03-14 16:44 +0000 [r108737]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: Fix a race condition in the SIP packet
          scheduler which could cause a crash. chan_sip uses the scheduler
          API in order to schedule retransmission of reliable packets (such
          as INVITES). If a retransmission of a packet is occurring, then
          the packet is removed from the scheduler and retrans_pkt is
          called. Meanwhile, if a response is received from the packet as
          previously transmitted, then when we ACK the response, we will
          remove the packet from the scheduler and free the packet. The
          problem is that both the ACK function and retrans_pkt attempt to
          acquire the same lock at the beginning of the function call. This
          means that if the ACK function acquires the lock first, then it
          will free the packet which retrans_pkt is about to read from and
          write to. The result is a crash. The solution: 1. If the ACK
          function fails to remove the packet from the scheduler and the
          retransmit id of the packet is not -1 (meaning that we have not
          reached the maximum number of retransmissions) then release the
          lock and yield so that retrans_pkt may acquire the lock and
          operate. 2. Make absolutely certain that the ACK function does
          not recursively lock the lock in question. If it does, then
          releasing the lock will do no good, since retrans_pkt will still
          be unable to acquire the lock. (closes issue #12098) Reported by:
          wegbert (closes issue #12089) Reported by: PTorres Patches:
          12098-putnopvutv3.patch uploaded by putnopvut (license 60) Tested
          by: jvandal

2008-03-14 14:29 +0000 [r108682]  Jason Parker <jparker@digium.com>

        * res/res_musiconhold.c: Fix a potential segfault if chan (or
          chan->music_state) is NULL. Closes issue #12210, credit to
          edantie for pointing this out.

2008-03-13 21:38 +0000 [r108469-108583]  Russell Bryant <russell@digium.com>

        * apps/app_chanspy.c, main/channel.c, include/asterisk/channel.h:
          Fix another issue that was causing crashes in chanspy. This
          introduces a new datastore callback, called chan_fixup(). The
          concept is exactly like the fixup callback that is used in the
          channel technology interface. This callback gets called when the
          owning channel changes due to a masquerade. Before this was
          introduced, if a masquerade happened on a channel being spyed on,
          the channel pointer in the datastore became invalid. (closes
          issue #12187) (reported by, and lots of testing from atis) (props
          to file for the help with ideas)

        * channels/chan_sip.c: Make a tweak that gets the LEDs on polycom
          phones to blink when an extension that has been subscribed to
          goes on hold. Otherwise, they just stay on like it does when an
          extension is in use. (closes issue #11263) Reported by: russell
          Patches: notify_hold.rev1.txt uploaded by russell (license 2)
          Tested by: russell

        * apps/app_followme.c: Fix a couple uses of sprintf. The second one
          could actually cause an overflow of a stack buffer. It's not a
          security issue though, it only depends on your configuration.

2008-03-12 21:53 +0000 [r108227-108288]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: Change AST_SCHED_DEL use to ast_sched_del
          for autocongestion in chan_sip. The scheduler callback will
          always return 0. This means that this id is never rescheduled, so
          it makes no sense to loop trying to delete the id from the
          scheduler queue. If we fail to remove the item from the queue
          once, it will fail every single time. (Yes I realize that in this
          case, the macro would exit early because the id is set to -1 in
          the callback, but it still makes no sense to use that macro in
          favor of calling ast_sched_del once and being done with it) This
          is the first of potentially several such fixes.

        * include/asterisk/sched.h: Added a large comment before the
          AST_SCHED_DEL macro to explain its purpose as well as when it is
          appropriate and when it is not appropriate to use it. I also
          removed the part of the debug message that mentions that this is
          probably a bug because there are some perfectly legitimate places
          where ast_sched_del may fail to delete an entry (e.g. when the
          scheduler callback manually reschedules with a new id instead of
          returning non-zero to tell the scheduler to reschedule with the
          same idea). I also raised the debug level of the debug message in
          AST_SCHED_DEL since it seems like it could come up quite
          frequently since the macro is probably being used in several
          places where it shouldn't be. Also removed the redundant line,
          file, and function information since that is provided by ast_log.

2008-03-12 19:57 +0000 [r108135]  Russell Bryant <russell@digium.com>

        * apps/app_chanspy.c, main/channel.c: (closes issue #12187,
          reported by atis, fixed by me after some brainstorming on the
          issue with mmichelson) - Update copyright info on app_chanspy. -
          Fix a race condition that caused app_chanspy to crash. The issue
          was that the chanspy datastore magic that was used to ensure that
          spyee channels did not disappear out from under the code did not
          completely solve the problem. It was actually possible for
          chanspy to acquire a channel reference out of its datastore to a
          channel that was in the middle of being destroyed. That was
          because datastore destruction in ast_channel_free() was done near
          the end. So, this left the code in app_chanspy accessing a
          channel that was partially, or completely invalid because it was
          in the process of being free'd by another thread. The following
          sort of shows the code path where the race occurred:
          =============================================================================
          Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy)
          --------------------------------------||-------------------------------------
          ast_channel_free() || - remove channel from channel list || -
          lock/unlock the channel to ensure || that no references retrieved
          from || the channel list exist. ||
          --------------------------------------||-------------------------------------
          || channel_spy() - destroy some channel data || - Lock chanspy
          datastore || - Retrieve reference to channel || - lock channel ||
          - Unlock chanspy datastore
          --------------------------------------||-------------------------------------
          - destroy channel datastores || - call chanspy datastore d'tor ||
          which NULL's out the ds' || - Operate on the channel ...
          reference to the channel || || - free the channel || || || -
          unlock the channel
          --------------------------------------||-------------------------------------
          =============================================================================

2008-03-12 19:16 +0000 [r108086]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_sip.c: if we receive an INVITE with a
          Content-Length that is not a valid number, or is zero, then don't
          process the rest of the message body looking for an SDP closes
          issue #11475 Reported by: andrebarbosa

2008-03-12 18:26 +0000 [r108083]  Joshua Colp <jcolp@digium.com>

        * apps/app_mixmonitor.c, include/asterisk/audiohook.h,
          main/audiohook.c: Add a trigger mode that triggers on both read
          and write. The actual function that returns the combined audio
          frame though will wait until both sides have fed in audio, or
          until one side stops (such as the case when you call Wait).
          (closes issue #11945) Reported by: xheliox

2008-03-12 16:59 +0000 [r108031]  Russell Bryant <russell@digium.com>

        * main/channel.c: Destroy the channel lock after the channel
          datastores. (inspired by issue #12187)

2008-03-12 01:52 +0000 [r107877]  Tilghman Lesher <tlesher@digium.com>

        * contrib/scripts/iax-friends.sql, contrib/scripts/sip-friends.sql:
          Document all of the possible realtime fields

2008-03-11 23:37 +0000 [r107714-107826]  Jason Parker <jparker@digium.com>

        * doc/voicemail_odbc_postgresql.txt: Update documentation for pgsql
          ODBC voicemail. (closes issue #12186) Reported by: jsmith
          Patches: vm_pgsql_doc_update.patch uploaded by jsmith (license
          15)

        * channels/chan_gtalk.c: Copy voicemail dependency logic for
          res_adsi to chan_gtalk (for jabber). (closes issue #12014)
          Reported by: junky

2008-03-11 20:48 +0000 [r107713]  Kevin P. Fleming <kpfleming@digium.com>

        * Makefile.rules, channels/Makefile: get chan_vpb to build properly
          in dev mode

2008-03-11 20:47 +0000 [r107712]  Jason Parker <jparker@digium.com>

        * apps/app_voicemail.c: Add a newline on a log

2008-03-11 19:20 +0000 [r107582-107646]  Joshua Colp <jcolp@digium.com>

        * res/res_features.c: Make sure the visible indication is on the
          right channel so when the masquerade happens the proper
          indication is enacted. (closes issue #11707) Reported by: iam

        * apps/app_meetme.c: Add an additional check for setting conference
          parameter when using the marked user options. It was possible for
          it to return to a no listen/no talk state if a masquerade
          happened. (closes issue #12136) Reported by: aragon

        * apps/app_exec.c: Fix a minor spelling error. (closes issue
          #12183) Reported by: darrylc

2008-03-11  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.19-rc2 released.

2008-03-11 15:18 +0000 [r107352-107472]  Kevin P. Fleming <kpfleming@digium.com>

        * apps/app_rpt.c: backport a fix from trunk

        * channels/misdn/isdn_lib.c, codecs/Makefile,
          channels/chan_misdn.c: fix various other problems found by gcc
          4.3

        * configure, include/asterisk/autoconfig.h.in, configure.ac,
          apps/app_sms.c: stop checking for mktime() in the configure
          script... we don't use it, and the test is buggy under gcc 4.3

        * configure, main/Makefile, configure.ac, makeopts.in: check for
          compiler support for -fno-strict-overflow before using it (tested
          with Debian's gcc 4.3, 4.1 and 3.4) (closes issue #12179)
          Reported by: Netview

        * configure, configure.ac: fix small bug in IMAP toolkit testing

        * main/udptl.c, utils/Makefile, main/Makefile,
          main/editline/readline.c, pbx/Makefile: fix up various compiler
          warnings found with gcc-4.3: - the output of flex includes a
          static function called 'input' that is not used, so for the
          moment we'll stop having the compiler tell us about unused
          variables in the flex source files (a better fix would be to
          improve our flex post-processing to remove the unused function) -
          main/stdtime/localtime.c makes assumptions about signed integer
          overflow, and gcc-4.3's improved optimizer tries to take
          advantage of handling potential overflow conditions at compile
          time; for now, suppress these optimizations until we can fiure
          out if the code needs improvement - main/udptl.c has some
          references to uninitialized variables; in one case there was no
          bug, but in the other it was certainly possibly for unexpected
          behavior to occur - main/editline/readline.c had an unused
          variable

2008-03-11 00:59 +0000 [r107290]  Terry Wilson <twilson@digium.com>

        * channels/chan_sip.c: If we fail to alloc a channel, we should
          re-lock the pvt structure before returning.

2008-03-10 21:32 +0000 [r107230]  Tilghman Lesher <tlesher@digium.com>

        * main/pbx.c: Use non-global storage for eswitch

2008-03-10 20:27 +0000 [r107173]  Jason Parker <jparker@digium.com>

        * channels/chan_zap.c: Make sure to reenable echo can after a
          "failed" (canceled, etc) three-way call. (closes issue #11335)
          Reported by: rebuild

2008-03-10 20:17 +0000 [r107099-107161]  Russell Bryant <russell@digium.com>

        * main/pbx.c: Fix another bug specifically related to asynchronous
          call origination. Once the PBX is started on the channel using
          ast_pbx_start(), then the ownership of the channel has been
          passed on to another thread. We can no longer access it in this
          code. If the channel gets hung up very quickly, it is possible
          that we could access a channel that has been free'd. (inspired by
          BE-386)

        * main/pbx.c: Fix some bugs related to originating calls. If the
          code failed to start a PBX on the channel (such as if you set a
          call limit based on the system's load average), then there were
          cases where a channel that has already been free'd using
          ast_hangup() got accessed. This caused weird memory corruption
          and crashes to occur. (fixes issue BE-386) (much debugging credit
          goes to twilson, final patch written by me)

        * main/channel.c: Resolve a compiler warning.

        * main/channel.c: Fix a race condition where the generator can go
          away (closes issue #12175, reported by edantie, patched by me)

2008-03-10 14:33 +0000 [r107016]  Joshua Colp <jcolp@digium.com>

        * apps/app_dial.c, main/cdr.c, include/asterisk/cdr.h: Move where
          unanswered CDRs are dropped to the CDR core, not everything uses
          app_dial. (closes issue #11516) Reported by: ys Patches:
          branch_1.4_cdr.diff uploaded by ys (license 281) Tested by:
          anest, jcapp, dartvader

2008-03-08 15:59 +0000 [r106945]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_zap.c: don't generate D-Channel "up" and "down"
          messages unless the channel state is actually changing; also,
          generate the "up" message when an implicit "up" occurs due to
          reception of a normal event when we thought the channel was
          "down"

2008-03-07 22:51 +0000 [r106895]  Russell Bryant <russell@digium.com>

        * apps/app_meetme.c: Only start the SLA thread if SLA has actually
          been configured.

2008-03-07 22:14 +0000 [r106842]  Jason Parker <jparker@digium.com>

        * main/editline/Makefile.in: Fix hardcoded grep in editline, were
          GNU grep is required. (closes issue #12124) Reported by: dmartin

2008-03-07 19:32 +0000 [r106788]  Joshua Colp <jcolp@digium.com>

        * main/channel.c: Ignore source update control frame. (closes issue
          #12168) Reported by: plack

2008-03-07 17:16 +0000 [r106704]  Russell Bryant <russell@digium.com>

        * include/asterisk/sched.h: Change a warning message to a debug
          message. This is happening quite frequently, and it is not worth
          spamming users with these messages unless we are pretty confident
          that it should never happen. As it stands today, it _will_ and
          _does_ happen and until that gets cleaned up a reasonable amount
          on the development side, let's not spam the logs of everyone
          else. (closes issue #12154)

2008-03-07 16:22 +0000 [r106552-106635]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_voicemail.c: Warn the user when a temporary greeting
          exists (Closes issue #11409)

        * main/rtp.c: Properly initialize rtp->schedid (Closes issue
          #12154)

        * apps/app_chanspy.c, apps/app_rpt.c, main/asterisk.c,
          apps/app_speech_utils.c, apps/app_voicemail.c, main/channel.c,
          funcs/func_enum.c, channels/chan_misdn.c, main/frame.c,
          main/manager.c: Safely use the strncat() function. (closes issue
          #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt
          uploaded by Corydon76 (license 14)

2008-03-06 22:10 +0000 [r106437]  Mark Michelson <mmichelson@digium.com>

        * main/pbx.c: Quell an annoying message that is likely to print
          every single time that ast_pbx_outgoing_app is called. The reason
          is that __ast_request_and_dial allocates the cdr for the channel,
          so it should be expected that the channel will have a cdr on it.
          Thanks to joetester on IRC for pointing this out

2008-03-06 04:40 +0000 [r106328]  Tilghman Lesher <tlesher@digium.com>

        * sounds/Makefile: Upgrade to the next release of sounds

2008-03-05 22:37 +0000 [r106237]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix a potential deadlock and a few
          different potential crashes. (closes issue #12145, reported by
          thiagarcia, patched by me)

2008-03-05 22:32 +0000 [r106235]  Joshua Colp <jcolp@digium.com>

        * channels/chan_oss.c, main/rtp.c, channels/chan_mgcp.c,
          apps/app_dial.c, main/channel.c, channels/chan_phone.c,
          main/dial.c, channels/chan_zap.c, channels/chan_sip.c,
          channels/chan_skinny.c, channels/chan_h323.c, main/file.c,
          channels/chan_alsa.c, apps/app_followme.c,
          include/asterisk/frame.h: Add a control frame to indicate the
          source of media has changed. Depending on the underlying
          technology it may need to change some things. (closes issue
          #12148) Reported by: jcomellas

2008-03-05 21:12 +0000 [r106178]  Michiel van Baak <michiel@vanbaak.info>

        * doc/realtime.txt: document var_metric so no bugreports will come
          in when it's actually a configuration issue. (issue #12151)
          Reported and patched by: caio1982 1.4 patch by me

2008-03-05 15:32 +0000 [r106038]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_zap.c: when a PRI call must be moved to a different
          B channel at the request of the other endpoint, ensure that any
          DSP active on the original channel is moved to the new one
          (closes issue #11917) Reported by: mavetju Tested by: mavetju

2008-03-05 15:17 +0000 [r106015]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_sip.c, include/asterisk/sched.h: Correctly
          initialize retransid in SIP, and ensure that the warning when
          failing to delete a schedule entry can actually hit the log.
          (closes issue #12140) Reported by: slavon Patches: sch2.patch
          uploaded by slavon (license 288) (Patch slightly modified by me)

2008-03-05 01:52 +0000 [r105932]  Russell Bryant <russell@digium.com>

        * main/rtp.c, main/translate.c, include/asterisk/frame.h: Fix a bug
          that I just noticed in the RTP code. The calculation for setting
          the len field in an ast_frame of audio was wrong when G.722 is in
          use. The len field represents the number of ms of audio that the
          frame contains. It would have set the value to be twice what it
          should be.

2008-03-04 18:10 +0000 [r105674-105676]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: In addition to setting the marker bit let's change
          our ssrc so they know for sure it is a different source.

        * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: When a
          new source of audio comes in (such as music on hold) make sure
          the marker bit gets set. (closes issue #10355) Reported by:
          wdecarne Patches: 10355.diff uploaded by file (license 11)
          (closes issue #11491) Reported by: kanderson

2008-03-04  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.19-rc1 released.

2008-03-04 04:31 +0000 [r105591]  Russell Bryant <russell@digium.com>

        * main/pbx.c: Backport a minor bug fix from trunk that I found
          while doing random code cleanup. Properly break out of the loop
          when a context isn't found when verify that includes are valid.

2008-03-03 18:06 +0000 [r105572]  Jason Parker <jparker@digium.com>

        * res/snmp/agent.c: Fix type for astNumChannels. (closes issue
          #12114) Reported by: jeffg Patches: 12114.patch uploaded by jeffg
          (license 192)

2008-03-03 17:16 +0000 [r105563-105570]  Russell Bryant <russell@digium.com>

        * channels/chan_local.c: In the case of an ast_channel allocation
          failure, take the local_pvt out of the pvt list before destroying
          it.

        * channels/chan_local.c: Fix a potential memory leak of the
          local_pvt struct when ast_channel allocation fails. Also, in
          passing, centralize the code necessary to destroy a local_pvt.

        * main/autoservice.c: Update the copyright information for
          autoservice. Most of the code in this file now is stuff that I
          have written recently ...

        * main/asterisk.c, main/channel.c, include/asterisk.h,
          main/autoservice.c: Merge in some changes from
          team/russell/autoservice-nochans-1.4 These changes fix up some
          dubious code that I came across while auditing what happens in
          the autoservice thread when there are no channels currently in
          autoservice. 1) Change it so that autoservice thread doesn't keep
          looping around calling ast_waitfor_n() on 0 channels twice a
          second. Instead, use a thread condition so that the thread
          properly goes to sleep and does not wake up until a channel is
          put into autoservice. This actually fixes an interesting bug, as
          well. If the autoservice thread is already running (almost always
          is the case), then when the thread goes from having 0 channels to
          have 1 channel to autoservice, that channel would have to wait
          for up to 1/2 of a second to have the first frame read from it.
          2) Fix up the code in ast_waitfor_nandfds() for when it gets
          called with no channels and no fds to poll() on, such as was the
          case with the previous code for the autoservice thread. In this
          case, the code would call alloca(0), and pass the result as the
          first argument to poll(). In this case, the 2nd argument to
          poll() specified that there were no fds, so this invalid pointer
          shouldn't actually get dereferenced, but, this code makes it
          explicit and ensures the pointers are NULL unless we have valid
          data to put there. (related to issue #12116)

2008-03-03 15:28 +0000 [r105557-105560]  Joshua Colp <jcolp@digium.com>

        * main/channel.c: It is possible for no audio to pass between the
          current digit and next digit so expand logic that clears
          emulation to AST_FRAME_NULL. (closes issue #11911) Reported by:
          edgreenberg Patches: v1-11911.patch uploaded by dimas (license
          88) Tested by: tbsky

        * channels/chan_sip.c: Add a comment to describe some logic.
          (closes issue #12120) Reported by: flefoll Patches:
          chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license
          244)

2008-02-29 23:34 +0000 [r105409]  Russell Bryant <russell@digium.com>

        * main/autoservice.c: Fix a major bug in autoservice. There was a
          race condition in the handling of the list of channels in
          autoservice. The problem was that it was possible for a channel
          to get removed from autoservice and destroyed, while the
          autoservice thread was still messing with the channel. This led
          to memory corruption, and caused crashes. This explains multiple
          backtraces I have seen that have references to autoservice, but
          do to the nature of the issue (memory corruption), could cause
          crashes in a number of areas. (fixes the crash in BE-386) (closes
          issue #11694) (closes issue #11940) The following issues could be
          related. If you are the reporter of one of these, please update
          to include this fix and try again. (potentially fixes issue
          #11189) (potentially fixes issue #12107) (potentially fixes issue
          #11573) (potentially fixes issue #12008) (potentially fixes issue
          #11189) (potentially fixes issue #11993) (potentially fixes issue
          #11791)

2008-02-29 14:47 +0000 [r105326]  Philippe Sultan <philippe.sultan@gmail.com>

        * res/res_jabber.c: Fix a potential memory leak

2008-02-29 14:34 +0000 [r105296]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_voicemail.c: If the message file does not exist, just
          return harmlessly, instead of crashing. (Closes issue #12108)

2008-02-29 13:48 +0000 [r105261]  Joshua Colp <jcolp@digium.com>

        * apps/app_voicemail.c: Bump up the size of the uniqueid variable.
          (closes issue #12107) Reported by: asgaroth

2008-02-29 13:05 +0000 [r105209]  Philippe Sultan <philippe.sultan@gmail.com>

        * res/res_jabber.c: Automatically create new buddy upon reception
          of a presence stanza of type subscribed. (closes issue #12066)
          Reported by: ffadaie Patches: branch-1.4-12066-1.diff uploaded by
          phsultan (license 73) trunk-12066-1.diff uploaded by phsultan
          (license 73) Tested by: ffadaie, phsultan

2008-02-28 22:23 +0000 [r105116]  Russell Bryant <russell@digium.com>

        * main/utils.c, include/asterisk/lock.h: Fix a bug in the lock
          tracking code that was discovered by mmichelson. The issue is
          that if the lock history array was full, then the functions to
          mark a lock as acquired or not would adjust the stats for
          whatever lock is at the end of the array, which may not be
          itself. So, do a sanity check to make sure that we're updating
          lock info for the proper lock. (This explains the bizarre stats
          on lock #63 in BE-396, thanks Mark!)

2008-02-28 21:56 +0000 [r105113]  Tilghman Lesher <tlesher@digium.com>

        * contrib/init.d/rc.debian.asterisk: Update init script for LSB
          compat (closes issue #9843) Reported by: ibc Patches:
          rc.debian.asterisk.patch uploaded by ibc (license 211) Tested by:
          paravoid

2008-02-28 20:11 +0000 [r105059]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: When using autofill, members who are in use
          should be counted towards the number of available members to call
          if ringinuse is set to yes. Thanks to jmls who brought this issue
          up on IRC

2008-02-28 19:20 +0000 [r104920-105005]  Jason Parker <jparker@digium.com>

        * main/cdr.c, main/pbx.c: Make pbx_exec pass an empty string into
          applications, if we get NULL. This protects against possible
          segfaults in applications that may try to use data before
          checking length (ast_strdupa'ing it, for example) (closes issue
          #12100) Reported by: foxfire Patches: 12100-nullappargs.diff
          uploaded by qwell (license 4)

        * channels/chan_skinny.c: According to a video at www.cisco.com,
          the 7921G supports 6 line appearances.

2008-02-28 00:05 +0000 [r104868]  Tilghman Lesher <tlesher@digium.com>

        * main/Makefile, build_tools/strip_nonapi: Compatibility fix for
          PPC64 (closes issue #12081) Reported by: jcollie Patches:
          asterisk-1.4.18-funcdesc.patch uploaded by jcollie (license 412)
          Tested by: jcollie, Corydon76

2008-02-27 21:49 +0000 [r104841]  Mark Michelson <mmichelson@digium.com>

        * main/dial.c: Two fixes: 1. Make the list of ast_dial_channels a
          lockable list. This is because in some cases, the ast_dial may
          exist in multiple threads due to asynchronous execution of its
          application, and I found some cases where race conditions could
          exist. 2. Check in ast_dial_join to be sure that the channel
          still exists before attempting to lock it, since it could have
          gotten hung up but the is_running_app flag on the
          ast_dial_channel may not have been cleared yet. (closes issue
          #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by
          putnopvut (license 60) Tested by: jvandal

2008-02-27 20:56 +0000 [r104787]  Joshua Colp <jcolp@digium.com>

        * apps/app_chanspy.c: Don't loop around infinitely trying to spy on
          our own channel, and don't forget to free/detach the datastore
          upon hangup of the spy.

2008-02-27 20:36 +0000 [r104783]  Mark Michelson <mmichelson@digium.com>

        * main/file.c: Bump a couple of more buffers up by 2 so that
          annoying warnings aren't generated like crazy on every
          fileexists_core call.

2008-02-27 18:15 +0000 [r104704]  Tilghman Lesher <tlesher@digium.com>

        * main/manager.c: Ensure the session ID can't be 0.

2008-02-27 17:41 +0000 [r104665]  Joshua Colp <jcolp@digium.com>

        * main/file.c: Bump up the buffer by 2.

2008-02-27 17:33 +0000 [r104625]  Russell Bryant <russell@digium.com>

        * apps/app_chanspy.c: Fix a problem in ChanSpy where it could get
          stuck in an infinite loop without being able to detect that the
          calling channel hung up. (closes issue #12076, reported by junky,
          patched by me)

2008-02-27 17:26 +0000 [r104598]  Jason Parker <jparker@digium.com>

        * res/res_features.c: Inherit language from the transfering channel
          on a blind transfer. (closes issue #11682) Reported by: caio1982
          Patches: local_atxfer_lang3-1.4.diff uploaded by caio1982
          (license 22) Tested by: caio1982, victoryure

2008-02-27 17:07 +0000 [r104596]  Joshua Colp <jcolp@digium.com>

        * main/loader.c: Use the lock (which already existed, it just
          wasn't used) on the updaters list to protect the contents instead
          of the overall module list lock. (closes issue #12080) Reported
          by: ChaseVenters

2008-02-27 16:53 +0000 [r104593]  Kevin P. Fleming <kpfleming@digium.com>

        * main/file.c: fallback to standard English prompts properly when
          using new prompt directory layout (closes issue #11831) Reported
          by: IgorG Patches: fallbacken.v1.diff uploaded by IgorG (license
          20) (modified by me to improve code and conform rest of function
          to coding guidelines)

2008-02-27 16:45 +0000 [r104591]  Russell Bryant <russell@digium.com>

        * channels/chan_zap.c: When we receive a known alarm, make sure
          that the unknown alarm flag is not still set to make sure that
          when we come back out of alarm, it gets reported in the log and
          manager interface (after discussion with tzafrir on the -dev
          list)

2008-02-27 15:52 +0000 [r104536]  Joshua Colp <jcolp@digium.com>

        * res/res_smdi.c: Only stop the MWI monitor thread if it was
          actually started. (closes issue #12086) Reported by: francesco_r

2008-02-27 01:15 +0000 [r104332-104334]  Russell Bryant <russell@digium.com>

        * apps/app_chanspy.c: Avoid some recursion in the cleanup code for
          the chanspy datastore (closes issue #12076, reported by junky,
          patched by me)

        * channels/chan_zap.c: Zaptel 1.4 now exposes FXO battery state as
          an alarm. However, Asterisk 1.4 does not know what to do with
          these alarms. Only Asterisk 1.6 cares about it. So, if we get an
          unknown alarm in chan_zap, don't generate confusing log messages
          about it.

2008-02-26 18:26 +0000 [r104132-104141]  Jason Parker <jparker@digium.com>

        * Makefile: Add badshell to .PHONY target (thanks Kevin)

        * Makefile: Since all shells aren't as awesome as bash, we have to
          fail if somebody tries to use a literal "~" in DESTDIR.

        * sounds/Makefile: Revert previous abspath change. ...abspath is
          new in GNU make 3.81. I feel so...defeated. Must find new fix!

        * sounds/Makefile: Fix a very bizarre issue we were seeing with our
          buildbot when using a DESTDIR that wasn't an absolute path (such
          as DESTDIR=~/asterisk-1.4). Apparently what was happening, was
          that some of the targets were being expanded to the full path, so
          $@ ended up being /root/asterisk-1.4/[...]/ rather than
          ~/asterisk-1.4/[...]/ It appears that this may be a new "feature"
          in GNU make. (*cough*
          http://en.wikipedia.org/wiki/Principle_of_least_surprise *cough*)

2008-02-26 00:25 +0000 [r104119]  Russell Bryant <russell@digium.com>

        * include/asterisk/smdi.h, apps/app_voicemail.c,
          channels/chan_zap.c, res/res_smdi.c, configs/smdi.conf.sample:
          Merge changes from team/russell/smdi-1.4 This commit brings in a
          significant set of changes to the SMDI support in Asterisk. There
          were a number of bugs in the current implementation, most notably
          being that it was very likely on busy systems to pop off the
          wrong message from the SMDI message queue. So, this set of
          changes fixes the issues discovered as well as introducing some
          new ways to use the SMDI support which are required to avoid the
          bugs with grabbing the wrong message off of the queue. This code
          introduces a new interface to SMDI, with two dialplan functions.
          First, you get an SMDI message in the dialplan using
          SMDI_MSG_RETRIEVE() and then you access details in the message
          using the SMDI_MSG() function. A side benefit of this is that it
          now supports more than just chan_zap. For example, with this
          implementation, you can have some FXO lines being terminated on a
          SIP gateway, but the SMDI link in Asterisk. Another issue with
          the current implementation is that it is quite common that the
          station ID that comes in on the SMDI link is not necessarily the
          same as the Asterisk voicemail box. There are now additional
          directives in the smdi.conf configuration file which let you map
          SMDI station IDs to Asterisk voicemail boxes. Yet another issue
          with the current SMDI support was related to MWI reporting over
          the SMDI link. The current code could only report a MWI change
          when the change was made by someone calling into voicemail. If
          the change was made by some other entity (such as with IMAP
          storage, or with a web interface of some kind), then the MWI
          change would never be sent. The SMDI module can now poll for MWI
          changes if configured to do so. This work was inspired by and
          primarily done for the University of Pennsylvania. (also related
          to issue #9260)

2008-02-26 00:03 +0000 [r104111]  Jason Parker <jparker@digium.com>

        * channels/chan_h323.c: IPTOS_MINCOST is not defined on Solaris.
          (closes issue #12050) Reported by: asgaroth Patches: 12050.patch
          uploaded by putnopvut (license 60)

2008-02-25 23:42 +0000 [r104102-104106]  Russell Bryant <russell@digium.com>

        * apps/app_chanspy.c: This patch fixes some pretty significant
          problems with how app_chanspy handles pointers to channels that
          are being spied upon. It was very likely that a crash would occur
          if the channel being spied upon hung up. This was because the
          current ast_channel handling _requires_ that the object is locked
          or else it could disappear at any time (except in the owning
          channel thread). So, this patch uses some channel datastore magic
          on the spied upon channel to be able to detect if and when the
          channel goes away. (closes issue #11877) (patch written by me,
          but thanks to kpfleming for the idea, and to file for review)

        * main/utils.c: Improve the lock tracking code a bit so that a
          bunch of old locks that threads failed to lock don't sit around
          in the history. When a lock is first locked, this checks to see
          if the last lock in the list was one that was failed to be
          locked. If it is, then that was a lock that we're no longer
          sitting in a trylock loop trying to lock, so just remove it.
          (inspired by issue #11712)

2008-02-25 21:37 +0000 [r104095]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Make it so a users.conf user creates both a
          SIP peer and a SIP user. The user will be used for inbound
          authentication for the device, and peer will be used for placing
          calls to the device. (closes issue #9044) Reported by: queuetue
          Patches: sip-gui-friend.diff uploaded by qwell (license 4)

2008-02-25 21:31 +0000 [r104094]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_voicemail.c: If the destination folder is full, don't
          delete a message when exiting. (closes issue #12065) Reported by:
          selsky Patch by: (myself)

2008-02-25 20:49 +0000 [r104092]  Jason Parker <jparker@digium.com>

        * main/config.c: Allow the use of #include and #exec in situations
          where the max include depth was only 1. Specifically, this fixes
          using #include and #exec in extconfig.conf. This was basically
          caused because the config file itself raises the include level to
          1. I opted not to raise the include limit, because recursion here
          could cause very bizarre behavior. Pointed out, and tested by
          jmls (closes issue #12064)

2008-02-25 18:38 +0000 [r104086]  Russell Bryant <russell@digium.com>

        * channels/chan_agent.c: Ensure that the channel doesn't disappear
          in agent_logoff(). If it does, it could cause a crash. (fixes the
          crash reported in BE-396)

2008-02-25 16:16 +0000 [r104082-104084]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: If a resubscription comes in for a dialog we
          no longer know about tell the remote side that the dialog does
          not exist so they subscribe again using a new dialog. (closes
          issue #10727) Reported by: s0l4rb03 Patches: 10727-2.diff
          uploaded by file (license 11)

        * channels/chan_sip.c: Due to recent changes tag will no longer be
          NULL if not present so we have to use ast_strlen_zero to see if
          it's actually blank. (closes issue #12061) Reported by: flefoll
          Patches: chan_sip.c.br14.patch_pedantic_no_totag uploaded by
          flefoll (license 244)

2008-02-22 22:45 +0000 [r104037]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_sip.c: Backwards debug message. (closes issue
          #12052) Reported by: flefoll Patches:
          chan_sip.c.br14.patch_found-notfound uploaded by flefoll (license
          244)

2008-02-21 21:05 +0000 [r104026-104027]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_zap.c: And as a followup to revision 104026,
          completely remove event-related calls from a section of code
          where we know there was no event to handle or get.

        * channels/chan_zap.c: Remove an incorrect debug message. It
          reported that it had received a specific event and tried to
          report which event was received. What actually was happening was
          that it was reporting the number of bytes returned from a call to
          read(). Thanks to Jared Smith for bringing the issue up on IRC

2008-02-21 14:33 +0000 [r104015]  Kevin P. Fleming <kpfleming@digium.com>

        * main/manager.c: reduce the likelihood that HTTP Manager session
          ids will consist of primarily '1' bits

2008-02-20 22:32 +0000 [r103956]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Clear up confusion when viewing the
          QUEUE_WAITING_COUNT of a "dead" realtime queue. Since from the
          user's perspective, the queue does exist, we shouldn't tell them
          we couldn't find the queue. Instead since it is a dead queue,
          report a 0 waiting count This issue was brought up on IRC by jmls

2008-02-20 22:06 +0000 [r103953]  Joshua Colp <jcolp@digium.com>

        * channels/chan_zap.c: Don't wait for additional digits when
          overlap dialing is enabled if the setup message contains the
          sending_complete information element. (closes issue #11785)
          Reported by: klaus3000 Patches:
          sending_complete_overlap_asterisk-1.4.17.patch.txt uploaded by
          klaus3000 (license 65)

2008-02-20 21:40 +0000 [r103904]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_local.c: Fix a crash if the channel becomes NULL
          while attempting to lock it. (closes issue #12039) Reported by:
          danpwi

2008-02-20 17:53 +0000 [r103845]  Tilghman Lesher <tlesher@digium.com>

        * main/stdtime/localtime.c: Compat fix for Solaris (closes issue
          #12022) Reported by: asgaroth Patches:
          20080219__bug12022.diff.txt uploaded by Corydon76 (license 14)
          Tested by: asgaroth

2008-02-19 20:28 +0000 [r103823]  Joshua Colp <jcolp@digium.com>

        * channels/h323/ast_h323.cxx: Send CallerID Name in setup message.
          (closes issue #11241) Reported by: tusar Patches:
          h323id_as_callerid_name.patch uploaded by tusar (license 344)

2008-02-19 20:02 +0000 [r103821]  Russell Bryant <russell@digium.com>

        * channels/chan_local.c: Account for the fact that the "other"
          channel can disappear while the local pvt is not locked. (fixes a
          problem introduced in rev 100581) (closes issue #12012) Reported
          by: stevedavies Patch by me

2008-02-19 17:31 +0000 [r103807-103812]  Joshua Colp <jcolp@digium.com>

        * configure, configure.ac: Don't look for launchd when cross
          compiling. (closes issue #12029) Reported by: ovi

        * channels/chan_sip.c: Fix building of chan_sip.

2008-02-19 10:27 +0000 [r103806]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Make sure we send error replies correctly by
          checking the via header.

2008-02-18 23:56 +0000 [r103801]  Joshua Colp <jcolp@digium.com>

        * main/channel.c: Ensure that emulated DTMFs do not get interrupted
          by another begin frame. (closes issue #11740) Reported by: gserra
          Patches: v1-11740.patch uploaded by dimas (license 88) (closes
          issue #11955) Reported by: tsearle (closes issue #10530) Reported
          by: xmarksthespot

2008-02-18 22:28 +0000 [r103790-103795]  Jason Parker <jparker@digium.com>

        * channels/chan_zap.c: Fix previous commit so that we actually
          disable echocanbridged if echocancel is off.

        * channels/chan_zap.c: Correct a message when echocancelwhenbridged
          is on, but echocancel is not. Issue #12019

2008-02-18 20:52 +0000 [r103786]  Mark Michelson <mmichelson@digium.com>

        * main/app.c: There was an invalid assumption when calculating the
          duration of a file that the filestream in question was created
          properly. Unfortunately this led to a segfault in the situation
          where an unknown format was specified in voicemail.conf and a
          voicemail was recorded. Now, we first check to be sure that the
          stream was written correctly or else assume a zero duration.
          (closes issue #12021) Reported by: jakep Tested by: putnopvut

2008-02-18 17:31 +0000 [r103780]  Tilghman Lesher <tlesher@digium.com>

        * main/rtp.c, channels/chan_sip.c: When a SIP channel is being
          auto-destroyed, it's possible for it to still be in bridge code.
          When that happens, we crash. Delay the RTP destruction until the
          bridge is ended. (closes issue #11960) Reported by: norman
          Patches: 20080215__bug11960__2.diff.txt uploaded by Corydon76
          (license 14) Tested by: norman

2008-02-18 16:37 +0000 [r103770]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_zap.c: Fix a linked list corruption that under the
          right circumstances could lead to a looped list, meaning it will
          traverse forever. (closes issue #11818) Reported by: michael-fig
          Patches: 11818.patch uploaded by putnopvut (license 60) Tested
          by: michael-fig

2008-02-18 16:11 +0000 [r103763-103768]  Joshua Colp <jcolp@digium.com>

        * main/asterisk.c: Backport fix from issue #9325. (closes issue
          #11980) Reported by: rbrunka

        * channels/chan_sip.c: Don't care if the extension given doesn't
          exist for subscription based MWI.

2008-02-15 23:31 +0000 [r103726-103741]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix a crash in chan_iax2 due to a race
          condition (closes issue #11780) Reported by: guillecabeza
          Patches: bug_iax2_jb_1.4.patch uploaded by guillecabeza (license
          380) bug_iax2_jb_trunk.patch uploaded by guillecabeza (license
          380)

        * main/loader.c: In the case that you try to directly reload a
          module has returned AST_MODULE_LOAD_DECLINE, log a message
          indicating that the module is not fully initialized and must be
          initialized using "module load".

        * main/loader.c: Don't attempt to execute the reload callback for a
          module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash
          that was reported against chan_console in trunk. (closes issue
          #11953, reported by junky, fixed by me)

2008-02-15 17:26 +0000 [r103688-103722]  Mark Michelson <mmichelson@digium.com>

        * doc/imapstorage.txt, configure, configure.ac: Final round of
          changes for configure script logic for IMAP Now if a directory is
          specified, then we will search that directory for a source
          installation of the IMAP toolkit. If none is found, then we will
          use that directory as the basis for detecting a package
          installation of the IMAP c-client. If that check fails, then
          configure will fail.

        * configure, configure.ac: Fix a bit of wrong logic in the
          configure script that caused problems when trying to configure
          without IMAP. Patch suggestion from phsultan, but I modified it
          slightly. (closes issue #12003) Reported by: pj Tested by:
          putnopvut

        * doc/imapstorage.txt, configure, configure.ac: I apparently
          misunderstood one of the requirements of this configure change.
          Now, if a source directory is specified with the --with-imap
          option, and a valid source installation is not detected there,
          then configure will fail and will not check for a package
          installation.

        * doc/imapstorage.txt: Make a small clarification in the
          documentation

        * doc/imapstorage.txt: Update documentation regarding configuration
          of IMAP

        * apps/app_voicemail.c, configure,
          include/asterisk/autoconfig.h.in, configure.ac: Change to the
          configure logic regarding IMAP. Prior to this commit, if you
          wished to configure Asterisk with IMAP support, you would use the
          --with-imap configure switch in one of the following two ways:
          --with-imap=/some/directory would look in the directory specified
          for a UW IMAP source installation --with-imap would assume that
          you had imap-2004g installed in .. relative to the Asterisk
          source With this set of changes the two above options still work
          the same, but there are two new behaviors, too.
          --with-imap=system will assume that you have -libc-client.so
          where you store your shared objects and will attempt to find
          c-client headers in your include path either in the imap or
          c-client directory. If either of the two original methods of
          specifying the imap option should fail, then the check for
          --with-imap =system will be performed in addition. It is only
          after this "system" check that failure can happen.

        * apps/app_voicemail.c: Fix build for non-IMAP builds

        * apps/app_voicemail.c: Fix the new message count if delete=yes
          when using IMAP storage. (closes issue #11406) Reported by:
          jaroth Patches: deleteflag_v2.patch uploaded by jaroth (license
          50) Tested by: jaroth

2008-02-14 19:51 +0000 [r103683-103684]  Jason Parker <jparker@digium.com>

        * funcs/func_cdr.c: swap location for this..

        * funcs/func_cdr.c: Document the 'l' option to the CDR() function.
          (Thanks voipgate for pointing out the option, and Leif for
          providing text for it.) Closes issue #11695.

2008-02-13 06:25 +0000 [r103556-103607]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_agent.c: We aren't talking to ourselves; we're
          talking to someone else. (closes issue #11771) Reported by:
          msetim Patches: ami_agent_talkingto-1.4.diff uploaded by caio1982
          (license 22) Tested by: caio1982, msetim

        * apps/app_voicemail.c: Refuse to load app_voicemail if res_adsi is
          not loaded (which is a symbol dependency) (closes issue #11760)
          Reported by: non-poster Patches: 20080114__bug11760.diff.txt
          uploaded by Corydon76 (license 14) Tested by: Corydon76,
          non-poster, jamesgolovich

2008-02-12 22:24 +0000 [r103503-103504]  Jason Parker <jparker@digium.com>

        * main/asterisk.c: revert accidental change from last commit. oops

        * contrib/scripts/safe_asterisk, main/asterisk.c: Remove condition
          that was impossible.

2008-02-12 15:09 +0000 [r103324-103385]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Even if no CallerID name or number has been
          provided by the remote party still use the configured sip.conf
          ones. (closes issue #11977) Reported by: pj

        * apps/app_meetme.c: If entering a conference with the 'w' option
          ensure that we can't listen or speak until the marked user
          appears. (closes issue #11835) Reported by: alanmcmillan

2008-02-11 17:05 +0000 [r103315]  Kevin P. Fleming <kpfleming@digium.com>

        * configs/zapata.conf.sample: improve 2BCT documentation a bit
          (thanks Jared)

2008-02-09 06:23 +0000 [r103197]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_voicemail.c: Commit fix for being unable to send
          voicemail from VoiceMailMain Reported by: William F Acker (via
          the -users mailing list) Patch by: Corydon76 (license 14)

2008-02-08 18:48 +0000 [r103070-103120]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Prevent a potential three-thread deadlock. Also
          added a comment block to explicitly state the locking order
          necessary inside app_queue. (closes issue #11862) Reported by:
          flujan Patches: 11862.patch uploaded by putnopvut (license 60)
          Tested by: flujan

        * channels/chan_iax2.c: Yield the thread and return -1 if the ioctl
          fails for Zaptel timing device. (closes issue #11891) Reported
          by: tzafrir

2008-02-08 15:08 +0000 [r102968]  Joshua Colp <jcolp@digium.com>

        * channels/chan_iax2.c: Make sure the presence of dbsecret is
          factored into user scoring. (closes issue #11952) Reported by:
          bbhoss

2008-02-07 19:53 +0000 [r102858]  Jason Parker <jparker@digium.com>

        * res/res_features.c: Specify which digit string was matched in
          debug message. (closes issue #11949) Reported by: dimas Patches:
          v1-feature-debug.patch uploaded by dimas (license 88)

2008-02-07 16:41 +0000 [r102807]  Kevin P. Fleming <kpfleming@digium.com>

        * configs/zapata.conf.sample: document usage of 'transfer'
          configuration option for ISDN PRI switch-side transfers

2008-02-06 17:59 +0000 [r102653-102725]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Only consider a T.38-only INVITE compatible
          if we have both a joint capability between us and them and if
          they provided T.38.

        * main/global_datastores.c: Add missing header file and
          ASTERISK_FILE_VERSION usage. (closes issue #11936) Reported by:
          snuffy

2008-02-06 15:19 +0000 [r102651]  Russell Bryant <russell@digium.com>

        * configs/features.conf.sample: Clarify setting DYNAMIC_FEATURES so
          that it gets inherited by outbound channels. (due to a discussion
          between me and a user via email)

2008-02-06 11:48 +0000 [r102627]  Kevin P. Fleming <kpfleming@digium.com>

        * pbx/Makefile, res/Makefile: ensure that all remaining
          multi-object modules are built using their proper CFLAGS and
          include directory paths

2008-02-06 00:26 +0000 [r102576]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_voicemail.c: Move around some defines to unbreak ODBC
          storage. (closes issue #11932) Reported by: snuffy

2008-02-05 20:02 +0000 [r102453]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_mgcp.c: Clear the DTMF buffer on hangup. (closes
          issue #11919) Reported by: eferro Patches:
          mgcp_dtmfclean_on_hangup.diff uploaded by eferro (license 337)
          Tested by: eferro

2008-02-05 19:52 +0000 [r102450]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: If a REGISTER attempt comes in that is a
          retransmission of a previous REGISTER do not create a new nonce
          value. (issue #BE-381)

2008-02-05 17:15 +0000 [r102425]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/Makefile: ensure that components of chan_misdn.so are
          built using any special build options that the configure script
          generated (reported by Philipp Kempgen on asterisk-dev)

2008-02-05 15:09 +0000 [r102378]  Joshua Colp <jcolp@digium.com>

        * res/res_clioriginate.c: Perform dialing asynchronously when using
          the originate CLI command so the CLI does not appear to block.
          (closes issue #11927) Reported by: bbhoss

2008-02-04 21:06 +0000 [r102214-102323]  Tilghman Lesher <tlesher@digium.com>

        * main/asterisk.c, utils/muted.c, configure,
          include/asterisk/autoconfig.h.in, configure.ac: Cross-platform
          fix: OS X now deprecates the use of the daemon(3) API. (closes
          issue #11908) Reported by: oej Patches:
          20080204__bug11908.diff.txt uploaded by Corydon76 (license 14)
          Tested by: Corydon76

        * funcs/func_strings.c: Missing braces. (closes issue #11912)
          Reported by: dimas Patches: sprintf.patch uploaded by dimas
          (license 88)

2008-02-03 16:38 +0000 [r102090-102142]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Use the same CSEQ on CANCEL as on INVITE
          (according to RFC 3261) (closes issue #9492) Reported by:
          kryptolus Patches: bug9492.txt uploaded by oej (license 306)
          Tested by: oej

        * channels/chan_sip.c: Handle ACK and CANCEL in an invite
          transaction - even if we get INFO transactions during the actual
          call setup. (closes issue #10567) Reported by: jacksch Tested by:
          oej Patch by: oej inspired by suggestions from neutrino88 in the
          bug tracker

2008-02-01 23:06 +0000 [r101989]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: Change the SDP_SAMPLE_RATE macro. It turns
          out that even though G.722 is 16 kHz, it is supposed to specified
          as 8 kHz in the RTP, and RTP timestamps are supposed to be
          calculated based on 8 kHz. (Apparently this is due to a bug in a
          spec, but people follow it anyway, because it's the spec ...)

2008-02-01 21:54 +0000 [r101894-101942]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_voicemail.c: Fix the VM_DUR variable for forwarded
          voicemail, and fixed several other bugs while I'm in the area.
          (closes issue #11615) Reported by: jamessan Patches:
          20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14)
          Tested by: Corydon76, jamessan

        * configure, include/asterisk/autoconfig.h.in, configure.ac,
          acinclude.m4: Change detection of getifaddrs to use
          AST_C_COMPILE_CHECK, backported from trunk (as suggested by
          kpfleming)

2008-02-01 17:41 +0000 [r101822]  Jason Parker <jparker@digium.com>

        * apps/app_authenticate.c: Remove a needless (and incorrect) call
          to feof() after fgets(). This would have exited the loop early if
          you had an authentication file with no newline at the end.

2008-02-01 17:27 +0000 [r101818-101820]  Russell Bryant <russell@digium.com>

        * apps/app_authenticate.c: off by one error

        * apps/app_authenticate.c: Don't overwrite the last character of a
          line if it's not a newline. This would happen if the last line in
          the file doesn't have a newline. (pointed out by Qwell)

2008-02-01 15:55 +0000 [r101772]  Tilghman Lesher <tlesher@digium.com>

        * configure, include/asterisk/autoconfig.h.in, configure.ac,
          main/acl.c: Compatibility fix for OpenWRT (reported by Brian
          Capouch via the mailing list)

2008-02-01 00:32 +0000 [r101693]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Add some more sanity checking on IAX2 dial
          strings for the case that no peer or hostname was provided, which
          is the one part of the dial string that is absolutely required.
          If it's not there, bail out. (closes issue #11897) Reported by
          sokhapkin Patch by me

2008-02-01 00:06 +0000 [r101649]  Mark Michelson <mmichelson@digium.com>

        * apps/app_amd.c: From bugtracker: "fix totalAnalysisTime to handle
          periods of no channel activity" (closes issue #9256) Reported by:
          cmaj Patches: amd-dont-wait-too-long-for-frames-take3.diff.txt
          uploaded by cmaj (license 111) Tested by: cmaj, skygreg, ZX81,
          rjain

2008-01-31  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.18 released.

2008-01-31 23:10 +0000 [r101601]  Russell Bryant <russell@digium.com>

        * main/translate.c, main/file.c: Fix a couple of places where
          ast_frfree() was not called on a frame that came from a
          translator. This showed itself by g729 decoders not getting
          released. Since the flag inside the translator frame never got
          unset by freeing the frame to indicate it was no longer in use,
          the translators never got destroyed, and thus the g729 licenses
          were not released. (closes issue #11892) Reported by: xrg
          Patches: 11892.diff uploaded by russell (license 2) Tested by:
          xrg, russell

2008-01-31 21:00 +0000 [r101531]  Mark Michelson <mmichelson@digium.com>

        * res/res_monitor.c: 1. Prevent the addition of an extra '/' to the
          beginning of an absolute pathname. 2. If ast_monitor_change_fname
          is called and the new filename is the same as the old, then exit
          early and don't set the filename_changed field in the monitor
          structure. Setting it in this case was causing ast_monitor_stop
          to erroneously delete them. (closes issue #11741) Reported by:
          garlew Tested by: putnopvut

2008-01-31 19:52 +0000 [r101482]  Jason Parker <jparker@digium.com>

        * channels/chan_sip.c, channels/chan_iax2.c: Solaris compat fixes
          for struct in_addr funkiness. Issue #11885, patch by snuffy.

2008-01-31 19:30 +0000 [r101480]  Steve Murphy <murf@digium.com>

        * main/pbx.c: closes issue #11845; that's the one where there's a
          1004 byte cdr leak with every AMI Redirect to a zap channel

2008-01-31 19:17 +0000 [r101413-101433]  Russell Bryant <russell@digium.com>

        * channels/chan_agent.c: Add more missing locking of the agents
          list ...

        * channels/chan_agent.c: Move the locking from find_agent() into
          the agent dialplan function handler to ensure that the agent
          doesn't disappear while we're looking at it.

        * channels/chan_agent.c: Add missing locking to the find_agent()
          function.

2008-01-30 15:41 +0000 [r101222]  Joshua Colp <jcolp@digium.com>

        * main/slinfactory.c: Fix an issue where if a frame of higher
          sample size preceeded a frame of lower sample size and
          ast_slinfactory_read was called with a sample size of the
          combined values or higher a crash would happen. (closes issue
          #11878) Reported by: stuarth

2008-01-30 15:34 +0000 [r101219]  Jason Parker <jparker@digium.com>

        * configs/extensions.conf.sample: Change default config to use
          descending channel order of groups, rather than ascending. Fixes
          a potential source of confusion in glare-type situations. Issue
          11875, reported by JimVanM.

2008-01-30 15:23 +0000 [r101216]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Fix a logic error with regards to autofill.
          Prior to this change, it was possible for a caller to go out of
          turn if autofill were enabled and callers ahead in the queue were
          attempting to call a member. This change fixes this.

2008-01-30 11:20 +0000 [r101152]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Stop musiconhold on attended transfer.
          (closes issue #11872) Reported by: gareth Patches:
          svn-101018.patch uploaded by gareth (license 208)

2008-01-29 23:50 +0000 [r101080]  Dwayne M. Hubbard <dhubbard@digium.com>

        * build_tools/make_version: updated build_tools to handle the
          autotag directory structure changes; changes related to BE-353.
          Patch by The Russell and reviewed by The Me.

2008-01-29 23:02 +0000 [r100973-101035]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Remove a memory leak from updating realtime
          queues

        * apps/app_queue.c: Fixing an erroneous return value returned when
          attempting to pause or unpause a queue member fails. Fixes
          BE-366, thanks to John Bigelow for writing the patch.

2008-01-29 17:57 +0000 [r100934]  Joshua Colp <jcolp@digium.com>

        * apps/app_mixmonitor.c: Don't forget to record the channel so we
          know whether it is bridged or not later. (closes issue #11811)
          Reported by: slavon

2008-01-29 17:43 +0000 [r100932]  Russell Bryant <russell@digium.com>

        * main/Makefile: Fix the last couple of issues related to building
          from a path that contains spaces. (closes issue #11834)

2008-01-29 17:41 +0000 [r100930]  Jason Parker <jparker@digium.com>

        * channels/misdn_config.c: Initialize an array to 0s if config
          option not specified. (closes issue #11860) Patches:
          misdn_get_config.v1.diff uploaded by IgorG (license 20)

2008-01-29 17:21 +0000 [r100882-100922]  Russell Bryant <russell@digium.com>

        * Makefile: Use GNU make magic instead of shell magic to escape
          spaces in the working directory. (related to issue #11834)

        * Makefile: Fix building Asterisk when the working path has spaces
          in it. (closes issue #11834) Reported by: spendergrass Patched
          by: me

2008-01-29 16:10 +0000 [r100835]  Jason Parker <jparker@digium.com>

        * channels/chan_zap.c: Allow zap groups above 30 to work properly.
          (closes issue #11590) Reported by: tbsky

2008-01-29 10:36 +0000 [r100793]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c: fixed potential segfault in misdn show
          channels CLI command

2008-01-29 08:26 +0000 [r100740]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: (closes issue #11736) Reported by: MVF
          Patches: bug11736-2.diff uploaded by oej (license 306) Tested by:
          oej, MVF, revolution (russellb: This was the showstopper for the
          release.)

2008-01-28 21:02 +0000 [r100675]  Tilghman Lesher <tlesher@digium.com>

        * main/pbx.c: WaitExten didn't handle AbsoluteTimeout properly
          (went to 't' instead of 'T')

2008-01-28 20:55 +0000 [r100673]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_vpb.cc, UPGRADE.txt: Undoing the deprecation of
          chan_vpb. It is alive and well.

2008-01-28 20:42 +0000 [r100672]  Jason Parker <jparker@digium.com>

        * apps/app_voicemail.c: When using ODBC_STORAGE, make sure we put
          greeting files into the database like we do with the others.
          Issue #11795 Reported by: dimas Patches: vmgreet.patch uploaded
          by dimas (license 88)

2008-01-28 18:34 +0000 [r100626-100629]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: For some reason, the use of this strdupa()
          is leading to memory corruption on freebsd sparc64. This trivial
          workaround fixes it. (closes issue #10300, closes issue #11857,
          reported by mattias04 and Home-of-the-Brave)

        * res/res_features.c: Fix a crash in ast_masq_park_call() (issue
          #11342) Reported by: DEA Patches: res_features-park.txt uploaded
          by DEA (license 3)

2008-01-28 18:23 +0000 [r100624]  Jason Parker <jparker@digium.com>

        * channels/chan_zap.c: Correct a comment which made little/no
          sense.

2008-01-28 17:15 +0000 [r100581]  Russell Bryant <russell@digium.com>

        * main/channel.c, channels/chan_local.c,
          include/asterisk/channel.h: Make some deadlock related fixes.
          These bugs were discovered and reported internally at Digium by
          Steve Pitts. - Fix up chan_local to ensure that the channel lock
          is held before the local pvt lock. - Don't hold the channel lock
          when executing the timing function, as it can cause a deadlock
          when using chan_local. This actually changes the code back to be
          how it was before the change for issue #10765. But, I added some
          other locking that I think will prevent the problem reported
          there, as well.

2008-01-27 21:59 +0000 [r100465]  Tilghman Lesher <tlesher@digium.com>

        * main/rtp.c, channels/chan_mgcp.c, main/cdr.c,
          channels/chan_misdn.c, main/dnsmgr.c, channels/chan_sip.c,
          channels/chan_h323.c, include/asterisk/sched.h, main/file.c,
          pbx/pbx_dundi.c, channels/chan_iax2.c: When deleting a task from
          the scheduler, ignoring the return value could possibly cause
          memory to be accessed after it is freed, which causes all sorts
          of random memory corruption. Instead, if a deletion fails, wait a
          bit and try again (noting that another thread could change our
          taskid value). (closes issue #11386) Reported by: flujan Patches:
          20080124__bug11386.diff.txt uploaded by Corydon76 (license 14)
          Tested by: Corydon76, flujan, stuarth`

2008-01-25 22:32 +0000 [r100418]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_vpb.cc, UPGRADE.txt: Deprecating chan_vpb. It is
          now preferred that users of Voicetronix products use chan_zap in
          combination with their zaptel drivers.

2008-01-25 21:24 +0000 [r100378]  Jason Parker <jparker@digium.com>

        * channels/chan_sip.c: This would have never been true, since we're
          passing (sizeof(req.data) - 1) as the len to recvfrom().

2008-01-24 21:57 +0000 [r100264]  Kevin P. Fleming <kpfleming@digium.com>

        * include/asterisk/app.h: make these macros not assume that the
          only other field in the structure is 'argc'... this is true when
          someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable
          to define your own structure as long as it has the right fields

2008-01-24 17:22 +0000 [r100164]  Russell Bryant <russell@digium.com>

        * main/asterisk.c: Update main Asterisk copyright info to 2008

2008-01-24 16:41 +0000 [r100138]  Jason Parker <jparker@digium.com>

        * main/acl.c: Fix compilation on Solaris. (closes issue #11832)
          Patches: bug-11832.diff uploaded by snuffy (license 35)

2008-01-23 21:07 +0000 [r99977-99978]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Second attempt. Don't change invitestate
          when receiving 18x messages in CANCEL state. (issue #11736)
          Reported by: MVF Patch by oej.

        * channels/chan_sip.c: Make sure we don't cancel destruction on
          calls in CANCEL state, even if we get 183 while waiting for
          answer on our CANCEL. (issue #11736) Reported by: MVF Patches:
          bug11736.txt uploaded by oej (license 306) Tested by: MVF

2008-01-23 20:25 +0000 [r99975]  Mark Michelson <mmichelson@digium.com>

        * apps/app_externalivr.c: Fixing a typo.

2008-01-23 17:46 +0000 [r99923]  Russell Bryant <russell@digium.com>

        * apps/app_chanspy.c: ChanSpy issues a beep when it starts at the
          beginning of a list of channels to potentially spy on. However,
          if there were no matching channels, it would beep at you over and
          over, which is pretty annoying. Now, it will only beep once in
          the case that there are no channels to spy on, but it will still
          beep again once it reaches the beginning of the channel list
          again. (closes issue #11738, patched by me)

2008-01-23 16:18 +0000 [r99878]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: These flag tests were illogical. They were
          testing sip_peer flags on a sip_pvt. Thanks to Russell for
          helping to get this odd problem figured out.

2008-01-23 04:31 +0000 [r99718-99777]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_voicemail.c: When we reset the password via an external
          command, we should also reset the password stored in the
          in-memory list, too (otherwise it doesn't really take effect).
          (closes issue #11809) Reported by: davetroy Patches:
          fix_externpass.diff uploaded by davetroy (license 384)

        * res/res_odbc.c: Oops, should have checked for a NULL obj, here,
          too

        * main/acl.c: Just confirmed that all current platforms need this
          header file

2008-01-22 20:56 +0000 [r99652]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Thanks to Russell's education I realize that
          BUFSIZ has changed since I learned the C language over 20 years
          ago... Resetting chan_sip to the size of BUFSIZ that I expected
          in my old head to avoid to heavy memory allocations on some
          systems.

2008-01-22 20:34 +0000 [r99643]  Tilghman Lesher <tlesher@digium.com>

        * main/acl.c: Fix the defines for OS X (and Solaris, too)

2008-01-22 17:41 +0000 [r99592-99594]  Olle Johansson <oej@edvina.net>

        * channels/chan_local.c, res/res_features.c, channels/chan_agent.c,
          apps/app_followme.c: Add more dependencies on chan_local and add
          a note to the description of chan_local so that people don't
          disable it in menuselect just to clean up.

        * apps/app_dial.c: Add dependency on chan_local to app_dial. Dial
          still runs without chan_local, but will be missing forwarding
          functionality.

2008-01-22 16:54 +0000 [r99540]  Tilghman Lesher <tlesher@digium.com>

        * main/acl.c: Ensure that we can get an address even when we don't
          have a default route. (closes issue #9225) Reported by: junky
          Patches: 20080122__bug9225.diff.txt uploaded by Corydon76
          (license 14) Tested by: oej, loloski, sergee

2008-01-22 15:08 +0000 [r99501]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Cleaning up some documentation that led to
          confusion in a bug report

2008-01-21 23:55 +0000 [r99426]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_local.c: Fixing an issue wherein monitoring local
          channels was not possible. During a channel masquerade, the
          monitors on the two channels involved are swapped. In 99% of the
          cases this results in the desired effect. However, if monitoring
          a local channel, this caused the monitor which was on the local
          channel to get moved onto a channel which is immediately hung up
          after the masquerade has completed. By swapping the monitors
          prior to the masquerade, we avoid the problem by tricking the
          masquerade into placing the monitor back onto the channel where
          we want it. During the investigation of the issue, the channel's
          monitor was the only thing that was swapped in such a manner
          which did not make sense to have done. All other variable
          swapping made sense.

2008-01-21 18:11 +0000 [r99341]  Tilghman Lesher <tlesher@digium.com>

        * res/res_odbc.c, configs/res_odbc.conf.sample,
          include/asterisk/res_odbc.h: Permit the user to specify number of
          seconds that a connection may remain idle, which fixes a crash on
          reconnect with the MyODBC driver. (closes issue #11798) Reported
          by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt
          uploaded by Corydon76 (license 14) Tested by: mvanbaak

2008-01-21 16:01 +0000 [r99301]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Bump the buffer size for Via headers up to
          512. There are some exceptionally large Via headers out there.
          (closes issue #11783) Reported by: ofirroval

2008-01-19 10:05 +0000 [r99187]  Russell Bryant <russell@digium.com>

        * main/slinfactory.c: Fix a couple of memory leaks with frame
          handling. Specifically, ast_frame_free() needed to be called on
          the frame that came from the translator to signed linear.

2008-01-18 22:57 +0000 [r99127]  Joshua Colp <jcolp@digium.com>

        * include/asterisk/channel.h: Remove the __ in front of the unused
          variable. This causes some compilers to freak out.

2008-01-18 21:37 +0000 [r99079-99081]  Russell Bryant <russell@digium.com>

        * include/asterisk/translate.h, main/frame.c: Revert adding the
          packed attribute, as it really doesn't make sense why that would
          do any good. Fix the real bug, which is to do the check to see if
          the frame came from a translator at the beginning of
          ast_frame_free(), instead of at the end. This ensures that it
          always gets checked, even if none of the parts of the frame are
          malloc'd, and also ensures that we aren't looking at free'd
          memory in the case that it is a malloc'd frame. (closes issue
          #11792, reported by explidous, patched by me)

        * include/asterisk/translate.h: Since we're relying on the offset
          between the frame and the beginning of the translator pvt struct,
          set the packed attribute to make sure we get to the right place.
          (potential fix for issue #11792)

2008-01-18 17:13 +0000 [r99032]  Terry Wilson <twilson@digium.com>

        * res/res_features.c: This should at least temporarily fix a
          problem where the 't' Dial option is incorrectly passed to the
          transferee when built-in attended transfers are used. There is
          still a problem with 'T', but better to fix some problems than no
          problems while we work on it. (closes issue #7904) Reported by:
          k-egg Patches: transfer-fix-b14-r97657.diff uploaded by sergee
          (license 138) Tested by: sergee, otherwiseguy

2008-01-17 23:42 +0000 [r99007-99014]  Pari Nannapaneni <paripurnachand@digium.com>

        * configs/cdr.conf.sample: doh! revert a revert of a revert
          (changed by mistake in 99010)

        * main/manager.c, configs/cdr.conf.sample: missed that one while
          reverting

        * main/manager.c: reverting 99001 - We need the Max-Age for
          extending the life of cookie mansession_id

2008-01-17 22:37 +0000 [r99004]  Russell Bryant <russell@digium.com>

        * main/frame.c, channels/chan_iax2.c, include/asterisk/frame.h:
          Have IAX2 optimize the codec translation path just like chan_sip
          does it. If the caller's codec is in our codec list, move it to
          the top to avoid transcoding. (closes issue #10500) Reported by:
          stevedavies Patches: iax-prefer-current-codec.patch uploaded by
          stevedavies (license 184) iax-prefer-current-codec.1.4.patch
          uploaded by stevedavies (license 184) Tested by: stevedavies, pj,
          sheldonh

2008-01-17 21:31 +0000 [r99001]  Kevin P. Fleming <kpfleming@digium.com>

        * main/manager.c: we should only send the Set-Cookie header to the
          browser on the first response after creating a manager session,
          not on every response (doing so causes the browser to clear any
          local cookies it may have associated with the session)

2008-01-17 16:19 +0000 [r98991]  Jason Parker <jparker@digium.com>

        * configs/zapata.conf.sample: Add a clarification about the
          immediate= option of zapata.conf Issue 11784, patch by klaus3000.

2008-01-16 22:36 +0000 [r98982]  Russell Bryant <russell@digium.com>

        * .cleancount, include/asterisk/channel.h: Add an unused pointer to
          the ast_channel struct. This makes the ast_channel structure
          retain the same size as it had in previous 1.4 releases. Also,
          all of the offsets for members in the structure are still the
          same (except for the two pointers that got replaced for the new
          spy/whisper architecture.)

2008-01-16 20:34 +0000 [r98966-98973]  Joshua Colp <jcolp@digium.com>

        * .cleancount: Bump up cleancount due to previous commit that
          changed the channel structure.

        * apps/app_chanspy.c, apps/app_mixmonitor.c, main/rtp.c,
          main/channel.c, apps/app_meetme.c, include/asterisk/audiohook.h
          (added), main/Makefile, include/asterisk/chanspy.h (removed),
          include/asterisk/channel.h, main/audiohook.c (added): Replace
          current spy architecture with backport of audiohooks. This should
          take care of current known spy issues.

        * channels/chan_iax2.c: Add missing NULLs at end of two
          ast_load_realtimes. (closes issue #11769) Reported by: tequ
          Patches: chaniax.patch uploaded by dimas (license 88)

2008-01-16 17:20 +0000 [r98964]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_local.c: Fix a deadlock in chan_local in
          local_hangup. There was contention because the local_pvt was held
          and it was attempting to lock a channel, which is the incorrect
          locking order. (closes issue #11730) Reported by: UDI-Doug
          Patches: 11730.patch uploaded by putnopvut (license 60) Tested
          by: UDI-Doug

2008-01-16 15:08 +0000 [r98951-98960]  Joshua Colp <jcolp@digium.com>

        * main/dial.c: Introduce a lock into the dialing API that protects
          it when destroying the structure. (closes issue #11687) Reported
          by: callguy Patches: 11687.diff uploaded by file (license 11)

        * main/rtp.c: Add two more SDP names for ulaw and alaw. (closes
          issue #11777) Reported by: tootai

        * channels/chan_sip.c: Don't drop the old record route information
          when dealing with packets related to a reinvite. (closes issue
          #11545) Reported by: kebl0155 Patches: reinvite-patch.txt
          uploaded by kebl0155 (license 356)

        * build_tools/menuselect-deps.in, configure,
          include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
          configure.ac, makeopts.in: Add autoconf logic for speexdsp. Later
          versions use a separate library for some things so we need to use
          it if present in codec_speex. (closes issue #11693) Reported by:
          yzg

2008-01-15 23:50 +0000 [r98943-98946]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: Change a buffer in check_auth() to be a
          thread local dynamically allocated buffer, instead of a massive
          buffer on the stack. This fixes a crash reported by Qwell due to
          running out of stack space when building with LOW_MEMORY defined.
          On a very related note, the usage of BUFSIZ in various places in
          chan_sip is arbitrary and careless. BUFSIZ is a system specific
          define. On my machine, it is 8192, but by definition (according
          to google) could be as small as 256. So, this buffer in
          check_auth was 16 kB. We don't even support SIP messages larger
          than 4 kB! Further usage of this define should be avoided, unless
          it is used in the proper context.

        * main/rtp.c, include/asterisk/translate.h, main/frame.c,
          main/translate.c, main/abstract_jb.c, channels/chan_iax2.c,
          codecs/codec_zap.c, include/asterisk/frame.h: Commit a fix for
          some memory access errors pointed out by the valgrind2.txt output
          on issue #11698. The issue here is that it is possible for an
          instance of a translator to get destroyed while the frame
          allocated as a part of the translator is still being processed.
          Specifically, this is possible anywhere between a call to
          ast_read() and ast_frame_free(), which is _a lot_ of places in
          the code. The reason this happens is that the channel might get
          masqueraded during this time. During a masquerade, existing
          translation paths get destroyed. So, this patch fixes the issue
          in an API and ABI compatible way. (This one is for you,
          paravoid!) It changes an int in ast_frame to be used as flag
          bits. The 1 bit is still used to indicate that the frame contains
          timing information. Also, a second flag has been added to
          indicate that the frame came from a translator. When a frame with
          this flag gets released and has this flag, a function is called
          in translate.c to let it know that this frame is doing being
          processed. At this point, the flag gets cleared. Also, if the
          translator was requested to be destroyed while its internal frame
          still had this flag set, its destruction has been deffered until
          it finds out that the frame is no longer being processed.
          Admittedly, this feels like a hack. But, it does fix the issue,
          and I was not able to think of a better solution ...

2008-01-15 20:08 +0000 [r98894-98934]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Based on the boundary found move over the
          correct amount. (closes issue #11750) Reported by: tasker

        * channels/chan_sip.c: Accept "; boundary=" not just ";boundary="
          in the multipart mixed content type. (closes issue #11750)
          Reported by: tasker

2008-01-14 20:59 +0000 [r98849]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Adding in appropriate unlocks for the locks
          I added. Thanks to joetester on IRC for pointing this out.

2008-01-14 17:38 +0000 [r98774]  Russell Bryant <russell@digium.com>

        * main/translate.c: Revert a change that introduces an unacceptable
          performance hit and is causing memory leaks ... (from rev 97973)

2008-01-14 16:35 +0000 [r98733-98737]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Fixing another compilation error. I'm a bit off
          today :(

        * apps/app_queue.c: Oops. Last commit had compilation error.

        * apps/app_queue.c: Adding explicit defaults for missing options to
          init_queue. This is necessary because if a user either removes or
          comments one of these options and reloads their queues, the
          option will not reset to its default, instead maintaining the
          value from prior to the reload. Thanks to John Bigelow for
          pointing this error out to me.

2008-01-12 00:05 +0000 [r98467]  Tilghman Lesher <tlesher@digium.com>

        * res/res_odbc.c: Add a connection timeout attribute, as that was
          what was intended with the login timeout, but ODBC divides it up
          into 2 different timeouts. (Closes issue #11745)

2008-01-11 22:46 +0000 [r98390]  Russell Bryant <russell@digium.com>

        * pbx/pbx_dundi.c: Fix up setting the EID on BSD based systems.
          (closes issue #11646) Reported by: caio1982 Patches:
          dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22)
          dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested
          by: caio1982, mvanbaak

2008-01-11 21:28 +0000 [r98372]  Pari Nannapaneni <paripurnachand@digium.com>

        * main/http.c: Comment explaining how to force browser to always
          read some html files from server.

2008-01-11 19:51 +0000 [r98317-98325]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: If the incoming RTP stream changes codec force the
          bridge to break if the other side does not support it. (closes
          issue #11729) Reported by: tsearle Patches:
          new_codec_patch_udiff.patch uploaded by tsearle (license 373)

        * res/res_agi.c: If the channel is hungup during RECORD FILE send a
          result code of -1 to be uniform with everything else. (closes
          issue #11743) Reported by: davevg Patches: res_agi.diff uploaded
          by davevg (license 209)

2008-01-11 19:10 +0000 [r98315]  Mark Michelson <mmichelson@digium.com>

        * main/channel.c: Properly report the hangup cause as no answer
          when someone does not answer (closes issue #10574, reported by
          boch, patched by moy)

2008-01-11 18:25 +0000 [r98266]  Tilghman Lesher <tlesher@digium.com>

        * codecs/gsm/Makefile: Add another exception (which doesn't work)
          for -march optimization flag. Reported by: thomasmebes Patch by:
          tilghman (Closes issue #11563)

2008-01-11 18:25 +0000 [r98265]  Russell Bryant <russell@digium.com>

        * doc/security.txt, main/asterisk.c, configure,
          include/asterisk/autoconfig.h.in, main/Makefile, configure.ac,
          makeopts.in: Backport the ability to set the ToS bits on Linux
          when not running as root. Normally, we would not backport
          features into 1.4, but, I was convinced by the justification
          supplied by the supplier of this patch. He pointed out that this
          patch removes a requirement for running as root, thus reducing
          the potential impacts of security issues. (closes issue #11742)
          Reported by: paravoid Patches: libcap.diff uploaded by paravoid
          (license 200)

2008-01-11 17:22 +0000 [r98219]  Joshua Colp <jcolp@digium.com>

        * apps/app_followme.c: Ensure the return value of ast_bridge_call
          is passed back up as the application return value. This is needed
          for transfers to function so the PBX core knows to continue
          execution. (closes issue #10327) Reported by: kkiely

2008-01-11 15:52 +0000 [r98164]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_sip.c: Back out changes from revision 97077, since
          it wasn't perfect

2008-01-11 03:39 +0000 [r97976-98082]  Russell Bryant <russell@digium.com>

        * main/frame.c: Fix samples vs. length calculations for g722

        * main/translate.c: Simplify this code with a suggestion from Luigi
          on the asterisk-dev list. Instead of using is16kHz(), implement a
          format_rate() function.

        * main/translate.c: Fix various timing calculations that made
          assumptions that the audio being processed was at a sample rate
          of 8 kHz.

2008-01-10 23:08 +0000 [r97973]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_sip.c, main/translate.c: 1) When we get a
          translated frame out, clone it, because if the translator pvt is
          freed before we use the frame, bad things happen. 2) Getting a
          failure from ast_sched_delete means that the schedule ID is
          currently running. Don't just ignore it. (Closes issue #11698)

2008-01-10 21:57 +0000 [r97925]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Let us leave a voicemail for ourself if we
          have logged into VoiceMailMain and chosen to leave a message.
          (closes issue #11735, reported and patched by jamessan)

2008-01-10 21:37 +0000 [r97849-97889]  Steve Murphy <murf@digium.com>

        * pbx/ael/ael_lex.c, pbx/Makefile, pbx/ael/ael.flex: Applied the
          same fixes for ael.flex as was done in 97849 for ast_expr2.fl;
          overrode the normally generate yyfree func with our own version
          that checks the pointer for non-null before passing to free().
          Also takes care of a little problem with 2.5.33 and the use of
          the __STDC_VERSION__ macro.

        * main/ast_expr2.fl, main/Makefile, main/ast_expr2f.c: This is a
          fix for 2 things: a problem Terry was having in OSX with null
          pointers, which was my fault, as I probably forgot to run the sed
          script last time I made mods. So, I moved the fix into the flex
          input itself. Then, I found when I used flex 2.5.33, that it was
          using __STDC_VERSION__, and that's not real good; so I added back
          in a DIFFERENT sed script to fix that little mess. Tested
          everything, a couple different ways. Hope I did no harm, at the
          least.

2008-01-10 20:12 +0000 [r97847]  Jason Parker <jparker@digium.com>

        * include/asterisk/frame.h: Fix a comment that is no longer true.

2008-01-10 16:19 +0000 [r97734-97753]  Russell Bryant <russell@digium.com>

        * pbx/pbx_kdeconsole.h (removed), configs/modules.conf.sample,
          pbx/kdeconsole_main.cc (removed): Remove other remnants of
          pbx_kdeconsole

        * pbx/pbx_kdeconsole.cc (removed), build_tools/menuselect-deps.in,
          configure, include/asterisk/autoconfig.h.in, configure.ac,
          makeopts.in: Remove pbx_kdeconsole from the tree. It hasn't
          worked in ages, and nobody has complained. (closes issue #11706,
          reported by caio1982)

2008-01-10 15:07 +0000 [r97697]  Joshua Colp <jcolp@digium.com>

        * funcs/func_groupcount.c: Don't try to copy the category from the
          group if no category exists. (closes issue #11724) Reported by:
          IgorG Patches: group_count.v1.patch uploaded by IgorG (license
          20)

2008-01-09 23:01 +0000 [r97640-97645]  Russell Bryant <russell@digium.com>

        * pbx/pbx_gtkconsole.c: Strip terminal sequences from the verbose
          messages

        * pbx/pbx_gtkconsole.c: Make pbx_gtkconsole build ... but doesn't
          actually load on my system still (related to issue #11706)

2008-01-09 20:28 +0000 [r97618-97622]  Jason Parker <jparker@digium.com>

        * main/cli.c: Correctly display a message if a command could not be
          found. Also fix a comment which may have led to this happening.
          Issue 11718, reported by kshumard.

        * main/cli.c: Fix some locking and return value funkiness. We
          really shouldn't be unlocking this lock inside of a function,
          unless we locked it there too.

2008-01-09 18:48 +0000 [r97575]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Part 2 of app_queue doxygen improvements. Some
          smaller functions this time

2008-01-09 18:02 +0000 [r97529]  Russell Bryant <russell@digium.com>

        * res/res_features.c: Fix saying the parking space number to the
          caller doing the parking ...

2008-01-09 17:21 +0000 [r97491]  Kevin P. Fleming <kpfleming@digium.com>

        * codecs/codec_zap.c: report the same message whether Zaptel does
          not have transcoder support loaded or no transcoders were found

2008-01-09 16:44 +0000 [r97489]  Philippe Sultan <philippe.sultan@gmail.com>

        * channels/chan_gtalk.c: Set the caller id within the gtalk_alloc
          function. As underlined in issue #10437 by Josh, we need to
          prevent a possible memory leak. We only set the name part of the
          caller id, the number part is not relevant when dealing with
          JIDs. Closes issue #11549.

2008-01-09 16:11 +0000 [r97450]  Joshua Colp <jcolp@digium.com>

        * apps/app_meetme.c: Don't do conferencing totally in Zaptel if
          Monitor is running on the channel. (closes issue #11709) Reported
          by: BigJimmy Patches: patch-meetmerec uploaded by BigJimmy
          (license 371)

2008-01-09 15:43 +0000 [r97410-97448]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_zap.c: pass the right variable to get an error
          string... oops

        * channels/chan_zap.c: add error number output to ioctl failure
          messages to help with debugging

2008-01-09 00:44 +0000 [r97350]  Tilghman Lesher <tlesher@digium.com>

        * main/cli.c, main/editline/readline.c: Allow filename completion
          on zero-length modules, remove a memory leak, remove a file
          descriptor leak, and make filename completion thread-safe.
          Patched and tested by tilghman. (Closes issue #11681)

2008-01-09 00:17 +0000 [r97206-97308]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: use the \retval doxygen command properly

        * apps/app_queue.c: Part 1 of N of adding doxygen comments to
          app_queue. I picked some of the most common functions used (which
          also happen to be some the biggest/ugliest functions too) to
          document first. I'm pretty new to doxygen so criticism is
          welcome.

        * apps/app_queue.c: Some coding guidelines-related cleanup

2008-01-08 20:48 +0000 [r97195]  Joshua Colp <jcolp@digium.com>

        * channels/chan_mgcp.c: Fix various DTMF issues in chan_mgcp.
          (closes issue #11443) Reported by: eferro Patches:
          dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license
          337)

2008-01-08 20:47 +0000 [r97194]  Tilghman Lesher <tlesher@digium.com>

        * main/autoservice.c, main/utils.c: Increase constants to where
          we're less likely to hit them while debugging. (Closes issue
          #11694)

2008-01-08 20:42 +0000 [r97192]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Making some changes designed to not allow
          for a corrupted mailstream for a vm_state. 1. Add locking to the
          vm_state retrieval functions so that no linked list corruption
          occurs. 2. Make sure to always grab the persistent vm_state when
          mailstream access is necessary. 3. Correct an incorrect return
          value in the init_mailstream function. (closes issue #11304,
          reported by dwhite)

2008-01-08 19:53 +0000 [r97093-97152]  Joshua Colp <jcolp@digium.com>

        * funcs/func_groupcount.c: If no group has been provided to the
          GROUP_COUNT dialplan function then use the first one specific to
          the channel. (closes issue #11077) Reported by: m4him

        * apps/app_queue.c: Make app_queue calls work with directed pickup.
          (closes issue #11700) Reported by: jbauer

2008-01-08 18:02 +0000 [r97077]  Tilghman Lesher <tlesher@digium.com>

        * main/asterisk.c, channels/chan_sip.c: Apply multiple crash fixes,
          found in issue #11386, but not completely closing that issue.

2008-01-07 20:47 +0000 [r96884-96932]  Russell Bryant <russell@digium.com>

        * configs/extensions.conf.sample, /: Merged revisions 96931 via
          svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07 Jan 2008) |
          2 lines Change misery.digium.com to pbx.digium.com ........

        * res/res_smdi.c: Don't crash if something happens when setting up
          an SMDI interface and it gets destroyed before the SMDI port
          handling thread gets created.

2008-01-07 14:34 +0000 [r96797-96815]  Philippe Sultan <philippe.sultan@gmail.com>

        * res/res_jabber.c: Indentation fix, makes the code easier to read

        * res/res_jabber.c: Compute the base64 value over the
          [authzid]\0authcid\0password string, thus excluding the trailing
          NULL byte. This change has already been committed to trunk, see
          #11644.

2008-01-05 02:09 +0000 [r96644]  Russell Bryant <russell@digium.com>

        * main/devicestate.c: Don't pass an empty string as the device
          name.

2008-01-04 23:03 +0000 [r96575]  Tilghman Lesher <tlesher@digium.com>

        * main/devicestate.c: Fix the problem of notification of a device
          state change to a device with a '-' in the name. Could probably
          do with a better fix in trunk, but this bug has been open way too
          long without a better solution. Reported by: stevedavies Patch
          by: tilghman (Closes issue #9668)

2008-01-04 22:55 +0000 [r96573]  Jason Parker <jparker@digium.com>

        * res/res_features.c: Properly continue in the dialplan if using
          PARKINGEXTEN and the slot is full. Issue 11237, patch by me.

2008-01-04 19:27 +0000 [r96525]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_sip.c: If you change the bindaddr in sip.conf to a
          non-bound address and reload, sip goes kablooie. Reported and
          patched by: one47 (Closes issue #11535)

2008-01-04 16:19 +0000 [r96394-96449]  Russell Bryant <russell@digium.com>

        * channels/chan_zap.c: Make use of the temporary channel pointer
          while the pvt is unlocked. (closes issue #11675) Reported by:
          flefoll Patches: chan_zap.c.patch-store-owner-before-unlock
          uploaded by flefoll (license 244)

        * channels/chan_iax2.c: Don't crash if the iax2 pvt structure has
          been destroyed before we get to this point (closes issue #11672,
          reported by snuffy, patched by me)

2008-01-03 21:37 +0000 [r96318]  Tilghman Lesher <tlesher@digium.com>

        * res/res_config_pgsql.c: Missed initialization caused crash.
          Reported and fixed by: tiziano (Closes issue #11671)

2008-01-03 12:12 +0000 [r96198-96199]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c: make sure frame is completely clean,
          before we send it to asterisk as DTMF. If we don't make it clean,
          it happens that one way audio occurs..

        * channels/chan_misdn.c: when overlapdial was used and no number
          was dialed, the call was dropped, now we just jump into the s
          extension, which makes a lot more sense.

2008-01-02 23:46 +0000 [r96102]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: We need to reset the membername to NULL on each
          iteration of this loop, otherwise the result is that multiple
          members can have the same name, since the variable was not reset
          on each iteration of the loop.

2008-01-02 22:14 +0000 [r96020-96024]  Russell Bryant <russell@digium.com>

        * pbx/pbx_config.c: Convert locks of the contexts list in
          pbx_config to the appropriate rdlock or wrlock

        * pbx/pbx_dundi.c: pbx_dundi only needs a rdlock on the contexts
          list.

        * apps/app_macro.c: app_macro only needs a rdlock on the contexts
          list.

2008-01-02  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.17 released.

2008-01-02 20:24 +0000 [r95946]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Allocate a SIP refer structure when
          performing a transfer using BYE with Also so that the transfer
          information is properly stored. (AST-2008-028) (closes issue
          #11637) Reported by: greyvoip

2008-01-02 17:51 +0000 [r95890]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: A change to improve the accuracy of queue
          logging in the case where a member does not answer during the
          specified timeout period. Prior to this change, there was a small
          chance that the member name recorded in this case would be blank.
          Also prior to this change, if using the ringall strategy, if no
          one answered the call during the specified timeout, the member
          name listed in the queue log would randomly be one of the members
          that was rung. (closes issue #11498, reported and tested by
          hloubser, patched by me)

2007-12-31 23:43 +0000 [r95577]  Mark Michelson <mmichelson@digium.com>

        * main/pbx.c: Avoiding a potentially bad locking situation.
          ast_merge_contexts_and_delete writelocks the conlock, then calls
          ast_hint_extension, which attempts to readlock the same lock.
          Recursion with read-write locks is dangerous, so the inner lock
          needs to be removed. I did this by copying the "guts" of
          ast_hint_extension into ast_merge_contexts_and_delete (sans the
          extra lock). (this change is inspired by the locking problems
          seen in issue #11080, but I have no idea if this is the
          problematic area experienced by the reporters of that issue)

2007-12-31 20:27 +0000 [r95470]  Tilghman Lesher <tlesher@digium.com>

        * funcs/func_env.c: Allow the default "0" to be returned if the
          STAT fails (Closes issue #11659)

2007-12-28 18:24 +0000 [r95191]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: Remove duplicate increment of the header
          count in the add_header() function. (closes issue #11648)
          Reported by: makoto Patch provided by sergee, committed patch by
          me, inspired by comments from putnopvut

2007-12-28 00:16 +0000 [r95095]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: I found a bug while browsing the queue code and
          managed to reproduce it in a small setup. If a queue uses the
          ringall strategy, it was possible through unfortunate coincidence
          for a single member at a given penalty level to make app_queue
          think that all members at that penalty level were unavailable and
          cause the members at the next penalty level to be rung. With this
          patch, we will only move to the next penalty level if ALL the
          members at a given penalty level are unreachable.

2007-12-27 21:40 +0000 [r95024]  Russell Bryant <russell@digium.com>

        * main/channel.c: Don't report a syntax error when an empty string
          is passed to ast_get_group. Just return 0. (closes issue #11540)
          Reported by: tzafrir Patches: group_empty.diff uploaded by
          tzafrir (license 46) -- slightly changed by me

2007-12-27 20:09 +0000 [r94977]  Mark Michelson <mmichelson@digium.com>

        * main/io.c: Fixing a typo in a comment.

2007-12-27 17:32 +0000 [r94905-94924]  Joshua Colp <jcolp@digium.com>

        * channels/chan_h323.c: Include types.h in chan_h323 as without it
          it can not be compiled on some operating systems like FreeBSD to
          name one. (closes issue #11585) Reported by: sobomax Patches:
          chan_h323.c.diff uploaded by sobomax (license 359)

        * channels/chan_sip.c: Use ast_strlen_zero to see if our_contact is
          set or not on the dialog. It is possible for it to be a pointer
          to NULL. (closes issue #11557) Reported by: FuriousGeorge

2007-12-27 15:16 +0000 [r94828-94831]  Russell Bryant <russell@digium.com>

        * main/pbx.c: Now that the contexts lock is a read/write lock, it
          should not be locked here in ast_hint_state_changed(). This makes
          it get locked recursively which now causes a deadlock. (closes
          issue #11080, thanks to callguy for the access to a deadlocked
          machine)

        * include/asterisk/translate.h, main/translate.c: Use the constant
          that I really meant to use here ...

        * main/translate.c: Change ast_translator_best_choice() to only pay
          attention to audio formats. This fixes a problem where Asterisk
          claims that a translation path can not be found for channels
          involving video. (closes issue #11638) Reported by: cwhuang
          Tested by: cwhuang Patch suggested by cwhuang, with some
          additional changes by me.

2007-12-27 01:01 +0000 [r94824]  Kevin P. Fleming <kpfleming@digium.com>

        * main/manager.c: make this comment explain the situation in an
          even more explicit fashion

2007-12-26 20:43 +0000 [r94808]  Tilghman Lesher <tlesher@digium.com>

        * main/manager.c: Workaround for what is probably a glibc bug (but
          we'll see this crop up again and again, if we don't add the
          workaround). Reported by: rolek Patch by: tilghman (Closes issue
          #11601, closes issue #11426)

2007-12-26 19:04 +0000 [r94789-94801]  Russell Bryant <russell@digium.com>

        * main/autoservice.c: Just in case the AST_FLAG_END_DTMF_ONLY flag
          was already set before starting autoservice, remember it and
          ensure that the channel has the same setting when autoservice
          gets stopped. (pointed out by d1mas, patched up by me)

        * main/autoservice.c: When a channel is in autoservice, mark a flag
          on the channel that says that we only care about the END of a
          digit. That way, no magic digit emulation stuff will happen when
          all we're doing is queueing up END frames.

        * res/res_features.c: Don't try to send a parked call back to
          itself. (closes issue #11622, reported by djrodman, patched by
          me)

        * main/autoservice.c: Don't store DTMF BEGIN frames while a channel
          is in autoservice. It's just going to make ast_read() do a lot of
          extra work when the channel comes back out of autoservice.
          (closes issue #11628, patched by me)

        * Makefile: List include/asterisk/version.h as a .PHONY target
          because we want the commands listed for this target to be
          executed regardless of whether the file exists or not. This fixes
          having the version not up to date when running from svn. (closes
          issue #11619, reported by plack, fixed by me)

2007-12-25 02:27 +0000 [r94769]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: file says... build on the builders.

2007-12-24 19:36 +0000 [r94763-94767]  Tilghman Lesher <tlesher@digium.com>

        * main/channel.c: Race: we need to wait to queue a NewChannel event
          until after the channel is inserted into the channel list. The
          reason is because some manager users immediately queue requests
          from the channel when they see that event and are confused when
          Asterisk reports no such channel. (Closes issue #11632)

        * channels/chan_sip.c: More deadlock avoidance code (this time
          between sip_monitor and sip_hangup) Reported by: apsaras Patch
          by: tilghman (Closes issue #11413)

        * channels/chan_sip.c: Another bit of bad logic in realtime_peer
          Reported by: dimas Patch by: dimas (Closes issue #11631)

2007-12-23 01:21 +0000 [r94660]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_sip.c: Argh... I suppose third time's the charm.

2007-12-21 20:21 +0000 [r94468-94543]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Bunch of coding guidelines cleanup

        * apps/app_voicemail.c: Better quota support for using IMAP storage
          voicemail (closes issue #11415, reported by jaroth) (closes issue
          #11152, reported by selsky) Patch provided by jaroth

        * apps/app_voicemail.c: The mail_copy c-client function does not
          expect a full imap mailbox string, just the name of the mailbox.
          (closes issue #11419, reported and patched by jaroth, with
          additional patchwork from me)

        * main/dial.c: Since we are freeing list elements within a list
          traversal, we need to use the safe traversal and remove the item
          from the list before freeing it. (closes issue 11612, reported by
          dtyoo)

2007-12-21 16:37 +0000 [r94466]  Russell Bryant <russell@digium.com>

        * main/pbx.c, include/asterisk/pbx.h: Convert the contexts lock to
          a read/write lock to resolve a deadlock. This has a nice side
          benefit of improving performance. :) (closes issue #11609)
          (closes issue #11080)

2007-12-21 16:11 +0000 [r94420-94464]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Removing a debug message I accidentally just
          committed

        * main/say.c, apps/app_queue.c: Fixing Portuguese syntax for saying
          dates and times. Also some coding guidelines cleanup. (closes
          issue #11599, reported and patched by caio1982, coding guidelines
          cleanup by me)

2007-12-21 15:07 +0000 [r94418]  Tilghman Lesher <tlesher@digium.com>

        * main/asterisk.c: Fix for restart-as-user problem reported via the
          -dev list

2007-12-20  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.16.2 released.

2007-12-20 20:22 +0000 [r94215-94256]  Russell Bryant <russell@digium.com>

        * /, channels/chan_iax2.c: Merged revisions 94255 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r94255 | russell | 2007-12-20 14:21:41 -0600 (Thu, 20 Dec 2007) |
          5 lines Fix another potential seg fault ... (closes issue #11606)
          Reported by: dimas ........

        * channels/chan_zap.c: Fix a deadlock in d-channel handling in
          chan_zap. This deadlock was introduced by the fix to ensure that
          channels are properly locked when handling channel variables.
          There were sections of this code where the channel pvt was locked
          before the channel lock, when in fact it _must_ be the other way
          around. (closes issue #11582) Reported by: bugi

2007-12-19 23:02 +0000 [r94122]  Mark Michelson <mmichelson@digium.com>

        * res/res_monitor.c: Sox versions 13.0.0 and newer do not have
          "soxmix" and instead use sox -m. res_monitor needs to use this if
          the user does not have soxmix. (closes issue #11589, reported by
          amessina, patch inspired by amessina but with a flourish from me)

2007-12-19 22:48 +0000 [r94077]  Russell Bryant <russell@digium.com>

        * configure, include/asterisk/autoconfig.h.in, configure.ac: Check
          for the existence of the soxmix application on the target
          platform and have the result available in autoconfig.h. (part of
          issue #11589)

2007-12-19  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.16.1 released.

2007-12-19 17:29 +0000 [r93955]  Joshua Colp <jcolp@digium.com>

        * channels/chan_iax2.c: Make the 1.4 builders happy, ensure var is
          NULL.

2007-12-19 17:04 +0000 [r93949]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_iax2.c: Avoid segfault in chan_iax when peer isn't
          defined (Closes issue #11602)

2007-12-18 22:42 +0000 [r93764]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: FreeBSD also does not have byte swap
          functions. Issue 11586, patch by sobomax.

2007-12-18  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.16 released.

2007-12-18 18:45 +0000 [r93668-93676]  Tilghman Lesher <tlesher@digium.com>

        * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
          93667 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18 Dec 2007)
          | 2 lines Fixing AST-2007-027 (Closes issue #11119) ........

2007-12-18 17:02 +0000 [r93625]  Mark Michelson <mmichelson@digium.com>

        * main/channel.c: Rework deadlock avoidance used in ast_write,
          since it meant that agent channels which were being monitored had
          one audio file recorded and one empty audio file saved. (closes
          issue #11529, reported by atis patched by me)

2007-12-17 22:56 +0000 [r93381-93420]  Jason Parker <jparker@digium.com>

        * main/translate.c: What was I thinking when I wrote this
          masterpiece? -1 + 1 = 0.. who woulda thunk it?.

2007-12-17 22:28 +0000 [r93377]  Joshua Colp <jcolp@digium.com>

        * main/utils.c: Do not try to access information about a lock when
          printing out a trylock attempt. It is possible for the lock that
          it references to no longer be valid. This would have caused
          segfaults or deadlocks. (issue #BE-263) (closes issue #11080)
          Reported by: callguy (closes issue #11100) Reported by: callguy

2007-12-17 21:12 +0000 [r93336]  Tilghman Lesher <tlesher@digium.com>

        * include/asterisk/time.h: Today is tomorrow's yesterday, and
          yesterday's tomorrow is today, and tomorrow's tomorrow is the day
          after tomorrow, so who cares if you recycle anyway? If this
          confuses you, that's nothing compared to what this fixes. ;-)

2007-12-17 19:53 +0000 [r93291]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: We need to create the directory for a
          voicemail user even if they are using IMAP storage since
          greetings are stored in the filesystem. (closes issue #11388,
          reported by spditner, patch by me inspired by a patch by
          spditner)

2007-12-17 18:05 +0000 [r93250]  Joshua Colp <jcolp@digium.com>

        * channels/chan_zap.c: If a call is received with a called number
          IE containing nothing go to the 's' extension. (closes issue
          #9099) Reported by: kb1_kanobe2 Patches: 20070906__9099.diff.txt
          uploaded by Corydon76 (license 14)

2007-12-17 07:21 +0000 [r93183]  Kevin P. Fleming <kpfleming@digium.com>

        * funcs/Makefile, codecs/Makefile, cdr/Makefile, pbx/Makefile,
          res/Makefile, channels/Makefile, formats/Makefile: fix some
          copy-and-paste leftovers

2007-12-17 07:15 +0000 [r93182]  Olle Johansson <oej@edvina.net>

        * channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c,
          apps/app_queue.c, channels/chan_iax2.c: Issue 11574: Add
          dependencies on res_monitor and res_features. I wonder if
          Asterisk can run at all without res_features. My guess is that
          there's propably a lot of more modules and the core that depends
          on it. Reported by: caio1982 (closes issue #11574)

2007-12-17 06:44 +0000 [r93180]  Kevin P. Fleming <kpfleming@digium.com>

        * formats, Makefile, codecs/Makefile, funcs, apps/Makefile,
          configure, cdr/Makefile, build_tools/prep_tarball, makeopts.in,
          formats/Makefile, pbx, res, channels, funcs/Makefile, codecs,
          include/asterisk/autoconfig.h.in, build_tools/make_version, apps,
          configure.ac, Makefile.moddir_rules, build_tools/prep_moduledeps
          (removed), res/Makefile, pbx/Makefile, cdr, channels/Makefile: In
          http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,
          rizzo brought up some issues related to the way that the metadata
          required for menuselect and the rest of the build system is
          extracted from the source files. Since I had a few hours to kill
          on an airplane today, I decided to improve this situation... so
          now the system caches the extracted metadata and uses it to build
          the menuselect 'tree' as much as it can. The result of this is
          that when a single source file is changed, only the metadata for
          that file needs to be extracted again, and the rest is used from
          the cache files. I also reduced the number of forked processes
          required to do the metadata extraction; it was actually possible
          to do most of what we needed in the Makefiles themselves without
          using any shell scripts at all! On my laptop, these changes
          resulted in an 80% decrease in the time required for the
          'menuselect.makeopts' automatic check to occur after editing a
          single source file. While doing this work I also cleaned up a few
          minor things in the Makefiles, adding a check for 'awk' to the
          configure script and changed all remaining places we use 'grep'
          or 'awk' to use the ones found by the configure script, and
          changed the 'prep_tarball' script to build the menuselect
          metadata so that tarballs of Asterisk will include it and won't
          require the user to wait while it is extracted after unpacking.

2007-12-14 17:36 +0000 [r93000]  Russell Bryant <russell@digium.com>

        * main/config.c: There are a lot of existing systems that #include
          non-existent files. So, to make the transition to treating this
          as an error a bit less painless, just issue a huge error message
          for now. Then, later, we can reinstate the code that treats it as
          a failure. (Thanks to philippel for the feedback)

2007-12-14 15:16 +0000 [r92937]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Up the length of the format on the SIP
          channel since it can now be rather long. (closes issue #11552)
          Reported by: francesco_r

2007-12-14 15:05 +0000 [r92934]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c: fixed the sequencing of WAITING_4DIGS
          state setting and overlap_task thread starting.

2007-12-14 15:01 +0000 [r92933]  Tilghman Lesher <tlesher@digium.com>

        * res/res_agi.c: Change help documentation to match actual behavior
          (FAILURE vs FAILED). Reported by: angeloxx-sir Patch by: tilghman
          (Closes issue #11548)

2007-12-14 01:24 +0000 [r92875]  Mark Michelson <mmichelson@digium.com>

        * include/asterisk/lock.h: When compiling with DETECT_DEADLOCKS,
          don't spam the CLI with messages about possible deadlocks.
          Instead just print the intended single message every five
          seconds. (closes issue 11537, reported and patched by dimas)

2007-12-13 21:28 +0000 [r92815]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_zap.c: Properly initialize polarity statuses, so
          that they are detected properly. Reported by: julianjm Patch by:
          julianjm (Closes issue #10238)

2007-12-13 20:13 +0000 [r92809]  Jason Parker <jparker@digium.com>

        * main/pbx.c: Make application help text a little more clear about
          the use of extensions in a filename.

2007-12-13 20:03 +0000 [r92803-92807]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Prevent another potential fd leak

        * apps/app_voicemail.c: Prevent a possible fd leak.

2007-12-13 00:11 +0000 [r92696]  Jason Parker <jparker@digium.com>

        * main/config.c, channels/chan_sip.c, channels/chan_h323.c,
          channels/chan_iax2.c: If a typo is found in a config file, we
          previous continued on with what was already loaded. We do not
          want to do this (see bug below for details). This makes it so
          that if a [ is found without a ], the entire config will fail,
          and nothing in it will be loaded. Isue #10690.

2007-12-12 22:00 +0000 [r92656]  Kevin P. Fleming <kpfleming@digium.com>

        * codecs/codec_zap.c: emit a warning message when we drop a G.729B
          CNG frame destined for the transcoder

2007-12-12 21:15 +0000 [r92617]  Jason Parker <jparker@digium.com>

        * apps/app_meetme.c: Don't increment user count until after name
          has been recorded (if enabled). Issue 11048, tested by pep.

2007-12-12 19:40 +0000 [r92556]  Russell Bryant <russell@digium.com>

        * res/res_features.c: resolve compiler warning

2007-12-12 17:46 +0000 [r92510]  Mark Michelson <mmichelson@digium.com>

        * res/res_features.c: Correctly detect where a dynamic feature was
          activated. Before this patch, the channel which initiated the
          bridge was always assumed to have been the one which activated
          the dynamic feature. This patch corrects this. (closes issue
          #11529, reported and patched by nic_bellamy)

2007-12-12 16:52 +0000 [r92463]  Tilghman Lesher <tlesher@digium.com>

        * configure, include/asterisk/autoconfig.h.in, configure.ac: Test
          directly for the API that fixed AST-2007-026, to ensure that
          older versions of PostgreSQL are no longer acceptable. (Closes
          issue #11526)

2007-12-12 16:08 +0000 [r92443]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Removing an unused variable.

2007-12-11 19:51 +0000 [r92363]  Joshua Colp <jcolp@digium.com>

        * main/global_datastores.c: Fix potential memory leak with the
          dialed interfaces list if another memory allocation fails.
          (closes issue #11507) Reported by: eliel Patches:
          global_datastores.c.patch uploaded by eliel (license 64)

2007-12-11 17:42 +0000 [r92323]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Fixing autofill to be more accurate.
          Specifically, if calls ahead of the current caller were ringing
          members (but not yet bridged) there could be available members
          and waiting callers who would not get matched up. The member
          availability checker was correctly determining the number of
          available members in this scenario, but the queue itself did not
          parallelly reflect this status on the pending calls. This commit
          corrects the issue. (closes issue #11459, reported by
          equissoftware, patched by me)

2007-12-10 16:36 +0000 [r92204]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: Add G729A as another possible payload name for G729.
          Some devices use this instead of G729, which is perfectly normal
          since the payload number itself is defined and can't be used by
          anything else so the name doesn't matter that much. (closes issue
          #11483) Reported by: revolution Patches: rtp.diff uploaded by
          revolution (license 346)

2007-12-10 16:29 +0000 [r92202]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: If there are no members in a queue, then the
          loop where the datastore for detecting duplicate dialed numbers
          will be skipped, meaning the datastore isn't created. This means
          that when we try to free it, there's a crash. This stops that
          crash from occurring. (closes issue #11499, reported by slavon,
          patched by eliel)

2007-12-10 16:13 +0000 [r92200]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: It is possible for nativeformats to contain
          more then one codec, so print out multiple ones. (closes issue
          #11366) Reported by: ovi

2007-12-10 14:04 +0000 [r92158]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Avoid reinvite race situations with two
          Asterisks trying to reinvite each other in 1.4 and trunk. This
          patch implements support for the 491 error code that Asterisk 1.4
          generates on situations where we get an incoming INVITE and
          already has one in progress. Thanks to mavetju for reporting and
          to Raj Jain for an excellent explanation of the problem. Patch by
          myself. Tested with 8 Asterisk servers connected to each other in
          a training network. Closes issue #10481

2007-12-07 23:29 +0000 [r91890]  Jason Parker <jparker@digium.com>

        * main/dsp.c: We need to make sure we free the input frame if we
          return a different frame in ast_dsp_process. Issue 11273, pointed
          out by dimas, with a patch by eliel.

2007-12-07 22:30 +0000 [r91870]  Kevin P. Fleming <kpfleming@digium.com>

        * codecs/codec_zap.c: even though Asterisk explicitly requests that
          endpoints using G.729 do *not* use Annex B (silence detection and
          comfort noise generation) some do anyway; the transcoder card
          interface does not currently work properly with CNG frames, so
          trim off the CNG before sending the data

2007-12-07 21:24 +0000 [r91777-91830]  Russell Bryant <russell@digium.com>

        * main/utils.c: Make the lock protecting each thread's list of
          locks it currently holds recursive. I think that this will fix
          the situation where some people have said that "core show locks"
          locks up the CLI. (related to issue #11080)

        * include/asterisk/lock.h: Fix another bug in the DEBUG_THREADS
          code. The ast_mutex_init() function had the mutex attribute
          object marked as static. This means that multiple threads
          initializing locks at the same time could step on each other and
          end up with improperly initialized locks. (found when tracking
          down locking issues related to issue #11080)

        * include/asterisk/lock.h: I love fixing lock related errors in the
          lock debugging code. That's about as ironic as it gets in
          Asterisk programming land. Anyway, I spotted this bug while
          trying to track down why systems are locking up and acting weird
          in issue #11080. The mutex attribute object was marked as static
          in this function when it should not have been.

        * apps/app_dial.c: * Add channel locking around datastore
          operations that expect the channel to be locked. * Document why
          we don't record Local channels in the dialed interfaces list. *
          Remove the dialed variable as it isn't needed. * Restructure some
          code for clarity and coding guidelines stuff

        * apps/app_queue.c: * Add channel locking around datastore
          operations that expect the channel to be locked. * Document why
          we don't record Local channels in the dialed interfaces list. *
          Handle memory allocation failure. * Remove the dialed variable,
          as it wasn't actually needed. * Tweak some formatting to conform
          to coding guidelines.

        * main/autoservice.c: * Add a bit more of a verbose comment as to
          why a hangup frame needs to be queued up if autoservice gets a
          NULL return from ast_read(). * Make the process of queueing the
          hangup frame more efficient by putting the frame where it is
          going to end up and avoiding some locking and extra memory
          allocations and freeing.

2007-12-07 15:39 +0000 [r91737]  Mark Michelson <mmichelson@digium.com>

        * main/autoservice.c: Hangups that happen during autoservice were
          not processed appropriately. This is because a hangup actually
          causes a NULL frame to be received, not a hangup frame. Queueing
          a hangup if we receive a NULL frame during autoservice corrects
          this problem (closes issue #11467, reported by jmls, patched by
          me)

2007-12-07 02:51 +0000 [r91675-91693]  Russell Bryant <russell@digium.com>

        * apps/app_dial.c: Don't unlock the dialed_interfaces list until
          we're done messing with the iterator.

        * apps/app_dial.c, apps/app_queue.c: Allow dialing local channels
          from Queue() and Dial() again. There was a slight flaw in the
          code to prevent call forwards from looping that caused this
          problem. (related to issue #11486)

        * apps/app_queue.c: Fix in an issue in the call forwarding handling
          code that was causing crashes on every call into a queue. I'm not
          entirely sure about the logic in this part of the code, so I want
          to look at it some more tomorrow. However, this makes it safe and
          keeps it from crashing. (closes issue #11486, reported by adamg,
          patched by me)

2007-12-07 00:52 +0000 [r91637]  Tilghman Lesher <tlesher@digium.com>

        * main/rtp.c: At the end of a call, when we're reporting, RTCP may
          already be partially torn down, so check for NULL dereference
          Reported by: blitzrage Patch by: tilghman (Closes issue #11450)

2007-12-06 20:25 +0000 [r91541]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: IMAP storage did not honor the maxmsg
          setting in voicemail.conf, and it also had the possibility of
          crashing if a user had more than 256 messages in their voicemail.
          This patch kills two birds with one stone by adding maxmsg
          support and also setting a hard limit on the number of messages
          at 255 so that the crashes cannot happen. (closes issue #11101,
          reported by Skavin, patched by me)

2007-12-06 19:11 +0000 [r91501]  Russell Bryant <russell@digium.com>

        * main/loader.c, include/asterisk/module.h: Add a new module flag
          to indicate that a build sum is present. Modules built against
          older Asterisk 1.4 headers will now load properly with just a
          warning indicating that they are old and may cause problems.
          (patch by paravoid)

2007-12-06 16:49 +0000 [r91439-91450]  Joshua Colp <jcolp@digium.com>

        * main/udptl.c: Fix various in the udptl implementation. It could
          return empty modem frames, have an incorrect sequence number on
          packets, and display the wrong sequence number in the debug
          messages. (closes issue #11228) Reported by: Cache Patches:
          udptl-4.patch uploaded by dimas (license 88)

        * channels/chan_sip.c: Add support for accepting and sending T.38
          in the initial INVITE. (closes issue #9402) Reported by: thdei

2007-12-06 12:54 +0000 [r91366]  Olle Johansson <oej@edvina.net>

        * main/loader.c, include/asterisk/logger.h, main/logger.c: Make
          sure logger is reloaded at general reload in the cli. (Discovered
          during Asterisk training in Portugal)

2007-12-05 22:57 +0000 [r91273-91292]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Reverting extra stuff I didn't mean to
          commit

        * apps/app_voicemail.c, apps/app_dial.c: The 'G' option for Dial()
          did not properly handle the case where only a label was provided.
          This was due to the fact that the answering channel did not have
          an extension set, so ast_parseable_goto would fail. This fix
          eliminates the call to ast_parseable_goto on the answering
          channel since it is a wasteful call. The answering channel and
          the calling channel are both directed to the same extension and
          context, just different priorities, so we can just copy the
          values from the calling channel to the answering channel and
          increment the answering channel's priority. (closes issue #11382,
          reported by jon, patch by me with correction by jon)

2007-12-05 21:38 +0000 [r91237]  Tilghman Lesher <tlesher@digium.com>

        * sounds/Makefile: Upgrade to the latest version of extra sounds

2007-12-05 17:31 +0000 [r90967-91192]  Russell Bryant <russell@digium.com>

        * main/threadstorage.c: Make the lock in the threadstorage
          debugging code untracked to avoid a deadlock on thread
          destruction. (closes issue #11207) Reported by: ys Patches:
          threadstorage.c.diff uploaded by ys (license 281) Also fixes an
          open bug report: (closes issue #11446)

        * main/utils.c: When DEBUG_THREADS is enabled, we only have the
          details about who is holding a lock that we are waiting on for a
          mutex, not rwlocks. This should fix the problem where people have
          reported "core show locks" crashing sometimes.

        * include/asterisk/lock.h: Fix some crashes in chan_iax2 that were
          reported as happening on Mac systems. It turns out that the
          problem was the Mac version of the ast_atomic_fetchadd_int()
          function. The Mac atomic add function returns the _new_ value,
          while this function is supposed to return the old value. So, the
          crashes happened on unreferencing objects. If the reference count
          was decreased to 1, ao2_ref() thought that it had been decreased
          to zero, and called the destructor. However, there was still an
          outstanding reference around. (closes issue #11176) (closes issue
          #11289)

        * include/asterisk/file.h, configure,
          include/asterisk/autoconfig.h.in, configure.ac,
          include/asterisk/compiler.h: Modify file.h to maintain API
          compatibility with earlier versions. If a recent compiler is
          being used, then a warning will show up for any modules still
          using the old name "private" instead of "_private". (patch
          suggested by paravoid)

        * main/pbx.c: Make some changes to some additions I made recently
          for doing channel autoservice when looking up extensions. This
          code was added to handle the case where a dialplan switch was in
          use that could block for a long time. However, the way that I
          added it, it did this for all extension lookups. However, lookups
          in the in-memory tree of extensions should _not_ take long enough
          to matter. So, move the autoservice stuff to be only around
          executing a switch.

2007-12-04 17:28 +0000 [r90876]  Jason Parker <jparker@digium.com>

        * main/channel.c: If we fail to create a channel after allocating a
          timing fd, we need to make sure to close it. Issue 11454, patch
          by eliel.

2007-12-04 05:29 +0000 [r90798]  Joshua Colp <jcolp@digium.com>

        * apps/app_dial.c: Fix build issue on the build cluster.

2007-12-03 23:50 +0000 [r90736-90753]  Tilghman Lesher <tlesher@digium.com>

        * include/asterisk/compat.h: Solaris requires the inclusion of
          sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by:
          snuffy,tilghman (Closes issue #11430)

        * res/res_config_pgsql.c: If both dbhost and dbsock were not set, a
          NULL deref could result Reported by: xrg Patch by: tilghman
          (Closes issue #11387)

2007-12-03 23:12 +0000 [r90735]  Mark Michelson <mmichelson@digium.com>

        * apps/app_dial.c, main/channel.c, main/global_datastores.c
          (added), channels/chan_local.c, main/Makefile,
          include/asterisk/channel.h, include/asterisk/global_datastores.h
          (added), apps/app_queue.c: A big one... This is the merge of the
          forward-loop branch. The main change here is that call-forwards
          can no longer loop. This is accomplished by creating a datastore
          on the calling channel which has a linked list of all devices
          dialed. If a forward happens, then the local channel which is
          created inherits the datastore. If, through this progression of
          forwards and datastore inheritance, a device is attempted to be
          dialed a second time, it will simply be skipped and a warning
          message will be printed to the CLI. After the dialing has been
          completed, the datastore is detached from the channel and
          destroyed. This change also introduces some side effects to the
          code which I shall enumerate here: 1. Datastore inheritance has
          been backported from trunk into 1.4 2. A large chunk of code has
          been removed from app_dial. This chunk is the section of code
          which handles the call forward case after the channel has been
          requested but before it has been called. This was removed because
          call-forwarding still works fine without it, it makes the code
          less error-prone should it need changing, and it made this set of
          changes much less painful to just have the forwarding handled in
          one place in each module. 3. Two new files, global_datastores.h
          and .c have been added. These are necessary since the datastore
          which is attached to the channel may be created and attached in
          either app_dial or app_queue, so they need a common place to find
          the datastore info. This approach was taken in case similar
          datastores are needed in the future, there will be a common place
          to add them.

2007-12-03 22:06 +0000 [r90696]  Jason Parker <jparker@digium.com>

        * apps/app_meetme.c: Make sure we always close the conference fd if
          we have an open one. Issue 11383, reported by markmhy, patch by
          eliel.

2007-12-03 20:59 +0000 [r90639]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_mgcp.c: Changing some bad logic when calculating
          the interdigit timeout. (closes issue #11402, reported and
          patched by eferro)

2007-12-03 20:51 +0000 [r90607]  Jason Parker <jparker@digium.com>

        * res/res_features.c: Fix crash in ParkAndAnnounce application.
          Issue #11436, reported by lytledd, patch by eliel.

2007-12-03 20:05 +0000 [r90548-90588]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: Do not create a smoother for G723.1 frames, they need
          to be left alone to their native 20/24 byte size.

        * .cleancount, main/channel.c, include/asterisk/channel.h: Preserve
          the indication currently playing on a channel when a masquerade
          operation happens. (issue #BE-88)

2007-12-03 18:20 +0000 [r90546]  Jason Parker <jparker@digium.com>

        * channels/chan_iax2.c: Only log debug messages if debug is
          enabled. Closes issue #11416, patch by casper.

2007-12-02 18:18 +0000 [r90470]  Russell Bryant <russell@digium.com>

        * apps/app_queue.c: The other day when I went through making
          changes as a result of the ao2_link() change, I added some code
          to set pointers to NULL after they were unreferenced. This
          pointed out that in this place, the object was unreferenced
          before the code was done using it. So, move the unref down a
          little bit. (crash reported by jmls on IRC)

2007-12-02 09:34 +0000 [r90432]  Tilghman Lesher <tlesher@digium.com>

        * main/autoservice.c: Clarify the return value on autoservice.
          Specifically, if you started autoservice and autoservice was
          already on, it would erroneously return an error. Reported by:
          adiemus Patch by: dimas (Closes issue #11433)

2007-11-30 19:26 +0000 [r90310-90348]  Russell Bryant <russell@digium.com>

        * main/astobj2.c, main/manager.c, include/asterisk/astobj2.h,
          apps/app_queue.c, channels/chan_iax2.c: Change the behavior of
          ao2_link(). Previously, in inherited a reference. Now, it
          automatically increases the reference count to reflect the
          reference that is now held by the container. This was done to be
          more consistent with ao2_unlink(), which automatically releases
          the reference held by the container. It also makes it so it is no
          longer possible for a pointer to be invalid after ao2_link()
          returns.

        * include/asterisk/astobj2.h: Add some notes on the behavior of
          ao2_unlink() after a discussion with Tilghman

2007-11-30 14:43 +0000 [r90269]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Fix locking issues under one legged replaces
          scenarios. (closes issue #11420) Reported by: irroot Patches:
          chan_sip_oneleg.patch uploaded by irroot (license 52)

2007-11-30 00:16 +0000 [r90231]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_mgcp.c: Clear the DTMF buffer if the call times
          out. (closes issue #11418, reported and patched by eferro)

2007-11-29  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.15 released.

2007-11-29 19:48 +0000 [r90166]  Tilghman Lesher <tlesher@digium.com>

        * cdr/cdr_pgsql.c: Properly escape cdr->src and cdr->dst and ensure
          we use thread-safe escaping (Fixes AST-2007-026)

2007-11-29 19:38 +0000 [r90163]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: This patch handles the case where a queue
          member with a negative penalty is added via the manager. If a
          negative value is submitted for a member penalty, we set it to 0.
          (closes issue #11411, reported and patched by Laureano)

2007-11-29 19:24 +0000 [r90154-90160]  Tilghman Lesher <tlesher@digium.com>

        * res/res_config_pgsql.c: Properly escape input buffers (Fixes
          AST-2007-025)

        * formats/format_g726.c, include/asterisk/file.h,
          formats/format_wav.c, formats/format_pcm.c,
          formats/format_ogg_vorbis.c, main/file.c, formats/format_h263.c,
          formats/format_h264.c, formats/format_wav_gsm.c: Use of "private"
          as a field name in a header file messes with C++ projects
          Reported by: chewbacca Patch by: casper (Closes issue #11401)

        * sounds/Makefile: Upgrade the core sounds release version

2007-11-29 00:36 +0000 [r90142-90147]  Russell Bryant <russell@digium.com>

        * funcs/func_callerid.c: fix some formatting i accidentally changed

        * funcs/func_callerid.c, main/channel.c,
          include/asterisk/channel.h: This set of changes is to make some
          callerID handling thread-safe. The ast_set_callerid() function
          needed to lock the channel. Also, the handlers for the CALLERID()
          dialplan function needed to lock the channel when reading or
          writing callerid values directly on the channel structure.

        * include/asterisk/file.h, main/file.c: Merge a change from
          team/russell/chan_refcount ... This makes ast_stopstream()
          thread-safe.

2007-11-28 22:59 +0000 [r90101]  Joshua Colp <jcolp@digium.com>

        * apps/app_queue.c: Fix a few memory leaks. (closes issue #11405)
          Reported by: eliel Patches: load_realtime.patch uploaded by eliel
          (license 64)

2007-11-28 22:30 +0000 [r90098]  Kevin P. Fleming <kpfleming@digium.com>

        * configs/users.conf.sample, main/manager.c: it is impossible to
          set permissions for manager accounts created by users.conf
          (reported internally, patched by me)

2007-11-28 22:08 +0000 [r89999-90059]  Mark Michelson <mmichelson@digium.com>

        * main/pbx.c: Removing some seemingly pointless code. This sets a
          channel variable for every priority executed in the dialplan if
          you have debug set to anything non-zero. This seems pointless due
          to the fact that these channel variables are not referenced
          anywhere else in the code and their names are esoteric enough
          that they would not be practical to reference in the dialplan.
          Plus the fact that this behavior isn't documented anywhere means
          that the change is not likely to cause any disruption. If
          anything, this may actually cause a slight performance increase
          if running with debug on. The motivating influence for this code
          change is the eventwhencalled option for queues. If set to vars,
          all channel variables will be output to the manager. These
          unnecessary channel variables make the output a lot more
          difficult to deal with.

        * apps/app_voicemail.c: Recording greetings when using IMAP storage
          was causing zero-length files to be stored. Since greetings are
          not retrieved from IMAP anyway, it is pointless to attempt
          storing them there. (closes issue #11359, reported by spditner,
          patched by me)

2007-11-28 00:20 +0000 [r89839-89893]  Russell Bryant <russell@digium.com>

        * main/pbx.c, include/asterisk/pbx.h: - update documentation for
          some of the goto functions to note that they handle locking the
          channel as needed - update ast_explicit_goto() to lock the
          channel as needed

        * main/autoservice.c: Don't do frame processing if ast_read()
          returned NULL.

        * apps/app_queue.c: Instead of depending on the return value of
          ast_true(), explicitly set the eventwhencalled variable to 1.

        * main/pbx.c: Don't start/stop autoservice in
          pbx_extension_helper() unless a channel exists

2007-11-27 23:10 +0000 [r89837]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Two changes with regards to the
          'eventwhencalled' option of queues.conf 1) Due to some signed vs.
          unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes'
          did exactly the same thing. Thus the sign change of the ast_true
          call. 2) The vars2manager function overwrote a \n for every
          channel variable it parsed, resulting in bizarre output for the
          channel variables. This patch remedies this. (related to issue
          #11385, however I'm not sure if this will actually be enough to
          close it)

2007-11-27 21:45 +0000 [r89790]  Russell Bryant <russell@digium.com>

        * main/autoservice.c, main/pbx.c: Merge changes from
          team/russell/autoservice_1.4 This set of changes fixes an issue
          that was reported to me on IRC yesterday. The user, d1mas, was
          using chan_zap for incoming calls and was having DTMF recognition
          issues in some situations. Specifically, he noticed that the
          problem occurred when using DISA or WaitExten. He also noticed
          that when using Read, the problem did not occur. His system also
          used DUNDi for dialplan lookups. So, he theorized that if the
          DUNDi lookups blocked for some period of time, that audio from
          the zap channel could get lost. If the audio got lost, then it
          wouldn't be run through the DTMF detector, and digits could get
          lost. He was correct, and the following set of changes fixes the
          problem. However, the changes go a little bit further than what
          was necessary to fix this exact problem. 1) I updated
          pbx_extension_helper() to autoservice the associated channel to
          handle cases where extension lookups may take a long time. This
          would normally be a dialplan switch that does some lookup over
          the network, such as the DUNDi or IAX2 switches. This ensures
          that even while a DUNDi lookup is blocking, the channel will be
          continuously serviced. 2) I made a change to the autoservice
          code. This is actually something that has bothered me for a long
          time. When a channel is in autoservice, _all_ frames get thrown
          away. However, some frames really shouldn't be thrown away. The
          most notable examples are signalling (CONTROL) frames, and DTMF.
          So, this patch queues up important frames while a channel is in
          autoservice. When autoservice is stopped on the channel, the
          queued up frames get stuck back on the channel so that they can
          get processed instead of thrown away. 3) I made another change to
          the autoservice code to handle the case where autoservice is
          started on channels recursively. Previously, you could call
          ast_autoservice_start() multiple times on a channel, and it would
          stop the first time ast_autoservice_stop() gets called. Now, it
          will ensure that autoservice doesn't actually stop until the
          final call to ast_autoservice_stop().

2007-11-27 20:22 +0000 [r89727]  Mark Michelson <mmichelson@digium.com>

        * res/res_config_pgsql.c: Changing some calls from free() to
          ast_free() since they were allocated with ast_calloc(). (closes
          issue #11390, reported and patched by Laureano)

2007-11-27 20:16 +0000 [r89701-89709]  Kevin P. Fleming <kpfleming@digium.com>

        * main/app.c: on second thought... revert all the other changes
          i've made in app options parsing leaving only one: if an empty
          argument is supplied for an option, set that argument pointer to
          point to an empty string rather than NULL, so that the
          application can do normal checks on it without worrying about it
          being NULL

        * main/app.c: generate a warning when an application option that
          requires an argument is ignored due to lack of an argument

2007-11-27 16:12 +0000 [r89634]  Russell Bryant <russell@digium.com>

        * configs/voicemail.conf.sample: Add a note to the sample voicemail
          config noting that when using IMAP storage, only the first format
          specified will be attached to the message.

2007-11-27 15:38 +0000 [r89631]  Tilghman Lesher <tlesher@digium.com>

        * funcs/func_env.c: Default result of STAT should be "0" not "".
          Reported via the -users mailing list, fixed by me.

2007-11-27 15:23 +0000 [r89624-89630]  Olle Johansson <oej@edvina.net>

        * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: If we
          get a codec offer using a well-known payload type, but using it
          for another codec that we don't know, Asterisk did not remove
          that codec from the list. With this patch, we remove the codec
          from audio and video rtp objects and deny it ever existed. Thanks
          to lasse for testing. (closes issue #11376) Reported by: lasse
          Patches: bug11376.txt uploaded by oej (license 306) Tested by:
          lasse

        * configs/sip.conf.sample: Clarify limitonpeers=yes (closes issue
          #11304) Reported by: pj

2007-11-27 06:24 +0000 [r89622]  Steve Murphy <murf@digium.com>

        * apps/app_dial.c, main/cdr.c, configs/cdr.conf.sample,
          include/asterisk/cdr.h: closes issue #11379; OK, this is an
          attempt to make both sides happy. To the cdr.conf file, I added
          the option 'unanswered', which defaults to 'no'. In this mode,
          you will see a cdr for a call, whether it was answered or not.
          The disposition will be NO ANSWER or ANSWERED, as appropriate.
          The src is as you'd expect, the destination channel will be one
          of the channels from the Dial() call, usually the last in the
          list if more than one chan was specified. With unanswered set to
          'yes', you will still see this cdr entry in both cases. But in
          the case where the dial timed out, you will also see a cdr for
          each line attempted, marked NO ANSWER, with no destination
          channel name. The new option defaults to 'no', so you don't see
          the pesky extra cdr's by default, and you will not see the
          irritating 'not posted' messages.

2007-11-26 23:10 +0000 [r89616-89618]  Mark Michelson <mmichelson@digium.com>

        * apps/app_playback.c: After issuing a "say load new", if a caller
          hangs up during the middle of playback of a number, app_playback
          will continue to try to play the remaining files. With this
          change, no more files will be played back upon hangup. (closes
          issue #11345, reported and patched by IgorG)

        * apps/app_playback.c: After issuing a "say load new" tons of
          warning messages are printed out to the CLI every time do_say in
          app_playback is called. Removing these warnings

2007-11-26 21:10 +0000 [r89599-89610]  Joshua Colp <jcolp@digium.com>

        * main/dial.c: Fix issues with async dialing with an application
          executing. The application has to be terminated and control
          returned to the thread before hanging things up. (issue #BE-252)

        * res/res_features.c: Add module counting removal for error
          conditions. (closes issue #11333) Reported by: Laureano Patches:
          res_features_v2.c.patch uploaded by Laureano (license 265)

2007-11-26 17:41 +0000 [r89594]  Russell Bryant <russell@digium.com>

        * main/pbx.c: Add channel locking to a function that needed to be
          doing it. This is just a little something I noticed while working
          on a completely unrelated issue.

2007-11-26 17:36 +0000 [r89587-89592]  Joshua Colp <jcolp@digium.com>

        * pbx/pbx_config.c: Use ast_free to free memory, or else we shall
          implode if MALLOC_DEBUG is enabled. (closes issue #11347)
          Reported by: ys Patches: pbx.pbx_config.c.diff uploaded by ys
          (license 281)

        * apps/app_mixmonitor.c: Close the audio file before sending it to
          the post processing application. (closes issue #11357) Reported
          by: reformed Patches: mixmonitor.patch uploaded by reformed
          (license 330)

2007-11-26 17:20 +0000 [r89586]  Kevin P. Fleming <kpfleming@digium.com>

        * main/app.c: when parsing application options that take arguments,
          don't indicate that the option was supplied unless a
          non-zero-length argument was found for it

2007-11-26 15:48 +0000 [r89580]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Revert vmu->email back to an empty string
          if it was empty when imap_store_file was called. This prevents
          sending a duplicate e-mail. (closes issue #11204, reported by
          spditner, patched by me)

2007-11-26 15:34 +0000 [r89571-89577]  Joshua Colp <jcolp@digium.com>

        * main/channel.c: If channel allocation fails because the alert
          pipe could not be created also free the scheduler context.
          (closes issue #11355) Reported by: eliel Patches:
          main.channel.c.patch uploaded by eliel (license 64)

        * apps/app_meetme.c: When unloading app_meetme destroy any auto
          created contexts created by SLA. (closes issue #11367) Reported
          by: eliel

2007-11-25 17:17 +0000 [r89559]  Tilghman Lesher <tlesher@digium.com>

        * res/res_odbc.c, configs/res_odbc.conf.sample,
          include/asterisk/res_odbc.h, res/res_config_odbc.c: We previously
          attempted to use the ESCAPE clause to set the escape delimiter to
          a backslash. Unfortunately, this does not universally work on all
          databases, since on databases which natively use the backslash as
          a delimiter, the backslash itself needs to be delimited, but on
          other databases that have no delimiter, backslashing the
          backslash causes an error. So the only solution that I can come
          up with is to create an option in res_odbc that explicitly
          specifies whether or not backslash is a native delimiter. If it
          is, we use it natively; if not, we use the ESCAPE clause to make
          it one. Reported by: elguero Patch by: tilghman (Closes issue
          #11364)

2007-11-24 16:59 +0000 [r89534-89545]  Tilghman Lesher <tlesher@digium.com>

        * res/res_adsi.c: Free some frames that would otherwise leak on
          error. Reported by: Laureano Patch by: Laureano,tilghman (Closes
          issue #11351)

        * apps/app_voicemail.c, main/app.c: Currently, zero-length
          voicemail messages cause a hangup in VoicemailMain. This change
          fixes the problem, with a multi-faceted approach. First, we do
          our best to avoid these messages from being created in the first
          place, and second, if that fails, we detect when the voicemail
          message is zero-length and avoid exiting at that point. Reported
          by: dtyoo Patch by: gkloepfer,tilghman (Closes issue #11083)

        * main/manager.c: Up until this point, the XML output of the
          manager has been technically invalid, due to the repetition of
          certain parameters in a single event. This caused various issues
          for XML parsers, some of which refused to parse at all, given the
          invalidity of the rendered XML. So this commit fixes the XML
          output, ensuring that each entity parameter has a unique name,
          thus ensuring valid XML. Reported by: msetim Patch by: tilghman
          (Closes issue #10220)

        * res/res_config_odbc.c: Use ESCAPE clause for the first parameter,
          not just 2nd-Nth parameters. Reported by: apsaras Patch by:
          tilghman (Closes issue #11353)

2007-11-22 17:29 +0000 [r89527]  Russell Bryant <russell@digium.com>

        * configs/agents.conf.sample: mvanbaak pointed out a spelling error
          in this sample configuration file. While I was at it, I went
          ahead and tweaked it a little bit more.

2007-11-21 19:27 +0000 [r89493-89495]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Fix a small error I made in my previous commit

        * apps/app_queue.c: Changing an inaccurate debug message to be less
          inaccurate. Under the circumstances, this message would always
          report that there were 0 members available, even though that may
          not be true.

2007-11-21 18:59 +0000 [r89491]  Terry Wilson <twilson@digium.com>

        * res/res_features.c: If a channel gets masqueraded in the middle
          of a park, don't play the announcement to the masqueraded
          channel, and dial back to the original channel on timeout.

2007-11-20 19:16 +0000 [r89461-89462]  Kevin P. Fleming <kpfleming@digium.com>

        * include/asterisk/module.h: re-doxygen some comments

        * main/loader.c, include/asterisk/module.h,
          build_tools/make_buildopts_h: bring back compile-option checking
          when loading modules, only this time use a string-based storage
          and comparison mechanism because it is easier to support on other
          platforms

2007-11-20 17:50 +0000 [r89457]  Mark Michelson <mmichelson@digium.com>

        * main/pbx.c: According to comments in main/pbx.c, it is essential
          that if we are going to lock the conlock as well as the hints
          lock, it must be locked in that respective order. In order to
          prevent a potential deadlock, we need to lock the conlock prior
          to locking the hints lock in ast_hint_state_changed (see the call
          stack example on issue #11323 for how this can happen). (closes
          issue #11323, reported by eelcob, suggestion for patch by eelcob,
          patch by me)

2007-11-20 15:22 +0000 [r89450]  Steve Murphy <murf@digium.com>

        * doc/queues-with-callback-members.txt: closes issue #11324; break
          statements missing in switch cases.

2007-11-20 13:40 +0000 [r89445]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c: added RR patch from iroot #10908, thanks.

2007-11-19 15:53 +0000 [r89416-89419]  Joshua Colp <jcolp@digium.com>

        * res/res_features.c: Print out the correct filename
          (features.conf) in the log message when parkpos options are
          incorrect. (closes issue #11295) Reported by: Laureano Patches:
          res_features.c.patch uploaded by Laureano (license 265)

        * doc/localchannel.txt: Clarify documentation a bit, include that a
          frame has to pass through the core in order for the Local channel
          optimization to happen. (closes issue #11246) Reported by: jon

2007-11-16  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.14 released.

2007-11-16 22:26 +0000 [r89339]  Russell Bryant <russell@digium.com>

        * main/loader.c, include/asterisk/module.h,
          build_tools/make_buildopts_h: Temporarily revert revision 89325,
          which added md5 magic for keeping track of what build options
          were used. We agreed that we should remove this before making a
          1.4 release, and then we can put it back in. Then, we can take a
          month or so to play around with it to get it how we want it.

2007-11-16 16:47 +0000 [r89325]  Kevin P. Fleming <kpfleming@digium.com>

        * main/loader.c, include/asterisk/module.h,
          build_tools/make_buildopts_h: To help combat problems where
          people build external modules (asterisk-addons or others) and
          then change the build options of the Asterisk build in a way that
          makes the incompatible without warning, this commit introduces an
          MD5 signature of the important build-time options and includes
          that signature into modules when they are built. When the loader
          loads one of these modules and notices the problem, it will emit
          a warning to console and refuse to initialize the module, as
          doing so could cause the system to be unstable or even crash. If
          you upgrade to this version of Asterisk, you must rebuild *all*
          of your modules that came from other sources before trying to run
          this version. If you are using Digium's G.729 binary codec
          module, you will need v33 or newer.

2007-11-16 15:28 +0000 [r89323]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Make realtime queues accessible from the
          QUEUE_MEMBER_COUNT function. (closes issue #11271, reported and
          patched by atis, with small modifications from me)

2007-11-15 18:37 +0000 [r89298-89302]  Tilghman Lesher <tlesher@digium.com>

        * Makefile: Start Asterisk in Debian at a more reasonable time
          (since zaptel is at level 20)

        * channels/misdn/isdn_lib.c: Fix an uninitialized memory read found
          by valgrind

        * channels/chan_iax2.c: Yet another memory corruption issue.
          Reported by: atis Patch by: tilghman Fixes issue #10923

2007-11-15 17:19 +0000 [r89296]  Russell Bryant <russell@digium.com>

        * apps/app_meetme.c: Update the SLAStation application to account
          for the case where the SLA thread has a call out to the station,
          but the user has pressed a line button to answer the call instead
          of picking up the handset. If they do, the phone sends out a new
          INVITE. So, the SLAStation app must check to see if it is picking
          up a ringing trunk, and ensure that the other stations stop
          ringing. (reported internally, patched by me, tested by mogorman)

2007-11-15 14:57 +0000 [r89286-89288]  Mark Michelson <mmichelson@digium.com>

        * main/manager.c: Undoing previous commit since I realize it was
          wrong

        * main/manager.c: Adding a missing mutex unlock. (closes issue
          11256, reported and patched by ys)

2007-11-15 11:26 +0000 [r89280-89281]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Don't send re-invites during pending INVITE
          transactions. Patch by one47 - thanks! Closes issue #9305

        * channels/chan_sip.c: Improve support for multipart messages. Code
          by gasparz, changes by me (mostly formatting). Thanks, gasparz!
          Closes issue #10947

2007-11-14 23:23 +0000 [r89275]  Tilghman Lesher <tlesher@digium.com>

        * main/app.c: When a recording ends with '#', we are improperly
          trimming an extra 200ms from the recording. Reported by: sim
          Patch by: tilghman Closes issue #11247

2007-11-14 01:15 +0000 [r89260]  Joshua Colp <jcolp@digium.com>

        * main/srv.c: Return the proper value when the srv_callback
          function executes properly. (closes issue #11240) Reported by:
          jtodd

2007-11-13 21:07 +0000 [r89248-89254]  Jason Parker <jparker@digium.com>

        * channels/chan_zap.c, channels/chan_iax2.c: Fix building on newer
          systems which require a third arg to open() when using O_CREAT.
          Issue 11238, reported by puzzled.

        * res/res_features.c: Revert change from revision 67064. It is
          documented behavior that if a parking extension already exists
          while using PARKINGEXTEN, dialplan execution will continue. If
          blind transferring to a Park with PARKINGEXTEN, you must keep
          this in mind, and handle the failure yourself. Issue 11237,
          reported by jon.

2007-11-13 17:34 +0000 [r89246]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_sip.c: If we set a value for qualify, we should
          actually pay attention to it, instead of overriding the value

2007-11-13 16:02 +0000 [r89241]  Mark Michelson <mmichelson@digium.com>

        * apps/app_mixmonitor.c: Reverting commit made in revision 89205
          since it is unnecessary. Thanks to Kevin for pointing this out

2007-11-13 13:51 +0000 [r89239]  Tilghman Lesher <tlesher@digium.com>

        * main/utils.c: Debugging is running into the 16-lock limit.
          Increase to avoid. (This define is only effective when debugging
          is turned on, so there's no effect for most installations.)

2007-11-13 00:56 +0000 [r89205]  Mark Michelson <mmichelson@digium.com>

        * apps/app_mixmonitor.c: Some sanity checking for MixMonitor. If
          only 1 argument is given, then the args.options and
          args.post_process strings are uninitialized and could contain
          garbage. This change handles this situation properly by only
          using arguments that we have parsed.

2007-11-12 20:46 +0000 [r89194]  Jason Parker <jparker@digium.com>

        * main/pbx.c: Fix a typo pointed out by De_Mon on #asterisk-dev

2007-11-12 20:16 +0000 [r89184-89191]  Tilghman Lesher <tlesher@digium.com>

        * main/config.c: If two config writes collide, file corruption
          could result. Use a mkstemp() file, instead. Reported by:
          paravoid Patch by: tilghman Closes issue #10781

        * main/channel.c, channels/chan_sip.c: Fix two cases of memory
          corruption caused by background threads. Reported by: atis Patch
          by: tilghman Fixes issue #10923

2007-11-12 11:26 +0000 [r89169-89173]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c, configs/misdn.conf.sample: if we're NT and
          no number was dialed and overlapdial is set, we wait for the ISDN
          timeout instead of starting our own timer. added a comment for
          the misdn.conf.sample for the overlapdial config option.

        * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h,
          channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: added
          restart all interfaces Restart_Indicator, to automatically send a
          RESTART after the L2 of a PTP Port comes up. Also fixed some
          places where we have send a RELEASE without need for it.

        * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a
          state/event issue with overlapdial=yes when no extension matched.
          removed the general sending of a RELEASE_COMPLETE when we receive
          a RELEASE, this is done by mISDNuser/mISDN. This makes it
          possible to use asterisk-1.4 with mISDN trunk, but requires users
          of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6
          (when using the NT mode at all)

        * channels/misdn/isdn_lib.c: fixed the support for CW and therefore
          for the reject_cause option.

        * channels/misdn/isdn_lib.c, channels/misdn_config.c,
          channels/misdn/isdn_lib.h, channels/chan_misdn.c,
          channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
          aded ntkeepcalls option, to avoid droÃpping calls when the L2
          goes down on a PTP link. There are some pbx which do turn off the
          L1 for a very short while and restart it immediately. normally
          T310 should be started and after 10 seconds or so the calls
          should be dropped, this is a simple fix wihtout this timer.

2007-11-08 23:52 +0000 [r89125]  Jason Parker <jparker@digium.com>

        * main/say.c: Properly say the seconds here.. Issue 11203, fix
          described by vma.

2007-11-08 21:00 +0000 [r89119]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: Rework of the commit I made yesterday to use
          the already built-in ast_uri_decode function as opposed to my
          home-rolled one. Also added comments. Thanks to oej for pointing
          me in the right direction

2007-11-08 18:45 +0000 [r89115]  Jason Parker <jparker@digium.com>

        * configs/res_odbc.conf.sample: Avoid warnings on load when using
          sample configuration files. Issue 11195, patch by eliel.

2007-11-08 16:47 +0000 [r89111]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: I made this same adjustment in trunk to fix
          a bug, and it makes sense to do it in 1.4 as well. If an
          imapfolder is specified in voicemail.conf, don't ever explicitly
          connect to INBOX since it may not exist.

2007-11-08 05:26 +0000 [r89105]  Kevin P. Fleming <kpfleming@digium.com>

        * main/srv.c: fix a glaring bug in the new SRV record handling that
          would cause incorrect weight sorting

2007-11-08 04:55 +0000 [r89103]  Tilghman Lesher <tlesher@digium.com>

        * doc/valgrind.txt: Typo

2007-11-08 02:26 +0000 [r89095-89101]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Do not add a sip: to the beginning of the To
          URI unless needed. (closes issue #10756) Reported by: goestelecom

        * channels/chan_sip.c: Improve the devicestate logic for multiple
          devices. If any are available then the extension is considered
          available. (closes issue #10164) Reported by: nic_bellamy
          Patches: sip-hinting-svn-branch-1.4.patch uploaded by nic
          (license 299)

        * channels/chan_sip.c: Add support for allowing one outgoing
          transaction. This means if a response comes back out of order
          chan_sip will still handle it. I dream of a chan_sip with real
          transaction support. (closes issue #10946) Reported by: flefoll
          (closes issue #10915) Reported by: ramonpeek (closes issue #9567)
          Reported by: atca_pres

        * channels/chan_sip.c: If callerid is configured in sip.conf use
          that for checking the presence of an extension in the dialplan.
          (closes issue #11185) Reported by: spditner

2007-11-07 23:39 +0000 [r89093]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_queue.c: The member refcount must be incremented, to
          avoid using it after deallocation. A huge thanks go to lvl- for
          patiently providing the necessary valgrind output that was
          necessary to finding this problem of memory corruption. Reported
          by: lvl- Patch by: tilghman Closes issue #11174

2007-11-07 22:40 +0000 [r89090]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: This patch makes it possible for SIP phones
          to dial extensions defined with '#' characters in extensions.conf
          AND maintain their escaped characters when forming URI's (closes
          issue #10681, reported by cahen, patched by me, code review by
          file)

2007-11-07 21:40 +0000 [r89088]  Steve Murphy <murf@digium.com>

        * cdr/cdr_tds.c, pbx/pbx_ael.c, res/res_jabber.c: In response to
          10578, I just ran 1.4 thru valgrind; some of the config leakage
          I've already fixed, but it doesn't hurt to double check. I found
          and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major,
          tho.

2007-11-07 15:56 +0000 [r89085]  Mark Michelson <mmichelson@digium.com>

        * main/manager.c: Fixing a segfault in the manager "core show
          channels concise" command. (closes issue #11183, reported by arnd
          and patched by ys)

2007-11-07 04:07 +0000 [r89079]  Tilghman Lesher <tlesher@digium.com>

        * configs/extensions.ael.sample: Suppress AEL warnings on load.
          Reported by: eliel Patch by: eliel Closes issue #11178

2007-11-06 20:18 +0000 [r89053]  Russell Bryant <russell@digium.com>

        * res/res_musiconhold.c: Fix init_classes() so that classes that
          actually do have files loaded aren't treated as empty, and
          immediately destroyed ...

2007-11-06 19:09 +0000 [r89046]  Jason Parker <jparker@digium.com>

        * codecs/codec_zap.c: Correctly set the total number of channels
          from a zaptel transcoder board. SPD-49, patch by Matthew
          Nicholson.

2007-11-06 19:09 +0000 [r89045]  Tilghman Lesher <tlesher@digium.com>

        * include/asterisk/lock.h: We went to the trouble of creating a
          method of tracking failed trylocks, then never turned it on
          (oops).

2007-11-06 18:53 +0000 [r89042]  Olle Johansson <oej@edvina.net>

        * main/tdd.c: Bug fixes to tdd support in zaptel.

2007-11-06 18:20 +0000 [r89037]  Russell Bryant <russell@digium.com>

        * res/res_musiconhold.c: If someone were to delete the files used
          by an existing MOH class, and then issue a reload, further use of
          that class could result in a crash due to dividing by zero. This
          set of changes fixes up some places to prevent this from
          happening. (closes issue #10948) Reported by: jcomellas Patches:
          res_musiconhold_division_by_zero.patch uploaded by jcomellas
          (license 282) Additional changes added by me.

2007-11-06 17:52 +0000 [r89036]  Steve Murphy <murf@digium.com>

        * main/config.c: closes issue #8786 - where the [catname](!) and
          [catname](othercat1,othercat2,...) notation gets dropped across a
          ConfigUpdate (or any other thing that would cause a config file
          to be written). While I was at it, I also cleaned up some of the
          destroy routines to free up comments, which was not being done.
          Made sure the new struct I introduced is also cleaned up properly
          at destruction time. My code handles multiple template
          inclusions. Many thanks to ssokol for his patch, which, while not
          literally used in the final merge, served as a foundation for the
          fix.

2007-11-06 17:08 +0000 [r88994-89032]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Make it so that if a peer is determined to
          be unreachable using qualify their devicestate will report back
          unavailable. (closes issue #11006) Reported by: pj

        * channels/chan_zap.c: Fix improbable but possible memory leaks in
          chan_zap. (closes issue #11166) Reported by: eliel Patches:
          chan_zap.c.patch uploaded by eliel (license 64)

2007-11-06 13:50 +0000 [r88931]  Russell Bryant <russell@digium.com>

        * include/asterisk/lock.h: Remove some checks to see if locks are
          initialized from the non-DEBUG_THREADS versions of the lock
          routines. These are incorrect for a number of reasons: - It
          breaks the build on mac. - If there is a problem with locks not
          getting initialized, then the proper fix is to find that place
          and fix the code so that it does get initialized. - If additional
          debug code is needed to help find the problem areas, then this
          type of things should _only_ be put in the DEBUG_THREADS
          wrappers.

2007-11-06 02:52 +0000 [r88862]  Kevin P. Fleming <kpfleming@digium.com>

        * include/asterisk/srv.h: update comment to match the state of the
          code

2007-11-05 23:29 +0000 [r88826]  Mark Michelson <mmichelson@digium.com>

        * main/channel.c: Reworked deadlock avoidance in __ast_read.
          Restored audio to callback agents. (closes issue #11071, reported
          by callguy, patched by me, tested by callguy and Ted Brown)

2007-11-05 22:07 +0000 [r88709-88805]  Russell Bryant <russell@digium.com>

        * main/pbx.c, include/asterisk/pbx.h: After seeing crashes related
          to channel variables, I went looking around at the ways that
          channel variables are handled. In general, they were not handled
          in a thread-safe way. The channel _must_ be locked when reading
          or writing from/to the channel variable list. What I have done to
          improve this situation is to make pbx_builtin_setvar_helper() and
          friends lock the channel when doing their thing. Asterisk API
          calls almost all lock the channel for you as necessary, but this
          family of functions did not. (closes issue #10923, reported by
          atis) (closes issue #11159, reported by 850t)

        * channels/chan_sip.c: When traversing the list of channel
          variables here in transmit_invite(), the asterisk channel must be
          locked, as this data may change at any time. (I have seen
          numerous reports of crashes related to the handling of channel
          variables. There are a couple of issues on the bug tracker
          related to it, but it has also been noted on IRC and mailing
          lists. So, I am finding and fixing some places where channel
          variables are handled improperly.)

        * channels/chan_sip.c: Fix up some indentation.

        * main/srv.c, include/asterisk/srv.h: Merge changes from
          asterisk/team/kpfleming/SRV-priority-handling Previously, the SRV
          record support in Asterisk was broken. There was no guarantee on
          what record Asterisk would choose to actually use. This set of
          changes improves the situation by ensuring that Asterisk will
          choose the highest priority record.

        * main/channel.c: Merge the last bit of changes from
          asterisk/team/russell/readq-1.4 The issue here is that the
          channel frame readq handling got broken when the code was
          converted to use the linked list macros. It caused corruption of
          the list head and tail pointers. So, I fixed up the usage of the
          linked list macros and in passing, simplified the code. I also
          documented what the code is doing, as it was a bit difficult to
          figure out at first. This bug showed itself with crashes showing
          messed up head/tail pointers for the readq. However, there are a
          couple of crashes that aren't quite as obvious, but I think may
          be related. So, if your bug gets closed by this commit, but you
          still have a problem, please reopen or create a new bug report.
          (closes issue #10936) (closes issue #10595) (closes issue #10368)
          (closes issue #11084) (closes issue #10040) (closes issue #10840)

2007-11-05 18:47 +0000 [r88671]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: If a SIP channel is put on hold multiple
          times do not keep incrementing the onHold value. (closes issue
          #11085) Reported by: francesco_r Tested by: blitzrage (closes
          issue #10474) Reported by: acennami

2007-11-05 17:46 +0000 [r88624]  Russell Bryant <russell@digium.com>

        * main/channel.c: Fix up datastore handling in ast_do_masquerade().
          The code is intended to move any channel datastores from the old
          channel to the new one. However, it did not use the linked list
          macros properly to accomplish the task. The existing code would
          only work if there was only a single datastore on the old
          channel.

2007-11-05 17:19 +0000 [r88585]  Jason Parker <jparker@digium.com>

        * channels/chan_sip.c: Make sure we destroy the config structure on
          configuration failure. Issue 11163, patch by eliel.

2007-11-05 16:20 +0000 [r88539]  Tilghman Lesher <tlesher@digium.com>

        * res/res_odbc.c: Don't check used pooled connections for
          connection status, as it will cause issues for prepared queries.
          Reported by: Nick Gorham (via -dev list) Patch by: tilghman

2007-11-04 22:38 +0000 [r88471]  Luigi Rizzo <rizzo@icir.org>

        * include/asterisk/stringfields.h, main/channel.c,
          apps/app_meetme.c, channels/chan_sip.c, channels/chan_iax2.c:
          Rename ast_string_field_free_pool to
          ast_string_field_free_memory, and ast_string_field_free_all to
          ast_string_field_reset_all to avoid misuse (due to too similar
          names and an error in documentation). Fix two related memory
          leaks in app_meetme. No need to merge to trunk, different fix
          already applied there. Not applicable to 1.2

2007-11-02 20:49 +0000 [r88328-88366]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Make subscribecontext behave as advertised.
          It will now look for the presence of a hint in the given context
          (be it subscribecontext or context). (closes issue #10702)
          Reported by: slavon

        * channels/chan_sip.c: If an INFO request within a dialog is
          received with a content length of 0 simply send back a 200 OK. It
          is valid to do this and the remote side is probably using it to
          make sure the signalling is still alive. (closes issue #5747)
          Reported by: chandi Patches: infofix-81430-1.patch uploaded by
          IgorG (license 20)

2007-11-02 16:51 +0000 [r88283]  Jason Parker <jparker@digium.com>

        * main/say.c: We need to make sure to specify a language to
          ast_fileexists, otherwise it may fail for anything besides en
          Issue 11147, fix discovered by both citats and myself
          (independently), with input from Corydon76

2007-11-02 13:03 +0000 [r88116-88210]  Tilghman Lesher <tlesher@digium.com>

        * include/asterisk/lock.h: Fix build on Solaris Reported by: snuffy
          Patch by: ys Closes issue #11143

        * doc/valgrind.txt (added): Add some notes on using valgrind

2007-11-01 16:21 +0000 [r88078]  Jason Parker <jparker@digium.com>

        * channels/chan_zap.c: Make sure we set the poll fds to NULL after
          free()ing it. Part of issue 11017, patch by tzafrir.

2007-11-01 13:27 +0000 [r87970-88026]  Joshua Colp <jcolp@digium.com>

        * apps/app_meetme.c: Fix up commit for my Zap channel with spies in
          Meetme fix. (thanks Tony Mountifield!)

        * apps/app_meetme.c: If a Zap channel contains a spy or a spy is
          added take it out of the conference in kernel space and make it
          go through Asterisk so the spy gets audio from both sides.
          (closes issue #10060) Reported by: mparker

2007-10-31 21:23 +0000 [r87906-87908]  Jason Parker <jparker@digium.com>

        * res/res_jabber.c: Make sure we free some allocated memory before
          returning. Issue 11131, patch by eliel.

        * channels/chan_gtalk.c: Don't try to allocate memory that we're
          just going to re-allocate later anyways. Issue 11130, patch by
          eliel.

2007-10-31 18:03 +0000 [r87852]  Tilghman Lesher <tlesher@digium.com>

        * Makefile: Create samples for ALL of the available options in
          asterisk.conf

2007-10-31 17:49 +0000 [r87775-87849]  Steve Murphy <murf@digium.com>

        * pbx/pbx_config.c: closes issue #11108 -- where the 'dialplan
          save' cli command saves a file where the semicolon is not
          escaped. Fixed this; User also wanted comments to be preserved
          across dialplan save, but this is impossible at this point in
          time, because comments are not stored in the dialplan. They are
          'compiled' out of extensions.conf. The only way to preserve those
          comments is to use the config file reader/writer that the GUI
          uses to allow online user edits. extensions.conf is first and
          foremost, a config file, and is read in by the normal config-file
          reading routines. Then, it is processed into a dialplan
          (context/exten structs).

        * pbx/pbx_ael.c: Included some verbage in the check_includes func,
          to inform the user that included contexts that have no match in
          the AEL, might be OK, as AEL cannot check in the extensions.conf
          or the in-memory contexts, as they may not be there at the time
          of the check.

2007-10-30 23:02 +0000 [r87739]  Tilghman Lesher <tlesher@digium.com>

        * include/asterisk/lock.h: Fix for uninitialized mutexes on *BSD
          Reported by: ys Fixed by: ys Closes issue #11116

2007-10-30 21:19 +0000 [r87686]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Merge the changes from
          team/russell/iax2_poke_fix and iax2-poke-fix-trunk There was a
          race condition related to the handling of POKEing peers.
          Essentially, a reference to a peer is held by the scheduler when
          there are pending callbacks, but the reference count didn't
          reflect it. So, it was possible for a peer to hit a reference
          count of zero and have its destructor begin to be called at the
          same time that the scheduler thread ran a POKE related callback.
          If that happened, a crash would likely occur. (closes issue
          #11082, closes issue #11094)

2007-10-30 20:29 +0000 [r87650]  Jason Parker <jparker@digium.com>

        * channels/Makefile: Only try to clean out h323/ if the
          h323/Makefile exists.

2007-10-30 16:13 +0000 [r87571]  Joshua Colp <jcolp@digium.com>

        * res/res_features.c: Add two more checks before printing out a
          warning message about bridging. If either channel has hungup of
          course the bridge will have failed. (closes issue #10009)
          Reported by: dimas

2007-10-30 15:45 +0000 [r87567]  Jason Parker <jparker@digium.com>

        * main/editline/np/vis.c: Fix build of editline on Solaris. Issue
          11113, patch by snuffy.

2007-10-30 15:10 +0000 [r87534]  Joshua Colp <jcolp@digium.com>

        * apps/app_followme.c: Return 1.4 to a state where it builds.
          Changing the arguments to a function and not changing where they
          are used is bad, mmmk?

2007-10-30 14:31 +0000 [r87514]  BJ Weschke <bweschke@btwtech.com>

        * apps/app_followme.c: Fix issue where the recorded name wasn't
          getting removed correctly. (closes issue #11115) Reported by:
          davevg Patches: followme-v3.diff

2007-10-29 22:13 +0000 [r87460-87465]  Kevin P. Fleming <kpfleming@digium.com>

        * codecs/gsm: missed one directory

        * codecs/ilbc, formats, utils/Makefile, agi/Makefile, funcs,
          codecs/lpc10, main/db1-ast, main/editline, main,
          codecs/ilbc/Makefile, pbx, res, channels, main/db1-ast/Makefile,
          codecs/lpc10/Makefile, utils, codecs, agi,
          main/editline/Makefile.in, apps, Makefile.moddir_rules, cdr:
          clean up (and ignore) assembler and preprocessor intermediate
          files if any are created during the build

        * Makefile: don't put '-pipe' into ASTCFLAGS if '-save-temps' is
          already there (used when debugging preprocessor issues) because
          the compiler will whine about each compile command

2007-10-29 21:06 +0000 [r87427]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Removing a completely unnecessary quota
          check from IMAP code.

2007-10-29 20:22 +0000 [r87373-87396]  Russell Bryant <russell@digium.com>

        * main/utils.c, include/asterisk/lock.h: Add some more details to
          the output of "core show locks". When a thread is waiting for a
          lock, this will now show the details about who currently has it
          locked. (inspired by issue #11100)

        * main/astmm.c: Remove a lock that doesn't make any sense. The
          regions lock needs to be held when traversing the list of
          allocated chunks so that they can be printed out to the CLI.
          (Thanks to eliel on #asterisk-dev for pointing this out!)

2007-10-29 17:20 +0000 [r87342]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Fix issue where if both sides of the dialog
          cancelled the dialog at the same time chan_sip could kepe
          retransmitting a response for no reason. (closes issue #9566)
          Reported by: atca_pres Patches: bug9566.patch uploaded by oej

2007-10-29 17:13 +0000 [r87340]  Jason Parker <jparker@digium.com>

        * funcs/func_realtime.c, funcs/func_cut.c: Allow some function
          modules to compile under dev mode. Issue 11104, patch by andrew.

2007-10-29 14:23 +0000 [r87294]  Joshua Colp <jcolp@digium.com>

        * main/utils.c: Fix issue with ast_unescape_semicolon going into an
          endless loop. (closes issue #10550) Reported by: ramonpeek
          Patches: unescape-85177-1.patch uploaded by IgorG (license 20)

2007-10-28 13:46 +0000 [r87262]  Tilghman Lesher <tlesher@digium.com>

        * funcs/func_realtime.c, funcs/func_odbc.c, funcs/func_strings.c,
          funcs/func_cut.c: Add autoservice to several more functions which
          might delay in their responses. Also, make sure that func_odbc
          functions have a channel on which to set variables. Reported by
          russell Fixed by tilghman Closes issue #11099

2007-10-26 16:34 +0000 [r87168]  Steve Murphy <murf@digium.com>

        * pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael.tab.c,
          pbx/ael/ael.y, pbx/ael/ael_lex.c, pbx/pbx_ael.c,
          include/asterisk/ael_structs.h, pbx/ael/ael.tab.h,
          utils/ael_main.c, pbx/ael/ael-test/ref.ael-test16,
          pbx/ael/ael.flex: closes issue #11086 where a user complains that
          references to following contexts report a problem; The problem
          was REALLy that he was referring to empty contexts, which were
          being ignored. Reporter stated that empty contexts should be OK.
          I checked it out against extensions.conf, and sure enough, empty
          contexts ARE ok. So, I removed the restriction from AEL. This,
          though, highlighted a problem with multiple contexts of the same
          name. This should be OK, also. So, I added the extend keyword to
          AEL, and it can preceed the 'context' keyword (mixed with
          'abstract', if nec.). This will turn off the warnings in AEL if
          the same context name is used 2 or more times. Also, I now call
          ast_context_find_or_create for contexts now, instead of just
          ast_context_create; I did this because pbx_config does this. The
          'extend' keyword thus becomes a statement of intent. AEL can now
          duplicate the behavior of pbx_config,

2007-10-26 13:54 +0000 [r87120]  Tilghman Lesher <tlesher@digium.com>

        * funcs/func_curl.c: The addition of autoservice to func_curl
          additionally made func_curl dependent on the existence of a
          channel, with no real reason. This should make func_curl once
          again work without a channel. Reported by jmls. Fixed by
          tilghman. Closes issue #11090

2007-10-25 23:03 +0000 [r87069]  Kevin P. Fleming <kpfleming@digium.com>

        * main/channel.c, include/asterisk/linkedlists.h: appending one
          list to another should leave the first list empty, and not
          require the user to do that

2007-10-25 22:53 +0000 [r87067]  Tilghman Lesher <tlesher@digium.com>

        * funcs/func_cut.c: Backport alternate encoding of newline
          delimiters from trunk to 1.4, as approved by Russell Reported by
          blitzrage Closes issue #10903

2007-10-24 20:56 +0000 [r86982]  Jason Parker <jparker@digium.com>

        * channels/chan_zap.c: Correctly respect hidecalleridname
          configuration option. Simplify code slightly in the process.
          Issue 11079, reported by ddv2005

2007-10-24 04:14 +0000 [r86880-86936]  Steve Murphy <murf@digium.com>

        * pbx/ael/ael.tab.c, pbx/ael/ael.y: closes issue #11037 -- unable
          to specify app:spec in hint arguments

        * funcs/func_logic.c: closes issue #11052 -- where nothing after
          the ? will allow un-initialized variable values to corrupt and
          crash asterisk on 64-bit platforms

        * main/Makefile: this update to Makefile corrects how ast_expr2f.c
          should be generated

        * main/ast_expr2f.c: This should get rid of a really, really
          irritating warning generated by some 64-bit platforms from libc,
          where free(0) is frowned upon

2007-10-22 21:36 +0000 [r86836]  Russell Bryant <russell@digium.com>

        * include/asterisk/lock.h: If lock tracking is not enabled, then we
          can not attempt to log any mutex failures. If so, we could end up
          in infinite recursion. The only lock that is affected by this is
          a mutex in astmm.c used when MALLOC_DEBUG is enabled. (closes
          issue #11044) Reported by: ys Patches: lock.h.diff uploaded by ys
          (license 281)

2007-10-22 17:38 +0000 [r86787]  Tilghman Lesher <tlesher@digium.com>

        * main/astmm.c: Minor FreeBSD build fix

2007-10-22 16:35 +0000 [r86754-86756]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: After reading online I have confirmed that
          Record-Route headers should be copied to 1xx responses as well.
          (closes issue #10113) Reported by: makoto

        * apps/app_controlplayback.c: Make sure res is a positive value
          before performing the check to determine whether the user stopped
          it or not. (closes issue #11023) Reported by: cfc

2007-10-22 15:52 +0000 [r86726-86750]  Russell Bryant <russell@digium.com>

        * main/channel.c: Don't leak a frame in the case that an END frame
          is received and the time since the BEGIN is less than that of the
          defined minimum DTMF duration. (closes issue #11051) Reported by:
          casper Patches: channel.c.86664.diff uploaded by casper (license
          55)

        * include/asterisk/lock.h: Update the static mutex initializer to
          include the initialization of the internal mutex used to protect
          the lock debugging data. (closes issue #11044, patch suggested by
          Ivan)

2007-10-22 14:48 +0000 [r86694]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Account for the fact that sometimes headers
          may be terminated with \r\n instead of just \n (closes issue
          #11043, reported by yehavi)

2007-10-22 14:27 +0000 [r86630-86663]  Joshua Colp <jcolp@digium.com>

        * main/channel.c: Move log message to before the frame it
          references is freed. (closes issue #11050) Reported by: slavon
          Patches: channel.c.86662.diff uploaded by casper (license 55)

        * pbx/pbx_dundi.c: Fix tab completion for dundi show peer. (closes
          issue #11041) Reported by: jsmith Patches:
          asterisk-dundicomplete.diff.txt uploaded by jamesgolovich
          (license 176)

        * main/loader.c: Fixes for building under OpenSolaris. (closes
          issue #11047) Reported by: snuffy Patches: 11047-fixes.diff
          uploaded by snuffy (license 35)

2007-10-22 09:21 +0000 [r86598]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/chan_misdn.c: we send
          DISCONNECT instead of RELEASE/RELEASE_COMPLETE if the dialplan
          does not match after an overlap call. Also added out_cause=1

2007-10-19 16:38 +0000 [r86469-86502]  Joshua Colp <jcolp@digium.com>

        * main/app.c: When returning a DTMF digit from
          ast_control_streamfile cast it as a char so that 0 does not
          overlap with the success return code. (closes issue #11023)
          Reported by: cfc

        * channels/chan_sip.c: Fix two issues with domains and transfers.
          If a port was given in the hostname it was treated as part of the
          hostname. If domains were configured but external domains were
          not enabled all transfers would be considered remote. (closes
          issue #11027) Reported by: ramonpeek Patches: 11027-1.diff
          uploaded by ramonpeek (license 266)

        * channels/chan_sip.c: Set port number in received as information
          for registrations as well. (closes issue #11028) Reported by:
          brad-x

2007-10-19 01:45 +0000 [r86438]  TransNexus OSP Development <support@transnexus.com>

        * apps/app_osplookup.c: Fixed OSP module did not report
          source/devinfo IP in correct format.

2007-10-18 22:01 +0000 [r86405-86406]  Jason Parker <jparker@digium.com>

        * Makefile: Correct documentation. I removed the wrong line..

        * Makefile: Add documentation for options in asterisk.conf Issue
          11029, patch by eserra

2007-10-18 21:16 +0000 [r86330-86372]  Russell Bryant <russell@digium.com>

        * configs/iax.conf.sample, channels/chan_iax2.c: Revert erroneous
          commit.

        * configs/iax.conf.sample, channels/chan_iax2.c: Add support for
          setting the maximum trunk size for IAX2 trunking

        * main/channel.c, include/asterisk/channel.h: The channel needs to
          stay locked while running timer callbacks, as they access and
          modify channel data that may change elsewhere. I went through
          every timer callback in the source tree to make sure that none of
          them did any additional locking that could introduce deadlocks,
          and all is well. (closes issue #10765) Reported by: Ivan Patches:
          ast_1_4_11_svn_patch_channel_rc.diff uploaded by Ivan (license
          229)

2007-10-18 17:38 +0000 [r86328]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: If a non-existent file is specified to be
          played either as a periodic announcement or as a hold/position
          announcement, the caller would be kicked out of the queue. No
          longer does this happen.

2007-10-18 15:45 +0000 [r86237-86296]  Russell Bryant <russell@digium.com>

        * codecs/codec_zap.c: Execute the RELEASE operation on transcoder
          channels in the destroy callback. (patch from jsloan)

        * main/utils.c: Revert a change that I made for issue #10979 which,
          as has been pointed out to me in issue #11018, doesn't really
          make sense. There is no reason to have the base64 decode function
          force a '\0' terminated buffer, when the result is almost always
          binary, anyway. In fact, this caused some breakage, as some code
          in res_crypto passed in a buffer exactly the right size to get
          its binary result, which got stomped on by this patch. (closes
          issue #11018, reported by dimas)

2007-10-17 21:39 +0000 [r86202]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Changing the strategy field of the call_queue
          struct to be signed instead of unsigned, since the code attempts
          to set the strategy to -1 if you specify a bogus strategy. While
          this isn't a huge issue in 1.4, it could be a problem for someone
          who, say, tries to use the roundrobin strategy in trunk (despite
          all the deprecation warnings in 1.4).

2007-10-17 17:57 +0000 [r86149]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: If Asterisk is in the middle of shutting
          down, respond to OPTIONS with 503 Unavailable. (closes issue
          #10994) Reported by: eserra Patches: sip-options-503.patch
          uploaded by eserra (license 45)

2007-10-17 16:58 +0000 [r86117]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Whoops, forgot to remove the original
          sip_scheddestroy. (closes issue #11010) Reported by: vadim

2007-10-17 15:23 +0000 [r86066]  Tilghman Lesher <tlesher@digium.com>

        * main/asterisk.c: When runuser/rungroup is specified, a remote
          console could only be attained by root (Closes issue #9999)

2007-10-17 15:06 +0000 [r86063]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Don't schedule dialog destruction if a
          MESSAGE is received using an existing dialog. (closes issue
          #11010) Reported by: vadim

2007-10-16 23:35 +0000 [r86028-86032]  Mark Michelson <mmichelson@digium.com>

        * configs/queues.conf.sample: Since monitor-join is deprecated now,
          remove the example from the sample queues.conf file

        * UPGRADE.txt: Updating UPGRADE.txt to reflect the deprecation of
          the monitor-join queue option

        * apps/app_queue.c: Adding deprecated warning to monitor-join
          option, since the plan is to no longer support this in favor of
          monitor-type = mixmonitor (related to issue #10885)

2007-10-16 22:36 +0000 [r85994-85997]  Russell Bryant <russell@digium.com>

        * include/asterisk/lock.h: really picky formatting tweak ...

        * include/asterisk/lock.h: Some locking errors exposed the fact
          that the lock debugging code itself was not thread safe. How
          ironic! Anyway, these changes ensure that the code that is
          accessing the lock debugging data is thread-safe. Many thanks to
          Ivan for finding and fixing the core issue here, and also thanks
          to those that tested the patch and provided test results. (closes
          issue #10571) (closes issue #10886) (closes issue #10875) (might
          close some others, as well ...) Patches: (from issue #10571)
          ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license
          229) - a few small changes by me

2007-10-16 21:14 +0000 [r85958]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Trying to remove a non-dynamic queue member via
          dynamic means can lead to some interesting (read nasty)
          situations. This patch clears up the issue by making only dynamic
          queue members removable via dynamic methods.

2007-10-16 19:41 +0000 [r85921]  Tilghman Lesher <tlesher@digium.com>

        * main/stdtime/localtime.c: Also set up gmtoff (this is used in the
          %z gnu extension to strftime) Reported and fixed by jcmoore
          Closes issue #11002

2007-10-16 19:10 +0000 [r85896]  Russell Bryant <russell@digium.com>

        * apps/app_voicemail.c: Remove a pointless lock.

2007-10-16 15:21 +0000 [r85852]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Fixing a double free which happens in the
          statechange thread. (closes issue #10987, reported by andrew)

2007-10-16 14:52 +0000 [r85818-85850]  Joshua Colp <jcolp@digium.com>

        * apps/app_hasnewvoicemail.c: Check to make sure a value has been
          given to the VMCOUNT dialplan function. (closes issue #10996)
          Reported by: marsosa

        * main/threadstorage.c: Fix memory allocation issue in
          threadstorage. (closes issue #10995) Reported by: snuffy Patches:
          new-patch.diff uploaded by snuffy (license 35)

2007-10-16 10:46 +0000 [r85800]  Philippe Sultan <philippe.sultan@gmail.com>

        * channels/chan_gtalk.c: Fix the output for this channel help CLI
          command

2007-10-15 21:10 +0000 [r85717-85720]  Russell Bryant <russell@digium.com>

        * apps/app_queue.c: Ensure that no pending state changes are leaked
          when the device state change thread gets stopped on module
          unload.

        * apps/app_queue.c: Previously, app_queue created a thread to
          handle every single device state change. I changed this a while
          ago in trunk for performance reasons. However, bug 8407 points
          out that it is actually a race condition, causing device state
          changes to get processed in random order. So, I backported my
          changes from trunk to 1.4. (closes issue #8407, patch provided by
          tim_ringenbach, committed patch by me)

2007-10-15 20:29 +0000 [r85687]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_stack.c: Don't execute a gosub if the arguments is
          zero-len (not just NULL) Reported by davevg Fixed by me Closes
          issue #10985

2007-10-15 20:21 +0000 [r85686]  Russell Bryant <russell@digium.com>

        * main/say.c: Add a small fix for the tw version of saying dates.
          (closes issue #7827) Reported by: sharkey Patches: say.nits.patch
          uploaded by sharkey (license 172)

2007-10-15 20:15 +0000 [r85684]  Jason Parker <jparker@digium.com>

        * Makefile: Properly use DESTDIR in 'config' target. Do not try to
          run chkconfig or similar if using DESTDIR. Issue 10938, patch by
          cabal95.

2007-10-15 19:22 +0000 [r85604-85649]  Russell Bryant <russell@digium.com>

        * main/utils.c: Be pedantic about handling memory allocation
          failure.

        * main/utils.c: The loop in the handler for the "core show locks"
          could potentially block for some amount of time. Be a little bit
          more careful and prepare all of the output in an intermediary
          buffer while holding a global resource. Then, after releasing it,
          send the output to ast_cli().

        * channels/chan_sip.c: Make the default for the srvlookup option to
          be yes. It doesn't really make sense for it to default to off.
          The default configuration file has it on, and proper RFC
          behavior, as indicated by a comment in the code, is for it to be
          on. So, let's have it on by default to make lives easier. (closes
          issue #10954, suggested by jtodd)

2007-10-15 16:39 +0000 [r85571]  Joshua Colp <jcolp@digium.com>

        * configs/features.conf.sample: Document that DTMF based features
          only work when two channels are bridged together. (closes issue
          #10773) Reported by: pbayley

2007-10-15 16:34 +0000 [r85561]  Russell Bryant <russell@digium.com>

        * include/asterisk/strings.h: Make a few changes so that characters
          in the upper half of the ISO-8859-1 character set don't get
          stripped when reading configuration. (closes issue #10982,
          dandre)

2007-10-15 16:22 +0000 [r85559]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: Bring both DTMF begin and end frames up through to
          the core for DTMF feature handling. (closes issue #10826)
          Reported by: dimas

2007-10-15 15:40 +0000 [r85556]  Russell Bryant <russell@digium.com>

        * pbx/pbx_dundi.c: Ensure the buffer passed to
          ast_canmatch_extension() is properly initialized so that it is
          null terminated. (issue #10977) Reported by: dimas Patches:
          pbxdundi.patch uploaded by dimas (license 88) - small mods by me

2007-10-15 14:55 +0000 [r85552]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: If Monitor or a spy was added to a P2P or native
          bridged channel bring the channel back to the generic bridging
          core so the monitor or spy operations work. (closes issue #10943)
          Reported by: julianjm

2007-10-15 13:16 +0000 [r85540-85548]  Russell Bryant <russell@digium.com>

        * main/db.c: Suppress a LOG_DEBUG message if debug is not enabled.
          (closes issue #10980) Reported by: casper Patches:
          db.c.84633.diff uploaded by casper (license 55)

        * main/asterisk.c: Make sure remote consoles unmute themselves
          again after reconnecting. (closes issue #10847) Reported by: atis
          Patches: console_unmute_on_reconnect.patch uploaded by atis
          (license 242)

        * main/utils.c: Make sure that the base64 decoder returns a
          terminated string. (closes issue #10979) Reported by: ys Patches:
          util.c.diff uploaded by ys (license 281) - small mods by me

        * pbx/pbx_config.c: Don't create the context for users in
          users.conf until we know at least one user exists. (closes issue
          #10971) Reported by: dimas Patches: pbxconfig.patch uploaded by
          dimas (license 88)

2007-10-13 15:26 +0000 [r85536]  Tilghman Lesher <tlesher@digium.com>

        * configs/extensions.ael.sample: Remove deprecated syntax from
          sample ael file Reported and patched by: dimas Closes issue
          #10967

2007-10-13 05:48 +0000 [r85532-85533]  Russell Bryant <russell@digium.com>

        * main/asterisk.c, main/cli.c, include/asterisk/logger.h: Fix an
          issue with console verbosity when running asterisk -rx to execute
          a command and retrieve its output. The issue was that there was
          no way for the main Asterisk process to know that the remote
          console was connecting in the -rx mode. The way that James has
          fixed this is to have all remote consoles muted by default. Then,
          regular remote consoles automatically execute a CLI command to
          unmute themselves when they first start up. (closes issue #10847)
          Reported by: atis Patches: asterisk-consolemute.diff.txt uploaded
          by jamesgolovich (license 176)

        * main/asterisk.c, main/cli.c, include/asterisk/cli.h: Properly
          handle the case where read() may return the text for more than
          one CLI command at once for a remote console. (closes issue
          #10888) Reported by: jamesgolovich Patches:
          asterisk-climultiple.diff.txt uploaded by jamesgolovich (license
          176)

2007-10-12 18:30 +0000 [r85523]  Tilghman Lesher <tlesher@digium.com>

        * doc/asterisk-mib.txt, doc/PEERING, LICENSE: Change Digium address

2007-10-12 15:45 +0000 [r85515-85517]  Russell Bryant <russell@digium.com>

        * res/res_smdi.c: Fix a spelling error in a log message. SMDI, not
          SDMI. (closes issue #10959)

        * pbx/pbx_realtime.c: Fix the potential use of an uninitialized
          buffer in a log message. (closes issue #10958) Reported by: dimas
          Patches: realtime.patch uploaded by dimas (license 88)

2007-10-11 15:26 +0000 [r85397]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: When creating a new packet don't try to stop
          retransmission of it. It was just allocated/created so it's
          impossible for it to have already been scheduled. (closes issue
          #10945) Reported by: flefoll Patches:
          chan_sip.c.br14.85280.xmit_reliable-patch uploaded by flefoll
          (license 244)

2007-10-11 04:35 +0000 [r85356]  Tilghman Lesher <tlesher@digium.com>

        * main/pbx.c: A dollar sign by itself, not indicating a start of a
          variable or expression prematurely ends substitution (closes
          issue #10939)

2007-10-10  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.13 released.

2007-10-10 15:56 +0000 [r85316]  Russell Bryant <russell@digium.com>

        * include/asterisk/file.h: I introduced a new member to the
          ast_filestream struct in 1.4.12, but put it in the middle of the
          struct, instead of at the end. One of the Debian folks, paravoid,
          pointed out that this breaks binary compatability with modules
          compiled against older headers. So, I'm moving the new member to
          the end of the struct to resolve the situation.

2007-10-10 15:51 +0000 [r85315]  Mark Michelson <mmichelson@digium.com>

        * main/utils.c: The thread ID should be unsigned.

2007-10-10 14:42 +0000 [r85277-85280]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: If devicestate is passed a port number strip
          it out. (closes issue #10930) Reported by: ibc

        * channels/chan_sip.c: Add support for handling a 182 Queued
          response. (closes issue #10924) Reported by: ramonpeek Patches:
          queued-182.diff uploaded by ramonpeek (license 266)

2007-10-10 14:26 +0000 [r85276]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: A bunch of changes from sprintf to
          snprintf. See security advisory AST-2002-022

2007-10-10 14:14 +0000 [r85242]  Joshua Colp <jcolp@digium.com>

        * apps/app_voicemail.c: Close voicemail message description file if
          duration did not meet the minimum, or else we will eventually run
          out of file descriptors. (closes issue #10918) Reported by:
          brak2718 Patches: vm1.4.12.1.patch uploaded by brak2718 (license
          279)

2007-10-10 06:24 +0000 [r85195]  Kevin P. Fleming <kpfleming@digium.com>

        * include/asterisk/frame.h: use a macro instead of an inline
          function, so that backtraces will report the caller of
          ast_frame_free() properly

2007-10-09 21:55 +0000 [r85158]  Tilghman Lesher <tlesher@digium.com>

        * main/channel.c, main/utils.c, include/asterisk/lock.h: This
          commit fixes the following issues: - Deadlock in ast_write (issue
          #10406) - Deadlock in ast_read (issue #10406) - Possible mutex
          initialization error in lock.h (issue #10571)

2007-10-09 14:30 +0000 [r84990-85093]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Don't perform a reinvite if a transfer is in
          progress. (issue #10915) Reported by: ramonpeek

        * main/rtp.c: Only update codec information if the channel has a
          technology private structure. (issue #10915) Reported by:
          ramonpeek

        * main/rtp.c: Update codec information as well as address when
          doing hold reinvites. (issue #10868) Reported by: mavince

        * main/channel.c: Don't keep trying to native bridge if either of
          the channels are involved in a masquerade operation to be done.
          (closes issue #10696) Reported by: tbelder

2007-10-08 03:28 +0000 [r84957]  Russell Bryant <russell@digium.com>

        * Makefile.rules: Enable file dependency tracking for _all_ builds,
          and not just for builds with dev-mode enabled. I have seen enough
          problems caused by this that I don't think it's worth keeping. I
          want to continue to encourage anybody that is interested to
          continue to run Asterisk from svn. Furthermore, I do not want
          their systems to break when we change a structure definition in a
          header file. :)

2007-10-07 16:15 +0000 [r84890-84902]  Philippe Sultan <philippe.sultan@gmail.com>

        * res/res_jabber.c: Presence packets from a client who's connected
          with our Jabber ID are valid, therefore, those clients must be
          considered as buddies. The resource string helps us make the
          distinction between clients. Closes issue #10707, reported by
          yusufmotiwala.

        * res/res_jabber.c: Prevent Asterisk from crashing when receiving a
          presence packet without resource from a buddy that is known to
          have a resource list. Revert a change I previously made, where
          Asterisk could point to a freed memory location.

2007-10-05 19:42 +0000 [r84851]  Tilghman Lesher <tlesher@digium.com>

        * main/db.c: Log exactly why we can't open the database, if we fail
          (closes issue #10887)

2007-10-05 18:55 +0000 [r84818]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: Update the remembered RTP peer information when
          putting an endpoint on hold or taking it off hold so that the RTP
          stack does not initiate a needless reinvite. (closes issue
          #10868) Reported by: mavince

2007-10-05 16:44 +0000 [r84783]  Russell Bryant <russell@digium.com>

        * channels/chan_zap.c: Do deadlock avoidance in a couple more
          places. You can't lock two channels at the same time without
          doing extra work to make sure it succeeds. (closes issue #10895,
          patch by me)

2007-10-05  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.12.1 released. (This is mainly to include the 
          app_queue fix for a memory leak on reload, but includes a couple
          of other bug fixes, as well.)

2007-10-05 01:39 +0000 [r84742]  Russell Bryant <russell@digium.com>

        * main/manager.c: Fix a copy/paste error in the description of
          UpdateConfig that was pointed out by JerJer on #asterisk-dev

2007-10-04 21:57 +0000 [r84692]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Don't allocate space for queue members unless
          it's needed. You end up deleting dynamic members on a reload. Not
          good. closes issue (#10879, reported by dazza76, patched by me)

2007-10-04 21:36 +0000 [r84690]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_zap.c: callers of sig2str already add the word
          'signalling' in the appropriate place, so don't duplicate it

2007-10-04 14:51 +0000 [r84637]  Joshua Colp <jcolp@digium.com>

        * apps/app_queue.c: Create a duplicate of the channel's member name
          as the tab completion stuff will free it. (closes issue #10884)
          Reported by: adamg

2007-10-03 22:59 +0000 [r84581]  Tilghman Lesher <tlesher@digium.com>

        * main/rtp.c: When an RFC 2833 event is sent that we don't
          recognize, ignore it, don't queue a NULL digit (closes issue
          #10877)

2007-10-03 18:20 +0000 [r84511-84544]  Steve Murphy <murf@digium.com>

        * pbx/pbx_ael.c: closes issue #10870 ; where a CUT() function call
          in a switch expr doesn't execute correctly, because the commas in
          the function args are not converted to vertbars before the func
          is called. I modified just the switch code to convert the commas
          to vertbars if there, but if more of these sort of probs are
          found, I may have to resort to something a little more
          fundamental. We'll see, I guess.

        * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
          pbx/ael/ael-test/ref.ael-vtest13,
          pbx/ael/ael-test/ref.ael-vtest17,
          pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
          pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c,
          pbx/ael/ael-test/ref.ael-test5: closes issue #10834 ; where a
          null input to a switch statement results in a hangup; since
          switch is implemented with extensions, and the default case is
          implemented with a '.', and the '.' matches 1 or more remaining
          characters, the case where 0 characters exist isn't matched, and
          the extension isn't matched, and the goto fails, and a hangup
          occurs. Now, when a default case is generated, it also generates
          a single fixed extension that will match a null input. That
          extension just does a goto to the default extension for that
          switch. I played with an alternate solution, where I just tack an
          extra char onto all the patterns and the goto, but not the
          default case's pattern. Then even a null input will still have at
          least one char in it. But it made me nervous, having that extra
          char in , even if that's a pretty secret and low-level issue.

2007-10-02  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.12 released.

2007-10-02 20:06 +0000 [r84474]  Russell Bryant <russell@digium.com>

        * Makefile, build_tools/prep_tarball: * Don't build the
          menuselect-tree for the tarball, as it requires running the
          configure script first * Change the Makefile to note that
          menuselect-tree depends on the configure script.

2007-10-02 19:01 +0000 [r84410-84437]  Jason Parker <jparker@digium.com>

        * res/res_features.c: Fix some odd formatting I missed..

        * res/res_features.c: Finish up on transferee channel before return
          on failure. Issue 10821, patch by Ivan

2007-10-02 14:12 +0000 [r84370]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: Use snprintf instead of sprintf in one
          place. There is no vulnerability here due to various buffer sizes
          around the code, but I still didn't like seeing a non
          length-limited copy of data coming off of the wire into a stack
          buffer, as this would be a problem in the future if buffer sizes
          elsewhere got changed or size limitations removed ...

2007-10-02 09:48 +0000 [r84345]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c: terminate USERUSER String with 0

2007-10-01 21:52 +0000 [r84291]  Jason Parker <jparker@digium.com>

        * Makefile, Makefile.rules, channels/Makefile: Add dist-clean
          support for subdirs. Change h323 to only remove the Makefile on a
          dist-clean, rather than a clean. This fixes a bug I found with
          trying to run make after a make clean

2007-10-01 21:25 +0000 [r84274]  Dwayne M. Hubbard <dhubbard@digium.com>

        * main/channel.c, main/manager.c, channels/chan_agent.c: moved
          get_base_channel() code from action_redirect to
          ast_channel_masquerade() for issue 7706 and BE-160

2007-10-01 21:18 +0000 [r84273]  Steve Murphy <murf@digium.com>

        * pbx/pbx_ael.c: Anything to keep gcc 4.2 happy...

2007-10-01 21:07 +0000 [r84271]  Russell Bryant <russell@digium.com>

        * main/utils.c, include/asterisk/lock.h: Fulfull a feature request
          from Qwell on the "core show locks" output. It will now note the
          lock type for each lock that a thread holds. (mutex, rdlock, or
          wrlock)

2007-10-01 20:27 +0000 [r84239]  Steve Murphy <murf@digium.com>

        * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: closes issue
          #10777 -- by returning a null for the parse tree when there's
          really nothing there, and making sure we don't try to do checking
          on a null tree.

2007-10-01 19:56 +0000 [r84166-84236]  Russell Bryant <russell@digium.com>

        * res/res_agi.c: Add another sanity check in the AGI read loop. We
          really don't care about EAGAIN unless we didn't read an entire
          line. If there is a newline at the end if the read buffer, break,
          because we got the whole thing. (reported and patched by bmd)

        * include/asterisk/lock.h: Show rwlocks in the "core show locks"
          output. Before, it only showed mutexes.

        * channels/Makefile: Remove another file in "make clean". (closes
          issue #10814, paravoid)

        * apps/app_dial.c: Simplify the CAN_EARLY_BRIDGE macro a bit.

2007-10-01 14:10 +0000 [r84158-84163]  Joshua Colp <jcolp@digium.com>

        * configs/usbradio.conf.sample (removed): Remove chan_usbradio
          config file from tree, it is not present in here. (closes issue
          #10839) Reported by: casper

        * res/res_musiconhold.c: Fix randomness. save_pos was being set to
          0 initially instead of -1, causing it to jump to position 0 when
          moh started. (closes issue #10859) Reported by: jamesgolovich
          Patches: asterisk-mohpos2.diff.txt uploaded by jamesgolovich
          (license 176)

        * apps/app_dial.c: Only attempt early bridging if the options given
          to Dial() permit it. (closes issue #10861) Reported by: peekyb

2007-09-30 20:02 +0000 [r84146]  Russell Bryant <russell@digium.com>

        * include/asterisk/module.h: Fix the AST_MODULE_INFO macro for C++
          modules. The load and reload parameters were in the wrong place.
          (closes issue #10846, alebm)

2007-09-29 23:00 +0000 [r84133-84135]  Steve Murphy <murf@digium.com>

        * pbx/ael/ael-test/ael-ntest22/t1/a.ael (added),
          pbx/ael/ael-test/ael-ntest22/t1/b.ael (added),
          pbx/ael/ael-test/ael-ntest22/t1/c.ael (added),
          pbx/ael/ael-test/ael-ntest22/t2/d.ael (added),
          pbx/ael/ael-test/ael-ntest22/t2/e.ael (added),
          pbx/ael/ael-test/ael-ntest22/t2/f.ael (added),
          pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22
          (added), pbx/ael/ael-test/ael-ntest22/t3/g.ael (added),
          pbx/ael/ael-test/ref.ael-test3,
          pbx/ael/ael-test/ael-ntest22/t3/h.ael (added),
          pbx/ael/ael-test/ref.ael-test4,
          pbx/ael/ael-test/ael-ntest22/t3/i.ael (added),
          pbx/ael/ael-test/ael-ntest22/t3/j.ael (added),
          pbx/ael/ael-test/ael-ntest22/qq.ael (added),
          pbx/ael/ael-test/ael-ntest22/t1 (added),
          pbx/ael/ael-test/ael-ntest22/t2 (added),
          pbx/ael/ael-test/ael-ntest22/t3 (added),
          pbx/ael/ael-test/ael-ntest22/extensions.ael (added),
          pbx/ael/ael-test/ael-ntest22 (added): This is a regression update
          that matches what I did in 84134 for AEL regressions.

        * pbx/ael/ael_lex.c, pbx/ael/ael.flex: This issue sort of closes
          10786; All config files support #include with globbing (you know,
          *,[chars],?,{list,list},etc), so I've updated the AEL system to
          support this also.

2007-09-28 14:13 +0000 [r84049-84078]  Tilghman Lesher <tlesher@digium.com>

        * main/say.c: Correct pronunciations of numbers for .nl (Closes
          issue #10837)

        * main/channel.c: Avoid a deadlock with ALL of the locks in the
          masquerade function, not just the pairs of channels. (Closes
          issue #10406)

2007-09-27 23:12 +0000 [r84018]  Dwayne M. Hubbard <dhubbard@digium.com>

        * main/manager.c, channels/chan_agent.c,
          include/asterisk/channel.h: if an Agent is redirected, the base
          channel should actually be redirected. This was causing multiple
          issues, especially issue 7706 and BE-160

2007-09-27 00:01 +0000 [r83976]  Russell Bryant <russell@digium.com>

        * pbx/pbx_dundi.c: remove a todo item that has been completed

2007-09-26 23:53 +0000 [r83974]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_alsa.c: avoid the weird usage of assert() in the
          ALSA header files that gcc 4.2 wants to complain about

2007-09-26 21:35 +0000 [r83910-83943]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: I changed my mind ... I think this should be
          a LOG_NOTICE.

        * channels/chan_sip.c: Add a log message that was requested by the
          masses in the developer tutorial session at Astricon. chan_sip
          did not output any message when a call was rejected because the
          extension was not found. This adds a verbose message (at verbose
          level 3) to note when this happens.

        * channels/chan_misdn.c: Fix building chan_misdn under dev-mode.
          (please run the configure script with --enable-dev-mode so this
          doesn't happen again ...)

2007-09-26 18:35 +0000 [r83879]  Tilghman Lesher <tlesher@digium.com>

        * channels/chan_zap.c: Remove unused 4k of memory on the program
          stack (closes issue #10827)

2007-09-25 14:13 +0000 [r83637-83773]  Tilghman Lesher <tlesher@digium.com>

        * main/app.c: jmls pointed out that unsetting the group and setting
          the group to the blank string aren't quite the same.

        * build_tools/make_defaults_h: In the source, keys are relative to
          the datadir, not varlib (which is the same in most cases, but
          it's good to be accurate). Closes issue #10811

        * doc/realtime.txt: Oops. Removed the unworkable workaround. This
          note should never have been in the release.

        * main/app.c: Making change to group splitting, as discussed on the
          -dev list. The main effect of this will be to permit
          Set(GROUP([cat])=), i.e. unsetting a group.

2007-09-24 07:54 +0000 [r83620]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c: fixed round_robin group dial method, this
          never worked well on BRI Ports (2 channels)

2007-09-22 19:39 +0000 [r83558-83589]  Steve Murphy <murf@digium.com>

        * pbx/pbx_ael.c: This closes issue #10788 -- The exact same fixes
          are made here for the first arg in the for(arg1; arg2; arg3) {}
          statement, as were done for the 3rd arg. It can now be an
          assignment that will embedded in a Set() app, or a macro call, or
          an app call.

        * pbx/pbx_ael.c: This closes issue #10788 -- the 3rd arg in the for
          statement is now wrapped in Set() only if there's an '=' in that
          string. Otherwise, if it begins with '&', then a Macro call is
          generated; otherwise it is made into an app call. A bit more
          accomodating, keeps the new guys happy, and the guys with ael-1
          code should be happy, too

2007-09-21 14:37 +0000 [r83432]  Russell Bryant <russell@digium.com>

        * main/rtp.c, channels/misdn_config.c, main/cdr.c, main/channel.c,
          channels/chan_misdn.c, pbx/ael/ael.tab.c, main/ast_expr2f.c,
          main/file.c, include/asterisk/sched.h, channels/chan_h323.c,
          pbx/pbx_dundi.c, utils/ael_main.c, main/ast_expr2.fl,
          channels/chan_mgcp.c, main/sched.c, res/res_config_pgsql.c,
          main/dnsmgr.c, channels/chan_sip.c, pbx/ael/ael.y,
          main/db1-ast/hash/hash.c, include/asterisk/channel.h,
          channels/chan_iax2.c: gcc 4.2 has a new set of warnings dealing
          with cosnt pointers. This set of changes gets all of Asterisk
          (minus chan_alsa for now) to compile with gcc 4.2. (closes issue
          #10774, patch from qwell)

2007-09-21 13:34 +0000 [r83400]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Fix video under certain circumstances. It
          would have been possible for the formats on the channel to not
          contain the video format. (closes issue #10782) Reported by:
          cwhuang

2007-09-20 21:16 +0000 [r83316-83348]  Russell Bryant <russell@digium.com>

        * main/asterisk.c: When daemonizing, don't change working directory
          to "/". It makes it not be able to do a core dump when not
          running as uid=root. (closes issue #10766, xrg)

        * contrib/scripts/safe_asterisk: Change safe_asterisk to explicitly
          ask for /bin/bash, as it uses bashisms. (closes issue #10772,
          reported by culrich)

2007-09-20 17:09 +0000 [r83246]  Jason Parker <jparker@digium.com>

        * apps/app_disa.c: If # is pressed after dialing an extension in
          DISA, stop trying to collect more digits. (issue #10754) Reported
          by: atis Patches: app_disa.c.branch.patch uploaded by atis
          (license 242) app_disa.c.trunk.patch uploaded by atis (license
          242)

2007-09-20 16:25 +0000 [r83230-83232]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Make sure the minimum T1 timer value is
          obeyed in all cases. (closes issue #10768) Reported by: flefoll
          Patches: chan_sip.c.trunk.83071.retrans-patch uploaded by flefoll
          (license 244) chan_sip.c.br14.83070.retrans-patch uploaded by
          flefoll (license 244)

        * channels/chan_sip.c: Fix a minor spelling error. (closes issue
          #10769) Reported by: flefoll Patches:
          chan_sip.c.trunk.83071.inita-patch uploaded by flefoll (license
          244) chan_sip.c.br14.83070.inita-patch uploaded by flefoll
          (license 244)

2007-09-19 19:50 +0000 [r83121-83179]  Russell Bryant <russell@digium.com>

        * apps/app_system.c: The System() and TrySystem() applications can
          take a substantial amount of time to execute while not servicing
          the channel. So, put the channel in autoservice while the command
          is being executed. (closes issue #10726, reported by mnicholson)

        * funcs/func_curl.c: Using curl can take a substantial amount of
          time, so the channel should be autoserviced while waiting for it
          to complete. (closes issue #10725, reported by mnicholson)

        * channels/chan_iax2.c: When handling a reload of chan_iax2, don't
          use an ao2_callback() to POKE all peers. Instead, use an
          iterator. By using an iterator, the peers container is not locked
          while the POKE is being done. It can cause a deadlock if the
          peers container is locked because poking a peer will try to lock
          pvt structs, while there is a lot of other code that will hold a
          pvt lock when trying to go lock the peers container. (reported to
          me directly by Loic Didelot. Thank you for the debug info!)

        * main/manager.c: Fix up another potential race condition. Do the
          loop decrementing use count on events with the eventq protected
          from being changed. (reported on IRC by Ivan)

2007-09-19 13:47 +0000 [r83070-83074]  Joshua Colp <jcolp@digium.com>

        * apps/app_queue.c: Protect the CDR record from modification by
          pbx_exec so that the application data contains the Queue data.
          (closes issue #10761) Reported by: snar Patches:
          app-queue-mixmonitor.patch uploaded by snar (license 245)

        * channels/chan_sip.c: (closes issue #10760) Reported by: dimas
          Patches: chan_sip.patch uploaded by dimas (license 88) Read in
          subscribecontext option in general to be the default.

2007-09-19 09:32 +0000 [r83023-83024]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c: removed comment which violates the coding
          guidelines.

        * channels/misdn_config.c, channels/chan_misdn.c,
          channels/misdn/chan_misdn_config.h: added 'astdtmf' option to
          allow configuring the asterisk dtmf detector instead of the
          mISDN_dsp ones. also added the patch from irroot #10190, so that
          dtmf tones detected by the asterisk detector are passed outofband
          to asterisk, to make any use of dtmf tones at all.

2007-09-19 00:19 +0000 [r82992]  Russell Bryant <russell@digium.com>

        * apps/app_flash.c: Change the description of app_flash to note how
          it can be a useful tool instead of just saying that it is
          generally a worthless feature. (Thanks to Jim Van Meggelen for
          pointing it out and providing the proposed text)

2007-09-18 23:41 +0000 [r82961]  Joshua Colp <jcolp@digium.com>

        * apps/app_queue.c: Initialize a variable to NULL to make the world
          happy.

2007-09-18 22:42 +0000 [r82929]  Russell Bryant <russell@digium.com>

        * include/asterisk/agi.h, res/res_agi.c: Add a new patch to handle
          interrupting the fgets() call when using FastAGI. This version of
          the patch maintains the original behavior of the code when not
          using FastAGI. (closes issue #10553) Reported by: juggie Patches:
          res_agi_fgets-4.patch uploaded by juggie (license 24)
          res_agi_fgets_1.4svn.patch uploaded by juggie (license 24) Slight
          mods by me Tested by: juggie, festr

2007-09-18 21:49 +0000 [r82887-82913]  Doug Bailey <dbailey@digium.com>

        * main/manager.c: Corrected patch applied in revision r82887.

        * main/manager.c: Fixed a bug where http manager sessions prevented
          the eventq from being cleaned out because http manager sessions
          do not have a valid file descriptor.

2007-09-18 20:56 +0000 [r82867]  Russell Bryant <russell@digium.com>

        * main/manager.c: Fix a memory leak that can occur on systems under
          higher load. The issue is that when events are appended to the
          master event queue, they use the number of active sessions as a
          use count so it will know when all active sessions at the time
          the event happened have consumed it. However, the handling of the
          number of sessions was not properly synchronized, so the use
          count was not always correct, causing an event to disappear
          early, or get stuck in the event queue for forever. (closes issue
          #9238, reported by bweschke, patch from Ivan, modified by me)

2007-09-18 20:09 +0000 [r82865]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Moving the logic for handling an empty
          membername to the create_member function so that there is a
          common place where this occurs instead of being spread out to
          several different places.

2007-09-18 18:59 +0000 [r82834]  Kevin P. Fleming <kpfleming@digium.com>

        * apps/app_queue.c: there is no need for conditional logic to
          select ->interface or ->membername, snince ->membername will
          always be populated

2007-09-18 16:31 +0000 [r82802]  Russell Bryant <russell@digium.com>

        * pbx/pbx_dundi.c: When copying the contents from the wildcard
          peer, do a deep copy instead of shallow copy so that it doesn't
          crash when beging destroyed. (closes issue #10546, patch by me)

2007-09-18 15:28 +0000 [r82751]  Jason Parker <jparker@digium.com>

        * configs/sip.conf.sample: Correct the allowexternaldomains option
          in SIP sample config. Issue 10753

2007-09-17 20:16 +0000 [r82594-82676]  Russell Bryant <russell@digium.com>

        * apps/app_voicemail.c, main/stdtime/localtime.c: Put a memset in
          ast_localtime() instead of a couple places in app_voicemail to
          prevent the problem everywhere instead of just a couple of
          places. (related to issue #10746)

        * apps/app_voicemail.c: Initialize some memory to fix crashes when
          leaving voicemail. This problem was fixed by running Asterisk
          under valgrind. (closes issue #10746, reported by arcivanov,
          patched by me) *** IMPORTANT NOTE: We need to check to see if
          this same bug exists elsewhere.

        * res/res_features.c: Handle the case where there are multiple
          dynamic features with the same digit mapping, but won't always
          match the activated on/by access controls. In that case, the code
          needs to keep trying features for a match. (reported by Atis on
          the asterisk-dev list, patched by me)

2007-09-17 16:40 +0000 [r82590-82592]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_iax2.c: revert a change that wasn't supposed to be
          committed... doh!

        * apps/app_queue.c, channels/chan_iax2.c: fix a couple of places
          where a logical member name (if specified) was not used, but
          instead the direct interface was listed

2007-09-17 02:00 +0000 [r82514]  Joshua Colp <jcolp@digium.com>

        * main/pbx.c: (closes issue #10734) Reported by: asgaroth Instead
          of passing a NULL pointer into snprintf pass "". It makes Solaris
          much happier.

2007-09-14 21:19 +0000 [r82444]  Steve Murphy <murf@digium.com>

        * main/cdr.c: closes issue #10668; thanks to arkadia for his patch;
          had to leave out the bit about ending the previous cdr in the
          fork; it would destroy current implementations.

2007-09-14 21:17 +0000 [r82435]  Russell Bryant <russell@digium.com>

        * configs/zapata.conf.sample: Add a note to help clarify the value
          set with the echocancel option. (inspired by Malcolm's blog post
          on blogs.digium.com about HPEC)

2007-09-14 18:35 +0000 [r82396-82398]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Crap, I broke the build. Fixed.

        * apps/app_queue.c: Adding member name field to manager events
          where they were missing before (closes issue #10721, reported by
          snar)

2007-09-14 17:48 +0000 [r82394]  Jason Parker <jparker@digium.com>

        * channels/chan_zap.c: If a channel does not have an owner, do not
          try to set a channel variable. This will end up making the
          channel variable global, which is not right. Closes issue #10720,
          patch by flefoll.

2007-09-14 15:50 +0000 [r82382-82385]  Russell Bryant <russell@digium.com>

        * build_tools/menuselect-deps.in, configure,
          include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
          checking for libusb here, so nobody has to deal with conflicts in
          the chan_usbradio-1.4 branch every time the configure script gets
          changed

        * channels/chan_usbradio.c (removed), channels/xpmr (removed),
          channels/Makefile: Remove chan_usbradio from the main 1.4 branch.
          It can't live here because we have a strict policy to not include
          new features in release branches. However, I'm going to merge it
          into trunk, and I also have a special 1.4 based branch that
          includes this module. svn co
          http://svn.digium.com/svn/asterisk/team/jdixon/chan_usbradio-1.4

2007-09-14 14:42 +0000 [r82376]  Mark Michelson <mmichelson@digium.com>

        * doc/CODING-GUIDELINES: Fixing a typo in the coding guidelines
          (closes issue #10717, reported and patched by leedm777)

2007-09-14 01:24 +0000 [r82368]  Jim Dixon <telesistant@hotmail.com>

        * apps/app_rpt.c: Fixed problem with changes made to cdr
          functionality

2007-09-14 00:52 +0000 [r82367]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_usbradio.c: this new driver may not live in this
          branch for long (since it is a new feature), but it definitely
          should not be built by default

2007-09-14 00:34 +0000 [r82366]  Jim Dixon <telesistant@hotmail.com>

        * apps/app_rpt.c, channels/xpmr/xpmr_coef.h (added),
          channels/chan_usbradio.c (added), channels/xpmr/xpmr.h (added),
          channels/xpmr (added), channels/xpmr/LICENSE (added),
          channels/xpmr/sinetabx.h (added), configs/usbradio.conf.sample
          (added), channels/Makefile, channels/xpmr/xpmr.c (added): Added
          channel driver for USB Radio device and support thereof.

2007-09-13 23:11 +0000 [r82358]  Jason Parker <jparker@digium.com>

        * pbx/pbx_spool.c: Fix a small typo. retrytime > waittime

2007-09-13 20:16 +0000 [r82346]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Preemptively fixing a possible segfault. It is
          possible that queuename is NULL (meaning pause ALL queues), so
          use q->name instead.

2007-09-13 20:11 +0000 [r82344]  Jason Parker <jparker@digium.com>

        * cdr/cdr_csv.c: Fix a crash that could occur in cdr_csv when
          mutliple threads tried to close the same file. Do we actually
          need the locking here? What happens if you open the same file
          twice, and two threads try to write to it at the same time? Is
          fputs() going to write out the entire line at once? I suspect
          that it could be possible for the second fopen to run during the
          first fputs, so the position could be in the middle of the
          previously written line... Issue 10347, initial patch by
          explidous (but I removed all of the paranoia stuff..)

2007-09-13 18:57 +0000 [r82337-82339]  Russell Bryant <russell@digium.com>

        * main/astobj2.c: resolve a warning when not building under dev
          mode

        * main/astobj2.c, main/asterisk.c, include/asterisk.h: Only compile
          in tracking astobj2 statistics if dev-mode is enabled. Also, when
          dev mode is enabled, register the CLI command that can be used to
          run the astobj2 test and print out statistics.

2007-09-13 18:12 +0000 [r82335]  Kevin P. Fleming <kpfleming@digium.com>

        * /, LICENSE: Merged revisions 82334 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r82334 | kpfleming | 2007-09-13 11:10:12 -0700 (Thu, 13 Sep 2007)
          | 2 lines clarify the OpenSSL and OpenH323 license exceptions
          ........

2007-09-13 16:25 +0000 [r82326]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Added logic to handle the unlikely case that
          someone has two queues with the same name. Asterisk will log a
          warning message letting the user know that one was already
          defined with that name and is it skipping all further instances.
          This also will work for realtime queues but in order for that to
          happen, the user would have to trigger a perfectly timed reload
          as a realtime queue is being looked up, which is highly unlikely
          (but taken care of nonetheless).

2007-09-13 11:47 +0000 [r82309]  Philippe Sultan <philippe.sultan@gmail.com>

        * channels/chan_gtalk.c: Closes issue #9401, reported and patched
          by irrot, with slight modifications by me. Handle DTMF sent by
          Asterisk properly.

2007-09-12 21:56 +0000 [r82296]  Russell Bryant <russell@digium.com>

        * res/res_agi.c: Fix a check of the wrong pointer, as pointed out
          by an XXX comment left in the code. The problem was harmless,
          however.

2007-09-12 21:28 +0000 [r82291]  Tilghman Lesher <tlesher@digium.com>

        * main/stdtime/tzfile.h: Oops, wrong location for FreeBSD zone
          files

2007-09-12 20:24 +0000 [r82286]  Dwayne M. Hubbard <dhubbard@digium.com>

        * apps/app_meetme.c: remove a race condition for the creation of
          recordthread's, and fix a small memory leak. This closes issue#
          10636

2007-09-12 20:12 +0000 [r82285]  Tilghman Lesher <tlesher@digium.com>

        * main/stdtime/private.h, main/stdtime/tzfile.h,
          include/asterisk/localtime.h, main/stdtime/localtime.c: Working
          on issue #10531 exposed a rather nasty 64-bit issue on
          ast_mktime, so we updated the localtime.c file from source. Next
          we'll have to write ast_strptime to match.

2007-09-12 15:16 +0000 [r82278-82280]  Russell Bryant <russell@digium.com>

        * main/asterisk.c: Clean up the output of "asterisk -h". This
          tweaks the wording and wraps lines at 80 characters. (closes
          issue #10699, seanbright)

        * res/res_agi.c: revert patch from issue #10553, as someone not
          using fastagi reported that this broke their system.

2007-09-12 14:30 +0000 [r82274-82276]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Accidentally committed changes to
          app_voicemail which do NOT need to be in the 1.4 branch yet.
          reverting...

        * apps/app_voicemail.c, apps/app_queue.c: We should only initialize
          a realtime queue when it is allocated, not every time we access
          it. This prevents the members ao2_container from being
          reallocated every time the queue is accessed. I also removed a
          debug message I had accidentally left in on a previous commit.

2007-09-11 22:37 +0000 [r82267]  Russell Bryant <russell@digium.com>

        * apps/app_queue.c: Fix incorrect uses of ao2_find(). Every one of
          these calls was reading bogus memory ...

2007-09-11 21:41 +0000 [r82265]  Joshua Colp <jcolp@digium.com>

        * codecs/gsm/src/lpc.c, codecs/gsm/src/long_term.c: (closes issue
          #10679) Reported by: andrew Build under dev mode when K6OPTS is
          enabled.

2007-09-11 20:49 +0000 [r82263]  Russell Bryant <russell@digium.com>

        * apps/app_queue.c: Fix another missing unref of member objects.
          This one was pointed out by Marta. When building the outgoing
          list in try_calling(), a member reference is stored in each
          outgoing entry. However, when this list got destroyed, the
          reference was not released.

2007-09-11 20:36 +0000 [r82261]  Steve Murphy <murf@digium.com>

        * main/cdr.c: this change should fix issue # 10659 -- what I worry
          about is how many other bug reports it may generate. Hopefully,
          we can please the/a majority. Hopefully. We shall see. Calls not
          marked ANSWERED and with only one channel name will not be
          posted. This should eliminate the double CDR's.

2007-09-11 16:05 +0000 [r82252]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: All instances of ao2_iterators which were just
          named 'i' have been renamed to 'mem_iter' so that when refcounted
          queues are merged into trunk, there will be little confusion
          regarding iterator names, especially when a queue and member
          iterator are used in the same function.

2007-09-11 16:03 +0000 [r82250]  Russell Bryant <russell@digium.com>

        * pbx/pbx_dundi.c: The sample dundi.conf claims support for a
          wildcard peer entry - [*], but the code did not support it. This
          patch makes it work. (closes issue #10546, patch by dds, with
          some changes by me)

2007-09-11 16:01 +0000 [r82249]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a
          hold/retrieve issue.

2007-09-11 15:26 +0000 [r82245]  Russell Bryant <russell@digium.com>

        * res/res_agi.c: (closes issue #10553) Reported by: juggie Patches:
          res_agi_fgets-2.patch uploaded by juggie (license 24) Tested by:
          juggie When using fastagi, fgets() can return before a full line
          is read. Add explicit handling for the case where it gets
          interrupted.

2007-09-11 14:56 +0000 [r82243]  Joshua Colp <jcolp@digium.com>

        * pbx/pbx_dundi.c: (closes issue #10577) Reported by: jamesgolovich
          Patches: asterisk-dundifree.diff.txt uploaded by jamesgolovich
          (license 176) Don't leak memory when unloading DUNDi.

2007-09-11 14:34 +0000 [r82198-82240]  Russell Bryant <russell@digium.com>

        * apps/app_queue.c: Add a couple more missing unrefs of queue
          member objects

        * apps/app_queue.c: Add a missing unref of a queue member in an
          error handling block

        * apps/app_queue.c: Document why membercount can not simply be
          replaced by ao2_container_count()

        * main/astobj2.c: backport astobj2 race condition fix. This
          function is the exact same as trunk so it applies here as well.

2007-09-10 18:02 +0000 [r82155]  Tilghman Lesher <tlesher@digium.com>

        * apps/app_queue.c: Convert struct member to use refcounts (closes
          issue #10199)

2007-09-10 15:02 +0000 [r82091]  Mark Michelson <mmichelson@digium.com>

        * configs/misdn.conf.sample: Removing non-existent options from
          misdn configuration sample. (closes issue #10678, reported and
          patched by IgorG)

2007-09-09 02:35 +0000 [r82028]  Tilghman Lesher <tlesher@digium.com>

        * include/asterisk/lock.h: Fix inline compiles on really old
          compilers (who uses gcc 2.7 anymore, really?)

2007-09-08 18:41 +0000 [r81952-81997]  Russell Bryant <russell@digium.com>

        * main/asterisk.c: Fix a small memory leak. ast_unregister_atexit()
          did not free the entry it removed.

        * .cleancount: (closes issue #10672) Bump the cleancount so that a
          "make clean" will be forced. This is needed because my fix in
          revision 81599 made a change to a data structure in file.h, and
          since file dependency tracking is only on with dev-mode enabled,
          file format modules that don't get rebuilt may crash, as is the
          case with this issue. This makes me wonder - how much faster does
          the code build without the file dependency tracking enabled? If
          it doesn't make much of a difference, then it may be worth just
          keeping it on all of the time, or perhaps just not in release
          tarballs, so that this type of issue is avoided.

2007-09-07 19:48 +0000 [r81923]  Jason Parker <jparker@digium.com>

        * apps/app_queue.c: Allow the MEMBERINTERFACE variable to be used
          as the mixmonitor filename. This moves the setting of the
          MEMBERINTERFACE variable to before mixmonitor. Issue 10671, patch
          by sim.

2007-09-07 15:25 +0000 [r81886]  Mark Michelson <mmichelson@digium.com>

        * configs/queues.conf.sample: Moving the explanation for joinempty
          to a more appropriate place

2007-09-06 22:28 +0000 [r81832]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: (closes issue #9724, closes issue #10374)
          Reported by: kenw Patches: 9724.txt uploaded by russell (license
          2) Tested by: kenw, russell Resolve a deadlock that occurs when
          doing a SIP transfer to parking. I come across this type of
          deadlock fairly often it seems. It is very important to mind the
          boundary between the channel driver and the core in respect to
          the channel lock and the channel-pvt lock. Channel drivers lock
          to lock the pvt and then the channel once it calls into the core,
          while the core will do it in the opposite order. The way this is
          avoided is by having channel drivers either release their pvt
          lock while calling into the core, or such as in this case,
          unlocking the pvt just long enough to acquire the channel lock.

2007-09-06 22:05 +0000 [r81778-81826]  Jason Parker <jparker@digium.com>

        * Makefile: We added COPTS for ASTCFLAGS additions, but not LDOPTS
          for ASTLDFLAGS. This adds LDOPTS

        * include/asterisk/astobj2.h: This should fix a build issue that
          people building against uClibc were seeing with the addition of
          astobj2

2007-09-06 19:40 +0000 [r81776]  Joshua Colp <jcolp@digium.com>

        * apps/app_meetme.c: (closes issue #10122) Reported by:
          stevefeinstein Patches: meetme-unmute-manager.diff uploaded by
          qwell (license 4) Tested by: stevefeinstein After looking over
          the code I agree with Qwell. Setting the file descriptor to
          conference each time just causes a fight back and forth.

2007-09-06 16:56 +0000 [r81743]  Philippe Sultan <philippe.sultan@gmail.com>

        * include/asterisk/jabber.h, channels/chan_gtalk.c: Various string
          length fixes. Removed an unused variable in aji_client structure
          (context)

2007-09-06 16:25 +0000 [r81682-81713]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Fixes an issue where valid DTMF had to be
          pressed twice to exit a queue if a member's phone was ringing.
          (closes issue #10655, reported by strider2k, patched by me)

        * res/res_features.c: Fixes a memory leak (closes issue #10658,
          reported and patched by Ivan)

2007-09-06 14:20 +0000 [r81650]  Philippe Sultan <philippe.sultan@gmail.com>

        * res/res_jabber.c: According to both RFC 3920 - section 9.1.2 -
          and Google's XMPP server complaint, if set, the 'from' attribute
          must be set to the user's full JID.

2007-09-05 20:53 +0000 [r81599]  Russell Bryant <russell@digium.com>

        * include/asterisk/file.h, main/say.c, res/res_features.c,
          main/file.c, include/asterisk/channel.h: Fix an issue that can
          occur when you do an attended transfer to parking. If you
          complete the transfer before the announcement of the parking spot
          finishes, then the channel being parked will hear the remainder
          of the announcement. These changes make it so that will not
          happen anymore. Basically, res_features sets a flag on the
          channel is playing the announcement to so that the file streaming
          core knows that it needs to watch out for a channel masquerade,
          and if it occurs, to abort the announcement. (closes BE-182)

2007-09-05 17:18 +0000 [r81569]  Tilghman Lesher <tlesher@digium.com>

        * include/asterisk/lock.h: Solaris x86 compatibility fix

2007-09-05 15:19 +0000 [r81525]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Fixing the build...

2007-09-05 15:14 +0000 [r81523]  Jason Parker <jparker@digium.com>

        * channels/chan_phone.c: Do not try to unregister a NULL channel
          tech. Also changed load_module function to use defines rather
          than numbers for return values. Issue 10651, patch by
          rbraun_proformatique, with additions by me.

2007-09-05 15:03 +0000 [r81520]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Reverting behavior of QUEUE_MEMBER_COUNT to
          only count members who are logged in and available. (related to
          issue #10652, reported by wuwu)

2007-09-05 13:11 +0000 [r81492]  Joshua Colp <jcolp@digium.com>

        * main/channel.c: (closes issue #10650) Reported by: tacvbo Only
          print out that the spy was removed while holding the spy lock.

2007-09-04 20:54 +0000 [r81453-81455]  Jason Parker <jparker@digium.com>

        * apps/app_followme.c: Rather than attempt to play a file, we can
          just check whether it exists. Issue 10634, patch by me, testing
          by pabelanger, sanity checked by bweschke

        * configs/followme.conf.sample: Change default followme config file
          to point to the correct files. Issue 10644, patch by pabelanger

2007-09-04 18:37 +0000 [r81448]  Russell Bryant <russell@digium.com>

        * main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c:
          Remove the typedefs on ao2_container and ao2_iterator. This is
          simply because we don't typedef objects anywhere else in
          Asterisk, so we might as well make this follow the same
          convention.

2007-09-04 16:40 +0000 [r81442]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_sip.c: there is no point in sending 401
          Unauthorized to a UAS that sent us a properly-formatted
          Authentication header with the expected username and nonce but an
          incorrect response (which indicates the shared secret does not
          match)... instead, let's send 403 Forbidden so that the UAS
          doesn't retry with the same authentication credentials repeatedly

2007-09-04 14:23 +0000 [r81435-81439]  Joshua Colp <jcolp@digium.com>

        * channels/chan_iax2.c: (closes issue #10632) Reported by:
          jamesgolovich Patches: asterisk-iaxfirmwareleak.diff.txt uploaded
          by jamesgolovich (license 176) Fix memory leak when unloading
          chan_iax2. The firmware files were not being freed.

        * main/channel.c: (closes issue #10476) Reported by: mdu113 Only
          look for the end of a digit when waiting for a digit. This in
          turn disables emulation in the core.

        * main/dns.c: (closes issue #10610) Reported by: john Patches:
          dns.c.patch uploaded by john (license 218) Tested by: mvanbaak
          Don't return a match if no SRV record actually exists.

2007-09-03 18:57 +0000 [r81433]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Remove a couple of calls to
          ast_string_field_free_pools() on peers in error handling blocks
          in the code for building peers. The peer object destructor does
          this and doing it twice will cause a crash. (closes issue #10625,
          reported by and patched by pnlarsson)

2007-09-01 15:57 +0000 [r81426-81428]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Changed a comment to be more accurate. (really
          this is just a test to make sure I can commit properly from home)

        * main/astobj2.c, include/asterisk/astobj2.h: Making match_by_addr
          into ao2_match_by_addr and making it available everywhere since
          it could be a handy callback to have

2007-08-31 21:27 +0000 [r81418]  Russell Bryant <russell@digium.com>

        * include/asterisk/astobj2.h: Remove references to a debugging
          parameter that does not exist

2007-08-31 19:48 +0000 [r81416]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Fixed broken behavior of a reload on realtime
          queues. Prior to this patch, if a reload was issued and a
          realtime queue had callers waiting in it, then the queue would be
          removed from the queue list, but it would not actually be freed
          (in fact, a debug message warning about a memory leak would come
          up). With this patch, reloads do not touch realtime queues at
          all.

2007-08-31 19:16 +0000 [r81415]  Tilghman Lesher <tlesher@digium.com>

        * funcs/func_logic.c: The IF() function was not allowing true
          values that had embedded colons (closes issue #10613)

2007-08-31 18:44 +0000 [r81412]  Jason Parker <jparker@digium.com>

        * apps/app_dial.c: Re-order dial options to be in line with the
          existing alpha order. Issue 10621, initial patch by junky

2007-08-31 17:38 +0000 [r81410]  Philippe Sultan <philippe.sultan@gmail.com>

        * channels/chan_gtalk.c: Make the 'gtalk show channels' CLI command
          available. Closes issue 10548, reported by keepitcool.

2007-08-31 15:53 +0000 [r81406]  Joshua Colp <jcolp@digium.com>

        * res/res_speech.c: Make it the engine's responsible to check for
          the presence of results.

2007-08-31 15:51 +0000 [r81405]  Kevin P. Fleming <kpfleming@digium.com>

        * codecs/codec_zap.c: add missing "transcoder show" (and deprecated
          "show transcoder") CLI commands that were in 1.2 but never added
          to 1.4

2007-08-31 14:38 +0000 [r81401-81403]  Joshua Colp <jcolp@digium.com>

        * res/res_features.c: (closes issue #10618) Reported by: dimas
          Don't pass through the stopped sounds frame.... just drop it.

        * res/res_features.c: (closes issue #10009) Reported by: dimas
          Don't output a bridge failed warning message if it failed because
          one of the channels was part of the masquerade process. That is
          perfectly normal.

2007-08-30 22:05 +0000 [r81397]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Removing an extraneous (and possibly
          misleading) log message. Firstly, if the announce file isn't
          found, the streaming functions will report it. Secondly, not all
          non-zero returns from play_file mean that the announce file
          wasn't found. Positive return values simply mean that a digit was
          pressed (most likely to skip through the announcement). (closes
          issue #10612, reported and patched by dimas)

2007-08-30 21:23 +0000 [r81395]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: (closes issue #10514) Reported by: casper
          Patches: chan_sip.c.80129.diff uploaded by casper (license 55)
          Remove needless check for AUTH_UNKNOWN_DOMAIN. It was impossible
          for it to ever be that value.

2007-08-30 21:11 +0000 [r81392]  Steve Murphy <murf@digium.com>

        * main/cdr.c: via issue 10599, where 'CDR already initialized'
          messages are being generated. Since all channels will have an
          init'd CDR attached at creation time, this message is now
          particularly useless. Removed.

2007-08-30 15:38 +0000 [r81383]  Russell Bryant <russell@digium.com>

        * channels/h323/ast_h323.cxx: Add missing checks for the PTRACING
          define. (closes issue #10559, paravoid)

2007-08-30 15:35 +0000 [r81381]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Changed some manager event messages to reflect
          whether a queue member is a realtime member or not

2007-08-30 15:33 +0000 [r81379]  Russell Bryant <russell@digium.com>

        * configs/modem.conf.sample (removed), configs/enum.conf.sample,
          configs/extensions.ael.sample: Fix a typo, update a reload
          command, and remove an unused configuration file. (closes issue
          #10606, casper)

2007-08-30 14:53 +0000 [r81375]  Joshua Colp <jcolp@digium.com>

        * main/pbx.c: (closes issue #10603) Reported by: jmls Patches:
          pbx.diff uploaded by jmls (license 141) Backport changes from
          81372. Add REASON dialplan variable for when an originated call
          fails and the failed extension is executed.

2007-08-30 14:43 +0000 [r81373]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c: Fixed some warnings.

2007-08-30 14:23 +0000 [r81369]  Joshua Colp <jcolp@digium.com>

        * res/res_features.c: (issue #10599) Reported by: dimas Handle the
          -1 control subclass during feature dialing (it indicates to stop
          sounds).

2007-08-30 08:31 +0000 [r81367]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/chan_misdn.c: Fixed a severe
          issue where a misdn_read would lock the channel, but read would
          not return because it blocks. later chan_misdn would try to queue
          a frame like a AST_CONTROL_ANSWER which could result in a
          deadlock situation. misdn_read will now not block forever
          anymore, and we don't queue the ANSWER frame at all when we
          already was called with misdn_answer -> answer would be called
          twice. Also we don't explicitly send a RELEASE_COMPLETE on
          receiption of a RELEASE anymore, because mISDN does that for us,
          this resulted in a problem on some switches, which would block
          our port after some calls for a short while.

2007-08-29 16:35 +0000 [r81346-81349]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: This patch, in essence, will correctly pause a
          realtime queue member and reflect those changes in the realtime
          engine. (issue #10424, reported by irroot, patch by me) This
          patch creates a new function called update_realtime_member_field,
          which is a generic function which will allow any one field of a
          realtime queue member to be updated. This patch only uses this
          function to update the paused status of a queue member, but it
          lays the foundation for persisting the state of a realtime member
          the same way that static members' state is maintained when using
          the persistentmembers setting

        * apps/app_queue.c: Changed some tabs to spaces

2007-08-29 15:57 +0000 [r81342]  Russell Bryant <russell@digium.com>

        * main/Makefile: If chan_h323 is not being built, don't use g++ to
          do the final link of Asterisk. (in response to a question on the
          asterisk-dev list)

2007-08-29 15:52 +0000 [r81340]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: This fix creates a more accurate way of
          detecting whether realtime members were deleted. (closes issue
          10541, reported by Alric, patched by me) The REALLY nice things
          about this patch is that queue members now have a "realtime"
          field which will be true if the member is a realtime member. This
          means we can check this value prior to certain processing if it
          should ONLY be done for realtime members.

2007-08-29 14:13 +0000 [r81331]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: (closes issue #9690) Reported by: mattv Make
          rtp timeouts work even if two RTP streams are directly bridged in
          the RTP stack.

2007-08-28 21:38 +0000 [r81226-81291]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Change the message about receiving a
          mini-frame before the first full voice frame to a DEBUG message.

        * pbx/pbx_dundi.c: revert unintentional changes in rev 81226

        * configs/indications.conf.sample, pbx/pbx_dundi.c: Add Russian
          tones. (closes issue #7953, hanabana)

2007-08-28 14:12 +0000 [r81120-81189]  Mark Michelson <mmichelson@digium.com>

        * contrib/scripts/vmail.cgi: Fixes a forwarding problem when using
          res_config_mysql (closes issue #10573, reported by chrisvaughan,
          patch suggested by chrisvaughan as well)

        * apps/app_queue.c: Resolve a potential deadlock. In this case, a
          single queue is locked, then the queue list. In changethread(),
          the queue list is locked, and then each individual queue is
          locked. Under the right circumstances, this could deadlock. As
          such, I have unlocked the individual queue before locking the
          queue list, and then locked the queue back after the queue list
          is unlocked.

        * channels/chan_agent.c: DTMF begin frames should be ignored so
          that when an agent acks a call with the '#' key, he doesn't cause
          a queue's announce file to be interrupted. Also went ahead and
          did the same for the '*' key and for ending a call. (closes issue
          #10528, reported by deskhack, patched by me)

2007-08-27 17:27 +0000 [r81042-81074]  Russell Bryant <russell@digium.com>

        * pbx/pbx_dundi.c: Add a \todo to note that this module leaks most
          of the memory it allocates on unload and should be fixed (when
          I'm not in the middle of something else ...).

        * pbx/pbx_dundi.c: explicity define a variable as a boolean

        * res/res_musiconhold.c: (closes issue #10419) Reported by:
          mustardman Patches: asterisk-mohposition.diff.txt uploaded by
          jamesgolovich (license 176) This patch fixes a few problems with
          music on hold. * Fix issues with starting at the beginning of a
          file when it shouldn't. * Fix the inuse counter to be decremented
          even if the class had not been set to be deleted when not in use
          anymore * Don't arbitrarily limit the number of MOH files to 255

2007-08-27 15:01 +0000 [r81012]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: (closes issue #10561) Reported by: jesselang
          Patches: chan_sip-ChannelReload-20080825.patch uploaded by
          jesselang (license 202) Remove an extra \r\n to make the
          ChannelReload event conform with every other event.

2007-08-27 14:55 +0000 [r81010]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Found a case where the queue's membercount is
          off. It does not take into account dynamic members on a reload.

2007-08-27 13:20 +0000 [r80974]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: (closes issue #10562) Reported by: idkpmiller Correct
          jitter value output in the CLI to be as expected.

2007-08-26 18:11 +0000 [r80932]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Remove an extra signal_condition() for the
          scheduler thread. (closes issue #10564, patch from casper)

2007-08-25 17:37 +0000 [r80895]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix some issues with the handling of the
          scheduler in chan_iax2. Most of the places that scheduled items
          to be executed by the scheduler thread did not signal the
          scheduler thread to wake up so that it could recalculate the time
          until the next action. These changes will make the scheduler
          thread more responsive and ensure that actions get executed as
          close to when intended as possible instead of it being possible
          for very long delays.

2007-08-24 22:59 +0000 [r80878]  Dwayne M. Hubbard <dhubbard@digium.com>

        * apps/app_zapateller.c: An empty string is an empty callerid ...
          so zap it. This closes issue #10502, which was pointed out by
          dswartz. Thank you, and may the swartz be with you

2007-08-24 21:22 +0000 [r80820-80849]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: If dnsmgr is in use, and no DNS servers are
          available when Asterisk first starts, then don't give up on
          poking peers. Allow the poke to get rescheduled so that it will
          work once the dnsmgr is able to resolve the host. (closes issue
          #10521, patch by jamesgolovich)

        * main/dsp.c: Improve the debouncing logic in the DTMF detector to
          fix some reliability issues. Previously, this code used a shift
          register of hits and non-hits. However, if the start of the digit
          isn't clean, it is possible for the leading edge detector to miss
          the digit. These changes replace the flawed shift register logic
          and also does the debouncing on the trailing edge as well.
          (closes issue #10535, many thanks to softins for the patch)

2007-08-24 19:52 +0000 [r80818]  BJ Weschke <bweschke@btwtech.com>

        * apps/app_queue.c: A minor correction to the available logic of
          autofill. If a queue member is paused, they're not really
          "available" so don't count them as such. Somewhat related to
          issue #10155

2007-08-24 18:52 +0000 [r80789]  Steve Murphy <murf@digium.com>

        * main/cdr.c: From a complaint by jmls, I realize that the message
          in cdr_disposition is unnecessary. To get failure disposition,
          just return -1; no use having more than one case do that.

2007-08-24 15:51 +0000 [r80750]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Fix a possible crash in IMAP voicemail.

2007-08-24 15:41 +0000 [r80747]  Tilghman Lesher <tlesher@digium.com>

        * main/pbx.c, UPGRADE.txt: Make the deprecation warning inline with
          the code, instead of only in documentation (closes issue #10549)

2007-08-24 15:28 +0000 [r80722]  Russell Bryant <russell@digium.com>

        * utils/ael_main.c: Tweak the formatting of this MODULEINFO block.
          I think this would have caused a "*" to get in the
          menuselect-tree file.

2007-08-24 14:48 +0000 [r80689-80717]  Steve Murphy <murf@digium.com>

        * utils/ael_main.c: This change addresses JerJer's complaint that
          aelparse builds and installs even if pbx_ael is unchecked in the
          menuselect stuff.

        * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/ael/ael-test/ref.ael-test6:
          backport of 80649, a fix to an unreported problem in the ael
          parser, that results in a crash on a 64bit machine

2007-08-24 11:42 +0000 [r80661]  Philippe Sultan <philippe.sultan@gmail.com>

        * channels/chan_gtalk.c: Closes issue #10509 Googletalk calls are
          answered too early, which results in CDRs wrongly stating that a
          call was ANSWERED when the calling party cancelled a call before
          before being established. We must not answer the call upon
          reception of a 'transport-accept' iq packet, but this packet
          still needs to be acknowledged, otherwise the remote peer would
          close the call (like in #8970).

2007-08-23 21:34 +0000 [r80601-80617]  Dwayne M. Hubbard <dhubbard@digium.com>

        * channels/misdn/isdn_lib.c: make misdn/isdn_lib compile without
          warnings

        * channels/chan_misdn.c: make chan_misdn compile without warnings

2007-08-23 20:16 +0000 [r80539-80573]  Russell Bryant <russell@digium.com>

        * include/asterisk/features.h, res/res_features.c: When executing a
          dynamic feature, don't look it up a second time by digit pattern
          after we already looked it up by name. This causes broken
          behavior if there is more than one feature defined with the same
          digit pattern. (closes issue #10539, reported by bungalow, patch
          by me)

        * funcs/func_timeout.c: Revert very broken fix for issue #10540 ...
          none of these values take ms so I don't know what I was thinking

        * funcs/func_timeout.c: Fix func_timeout to take values in floating
          point so 1.5 actually means 1.5 seconds instead of being rounded.
          (closes issue #10540, reported by spendergrass, patch by me)

2007-08-23 17:14 +0000 [r80505-80507]  Jason Parker <jparker@digium.com>

        * /: *sigh*

        * /: use autotagged externals

2007-08-23 17:08 +0000 [r80501]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_zap.c: report the actual channel number that was
          unregistered, instead of assuming that the interface list
          consists of channels 1 through <x> with no gaps in the sequence

2007-08-23 17:02 +0000 [r80360-80499]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix some code where it was possible for a
          reference to a peer to not get released when it should. Thank you
          to Marta Carbone for pointing this out!

        * main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c:
          This is a hack to maintain old behavior of chan_iax2. This
          ensures that if the peers and users are being stored in a linked
          list, that they go in the list in the same order that the older
          code used. This is necessary to maintain the behavior of which
          peers and users get matched when traversing the container.

        * res/res_agi.c: Revert res_agi fix that didn't quite work until we
          get it right ...

        * include/asterisk/astobj2.h: Add some more documentation on
          iterating ao2 containers. The documentation implies that is
          possible to miss an object or see an object twice while
          iterating. After looking through the code and talking with
          mmichelson, I have documented the exact conditions under which
          this can happen (which are rare and harmless in most cases).

        * main/astobj2.c: When converting this code to use the list macros,
          I changed it so objects are added to the head of a bucket instead
          of the tail. However, while looking over code with mmichelson, we
          noticed that the algorithm used in ao2_iterator_next requires
          that items are added to the tail. This wouldn't have caused any
          huge problem, but it wasn't correct. It meant that if an object
          was added to a container while you were iterating it, and it was
          added to the same bucket that the current element is in, then the
          new object would be returned by ao2_iterator_next, and any other
          objects in the bucket would be bypassed in the traversal.

        * channels/chan_sip.c: Don't crash when using realtime in chan_sip
          without an insecure setting in the database. (closes issue
          #10348, reported by link55, fixed by me)

        * main/astobj2.c (added), main/Makefile, include/asterisk/astobj2.h
          (added), doc/iax.txt, UPGRADE.txt, include/asterisk/strings.h,
          channels/chan_iax2.c: Merge changes from
          team/russell/iax_refcount. This set of changes fixes problems
          with the handling of iax2_user and iax2_peer objects. It was very
          possible for a thread to still hold a reference to one of these
          objects while a reload operation tries to delete them. The fix
          here is to ensure that all references to these objects are
          tracked so that they can't go away while still in use. To
          accomplish this, I used the astobj2 reference counted object
          model. This code has been in one of Luigi Rizzo's branches for a
          long time and was primarily developed by one of his students,
          Marta Carbone. I wanted to go ahead and bring this in to 1.4
          because there are other problems similar to the ones fixed by
          these changes, so we might as well go ahead and use the new
          astobj if we're going to go through all of the work necessary to
          fix the problems. As a nice side benefit of these changes, peer
          and user handling got more efficient. Using astobj2 lets us not
          hold the container lock for peers or users nearly as long while
          iterating. Also, by changing a define at the top of chan_iax2.c,
          the objects will be distributed in a hash table, drastically
          increasing lookup speed in these containers, which will have a
          very big impact on systems that have a large number of users or
          peers. The use of the hash table will be made the default in
          trunk. It is not the default in 1.4 because it changes the
          behavior slightly. Previously, since peers and users were stored
          in memory in the same order they were specified in the
          configuration file, you could influence peer and user matching
          order based on the order they are specified in the configuration.
          The hash table does not guarantee any order in the container, so
          this behavior will be going away. It just means that you have to
          be a little more careful ensuring that peers and users are
          matched explicitly and not forcing chan_iax2 to have to guess
          which user is the right one based on secret, host, and access
          list settings, instead of simply using the username. If you have
          any questions, feel free to ask on the asterisk-dev list.

        * res/res_agi.c: Juggie in #asterisk-dev was reporting problems
          where fgets would return without reading the whole line when
          using fastagi. When this happens, errno was set to EINTR or
          EAGAIN. This patch accounts for the possibility and lets fgets
          continue in that case.

2007-08-22 18:53 +0000 [r80302-80330]  Jason Parker <jparker@digium.com>

        * Makefile, build_tools/mkpkgconfig, build_tools/make_build_h,
          build_tools/strip_nonapi, build_tools/prep_moduledeps,
          build_tools/make_buildopts_h: Fix a few build issues in Solaris
          (and likely others). Use GREP and ID variables from autoconf.
          Reported to me in #asterisk-dev I forgot who reported this -
          sorry. :(

        * Makefile: Change a syntax that the GNU make in Solaris dislikes.

        * build_tools/make_version: Fix a bashism (we explicitly request
          /bin/sh). Remove some oddly placed quotes I found in passing.

2007-08-22 16:21 +0000 [r80257]  Russell Bryant <russell@digium.com>

        * Makefile: Honor the contents of the COPTS variable as custom
          target CFLAGS. Apparently this is what openwrt does. (reported by
          Brian Capouch on the asterisk-dev list, patch by me)

2007-08-22 16:14 +0000 [r80255]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: (closes issue #10526) Reported by: sinistermidget
          Revert commit from issue #10355 and return timestamp skew to 640.

2007-08-21  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.11 released.

2007-08-21 18:42 +0000 [r80183]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: Don't record SIP dialog history if it's not
          turned on. Also, put an upper limit on how many history entires
          will be stored for each SIP dialog. It is currently set to 50,
          but can be increased if deemed necessary. (closes issue #10421,
          closes issue #10418, patches suggested by jmoldenhauer, patches
          updated by me) (Security implications documented in AST-2007-020)

2007-08-21 16:39 +0000 [r80166-80167]  Steve Murphy <murf@digium.com>

        * include/asterisk/alaw.h, include/asterisk/ulaw.h: ugh. removing
          the diffs from ulaw.h and alaw.h for now; accidentally added them
          in 80166

        * main/alaw.c, include/asterisk/alaw.h, include/asterisk/ulaw.h:
          This patch solves problem 1 in 8126; it should not slow down the
          alaw codec, but should prevent signal degradation via multiple
          trips thru the codec. Fossil estimates the twice thru this codec
          will prevent fax from working. 4-6 times thru would result
          hearable, noticeable, voice degradation.

2007-08-21 15:22 +0000 [r80132]  Russell Bryant <russell@digium.com>

        * channels/chan_mgcp.c: Don't try to dereference the owner channel
          when it may not exist (issue #10507, maxper)

2007-08-21 15:03 +0000 [r80130]  Jason Parker <jparker@digium.com>

        * configs/cdr.conf.sample: (issue #10510) Reported by: casper
          Patches: cdr.conf.diff uploaded by casper (license 55) Fix a few
          errors in sample cdr config file.

2007-08-20 21:57 +0000 [r80088]  Russell Bryant <russell@digium.com>

        * apps/app_queue.c: Fix the build of app_queue

2007-08-20 21:39 +0000 [r80049-80086]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: After a discussion on #asterisk-dev, it was
          decided that this should be in 1.4 as well. (issue #10424,
          reported and patched by irroot)

        * apps/app_queue.c: Found a pointless ternary if. member->dynamic
          was set to 1 and has no opportunity to change between then and
          this line, so "dynamic" will ALWAYS be output.

2007-08-20 16:08 +0000 [r80047]  Jason Parker <jparker@digium.com>

        * configs/extensions.conf.sample: (issue #10499) Reported by:
          casper Patches: extensions.conf.sample.diff uploaded by casper
          (license 55) Update CLI examples in extensions.conf.sample to
          reflect command changes.

2007-08-20 15:34 +0000 [r80044]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Ukrainian language voicemail support.
          (closes issue #10458, reported and patched by Oleh)

2007-08-20 02:42 +0000 [r79998]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_voicemail.c: Missing curly braces. Oops. (Reported by
          snuffy via IRC)

2007-08-18 14:30 +0000 [r79947]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_voicemail.c: Don't allocate vmu for messagecount when we
          could just use the stack instead (closes issue #10490) Also,
          remove a useless (and leaky) SQLAllocHandle (closes issue #10480)

2007-08-17 21:01 +0000 [r79912]  Russell Bryant <russell@digium.com>

        * channels/chan_zap.c: Avoid a crash in the handling of DTMF based
          Caller ID. It is valid for ast_read to return NULL in the case
          that the channel has been hung up. (crash reported by
          anonymouz666 on IRC in #asterisk-dev)

2007-08-17 19:14 +0000 [r79906]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Patch allows for more seamless transition
          from file storage voicemail to ODBC storage voicemail. If a
          retrieval of a greeting from the database fails, but the file is
          found on the file system, then we go ahead an insert the greeting
          into the database. The result of this is that people who switch
          from file storage to ODBC storage do not need to rerecord their
          voicemail greetings.

2007-08-17 19:12 +0000 [r79902-79904]  Jason Parker <jparker@digium.com>

        * channels/chan_sip.c, main/utils.c, include/asterisk/strings.h:
          Don't send a semicolon over the wire in sip notify messages.
          Caused by fix for issue 9938. I basically took the code that
          existed before 9938 was fixed, and copied it into a new function
          - ast_unescape_semicolon There should be very few places this
          will be needed (pbx_config does NOT need this (see issue 9938 for
          details)) Issue 10430, patch by me, with help/ideas from murf
          (thanks murf).

        * channels/chan_local.c: Re-add the setting of callerid name and
          number. Issue 10485, reported by and fix explained by paradise.

2007-08-17 13:37 +0000 [r79857]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: Fix some crashes in chan_sip. This patch
          changes various places that add items to the scheduler to ensure
          that they don't overwrite the ID of a previously scheduled item.
          If there is one, it should be removed. (closes issue #10391,
          closes issue #10256, probably others, patch by me)

2007-08-17 08:22 +0000 [r79833]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c: sometimes we don't need to signal dtmf
          tones to asterisk, we just want them to go through as inband.
          Otherwise they might be generated by the other channel partner
          and then there is a double tone.

2007-08-16 22:32 +0000 [r79756-79792]  Russell Bryant <russell@digium.com>

        * res/res_musiconhold.c: Fix a little race condition that could
          cause a crash if two channels had MOH stopped at the same time
          that were using a class that had been marked for deletion when
          its use count hits zero.

        * res/res_musiconhold.c: This patch fixes a bug where reloading the
          module with "module reload" did not delete classes from memory
          that were no longer in the config. This patch fixes that problem
          as well as another one. Previously, if you reloaded MOH using the
          "moh reload" CLI command, which behaved differently than "module
          reload ...", MOH had to be stopped on every channel and started
          again immediately. However, there was no way to tell what class
          was being used, so they would all fall back to the default class.
          (closes issue #10139) Reported by: blitzrage Patches:
          asterisk-10139-advanced.diff.txt uploaded by jamesgolovich
          (license 176) Tested by: jamesgolovich

        * channels/chan_iax2.c: Fix more deadlocks in chan_iax2 that were
          introduced by making frame handling and scheduling
          multi-threaded. Unfortunately, we have to do some expensive
          deadlock avoidance when queueing frames on to the ast_channel
          owner of the IAX2 pvt struct. This was already handled for
          regular frames, but ast_queue_hangup and ast_queue_control were
          still used directly. Making these changes introduced even more
          places where the IAX2 pvt struct can disappear in the context of
          a function holding its lock due to calling a function that has to
          unlock/lock it to avoid deadlocks. I went through and fixed all
          of these places to account for this possibility. (issue #10362,
          patch by me)

2007-08-16 21:16 +0000 [r79690-79748]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_agent.c: Fixes a problem where agents would get
          stuck busy due to their wrapuptime being longer than the queue's
          wrapuptime and ringinuse=no for the queue. (closes issue #10215,
          reported by Doug, repaired by me) Special thanks to fkasumovic
          for pointing out the source of the problem and to bweschke for
          helping to come up with a solution!

        * apps/app_voicemail.c: base_encode is not trying to open a log
          file, so we should not call it a log file in the warning.
          (related to issue #10452, reported by bcnit)

2007-08-16 09:37 +0000 [r79665]  Philippe Sultan <philippe.sultan@gmail.com>

        * res/res_jabber.c: A fix for two critical problems detected while
          working with Daniel McKeehan in issue #10184. Upon priority
          change, the resource list is not NULL terminated when moving an
          item to the end of the list. This makes Asterisk endlessy loop
          whenever it needs to read the list. Jids with different resource
          and priority values, like in Gmail's and GoogleTalk's jabber
          clients put that problem in evidence. Upon reception of a 'from'
          attribute with an empty resource string, Asterisk crashes when
          trying to access the found->cap pointer if the resource list for
          the given buddy is not empty. This situation is perfectly valid
          and must be handled. The Gizmoproject's jabber client put that
          problem in evidence. Also added a few comments in the code as
          well as a handle for the capabilities from Gmail's jabber client,
          which are stored in a caps:c tag rather than the usual c tag.
          Closes issue #10184.

2007-08-16 08:21 +0000 [r79642]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/ie.c: 0x80 + protocol is wrong for USERUSER when
          we want to send IA5 Chars.

2007-08-15 14:40 +0000 [r79553]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: (closes issue #10440) Reported by: irroot (closes
          issue #10454) Reported by: flo_turc Increase maximum timestamp
          skew to 120. 20 was apparently far too low.

2007-08-15 14:26 +0000 [r79527]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Fixed an error in the Russian language
          voicemail intro. (issue #10458, reported and patched by Oleh)

2007-08-15 14:18 +0000 [r79523]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: (closes issue #10456) Reported by: irroot
          Patches: sip_timeout.patch uploaded by irroot (license 52) Change
          hardcoded timer value to defined value. I'm doing this in 1.4 as
          well so if it needs to be changed in the future this place would
          not have been forgotten.

2007-08-14 18:49 +0000 [r79436-79470]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix another spot where an iax2_peer would
          be leaked if realtime was in use.

        * channels/chan_iax2.c: Fix some memory leaks throughout chan_iax2
          related to the use of realtime. I found these while working on
          iax2_peer object reference tracking.

2007-08-14 15:27 +0000 [r79397]  Joshua Colp <jcolp@digium.com>

        * res/res_features.c: (closes issue #10415) Reported by: atis
          Revert fix for #10327 as it causes more issues then it solves.

2007-08-13 22:40 +0000 [r79363]  Steve Murphy <murf@digium.com>

        * pbx/pbx_ael.c: memset really, really needs to be used here.

2007-08-13 21:57 +0000 [r79334]  Joshua Colp <jcolp@digium.com>

        * res/res_speech.c, apps/app_speech_utils.c,
          include/asterisk/speech.h: Instead of accepting a single DTMF
          character accept a full string.

2007-08-13 20:37 +0000 [r79272-79301]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Don't call find_peer in
          registry_authrequest with the pvt lock held to avoid a deadlock.

        * channels/chan_iax2.c: Release the pvt lock before calling
          find_peer in register_verify to avoid a deadlock. Also, remove
          some unnecessary locking in auth_fail that was only done
          recursively.

        * channels/chan_iax2.c: Don't call find_peer within update_registry
          with a pvt lock held. This can cause a deadlock as the code will
          eventually call find_callno.

        * channels/chan_iax2.c: I am fighting deadlocks in chan_iax2. I
          have tracked them down to a single core issue. You can not call
          find_callno() while holding a pvt lock as this function has to
          lock another (every) other pvt lock. Doing so can lead to a
          classic deadlock. So, I am tracking down all of the code paths
          where this can happen and fixing them. The fix I committed
          earlier today was along the same theme. This patch fixes some
          code down the path of authenticate_reply.

2007-08-13 17:49 +0000 [r79255]  Steve Murphy <murf@digium.com>

        * pbx/ael/ael-test/ref.ael-vtest21 (added),
          pbx/ael/ael-test/ref.ael-test19,
          pbx/ael/ael-test/ael-vtest21/extensions.ael (added),
          pbx/ael/ael-test/ael-vtest21 (added),
          pbx/ael/ael-test/ref.ael-vtest17,
          pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
          pbx/ael/ael-test/ref.ael-test11, pbx/pbx_ael.c,
          pbx/ael/ael-test/ref.ael-test14, utils/ael_main.c: This patch
          fixes bug 10411. I added a new regression test, some regression
          test cleanups

2007-08-13 15:28 +0000 [r79214]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix a potential deadlock in socket_process.
          check_provisioning can eventually call find_callno. You can't
          hold a pvt lock while calling find_callno because it goes through
          and locks every single one looking for a match.

2007-08-13 14:51 +0000 [r79174-79207]  Joshua Colp <jcolp@digium.com>

        * res/res_speech.c, apps/app_speech_utils.c,
          include/asterisk/speech.h: Add an API call to allow the engine to
          know that DTMF was received.

        * channels/chan_oss.c, channels/chan_mgcp.c, channels/chan_phone.c,
          channels/chan_local.c, channels/chan_misdn.c,
          channels/chan_zap.c, channels/chan_sip.c, channels/chan_skinny.c,
          channels/chan_h323.c, channels/chan_gtalk.c,
          channels/chan_iax2.c: (closes issue #10437) Reported by: haklin
          Don't set the callerid name and number a second time on a newly
          created channel. ast_channel_alloc itself already sets it and
          setting it twice would cause a memory leak.

2007-08-11 05:23 +0000 [r79142]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * res/res_odbc.c: Ensure the connection gets marked as used at
          allocation time (closes issue #10429, report and fix by
          mnicholson)

2007-08-10 20:53 +0000 [r79044-79099]  Steve Murphy <murf@digium.com>

        * main/channel.c, pbx/pbx_spool.c, include/asterisk/channel.h: From
          a user complaint on #asterisk, I have forced pbx_spool to explain
          what reason codes mean, when they are logged

        * main/cdr.c: Re bug behavior mentioned in #asterisk, made this
          tweak to code, to prevent hundreds of log messages from being
          generated

        * main/cdr.c: This will help debug; from a question asked on
          #asterisk

2007-08-10  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.10.1 released.
        
2007-08-10 15:20 +0000 [r78995]  Russell Bryant <russell@digium.com>

        * include/asterisk/lock.h: The last set of changes that I made to
          "core show locks" made it not able to track mutexes unless they
          were declared using AST_MUTEX_DEFINE_STATIC. Locks initialized
          with ast_mutex_init() were not tracked. It should work now.

2007-08-10 14:15 +0000 [r78951-78955]  Joshua Colp <jcolp@digium.com>

        * main/file.c: Don't bother having the core pass through or emulate
          begin DTMF frames when in an ast_waitstream. It only cares about
          the end of DTMF.

        * configs/queues.conf.sample: (closes issue #10422) Reported by:
          bhowell Add note to sample configuration about module load order
          and how it can cause perfectly good queue members to be marked as
          invalid.

2007-08-10 13:24 +0000 [r78936]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c, channels/misdn/ie.c,
          channels/misdn/isdn_msg_parser.c: fixed a bug with the useruser
          information element. We send them now also in the disconnect
          message.

2007-08-09 23:47 +0000 [r78907]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Improved a bit of logic regarding
          comma-separated mailboxes in has_voicemail. Also added some
          braces to some compound if statements since unbraced if
          statements scare me in general.

2007-08-09 23:10 +0000 [r78891]  Steve Murphy <murf@digium.com>

        * Makefile: This fixes bug 10416; thanks to mvanbaak for the pretty
          output

2007-08-09 22:03 +0000 [r78826-78860]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Removing some extra debug code I left in my
          last commit

        * apps/app_voicemail.c: Quite a few changes regarding IMAP storage.
          1. instead of using inboxcount as the core message counting
          function, we use messagecount instead. This makes it possible to
          count messages in folders besides just INBOX and Old. 2.
          inboxcount and hasvoicemail now use messagecount as their means
          of determining return values. 3. Added a copy_message function
          for IMAP storage. Unfortunately I don't have the means to test
          it, but it seems like a pretty straightforward function. 4.
          Removed a #ifndef IMAP_STORAGE and matching #endif from
          leave_voicemail for a couple of reasons. One, we want to support
          copying mail to multiple IMAP boxes, and two, IMAP was broken
          because a STORE macro had been moved into this section of code.

        * channels/chan_sip.c: I broke canreinvite...Now I'm fixing it. I
          put some new code in the wrong place and so I've reverted the
          canreinvite section to how it was and put my new code where it
          should be.

2007-08-09 17:58 +0000 [r78717-78778]  Russell Bryant <russell@digium.com>

        * apps/app_voicemail.c: add a comment to indicate that inboxcount
          for ODBC_STORAGE needs to be fixed to support multiple mailboxes

        * apps/app_voicemail.c: Fix subscriptions to multiple mailboxes for
          ODBC_STORAGE. Also, leave a comment for this to be fixed for
          IMAP_STORAGE, as well. I left IMAP alone since I know MarkM was
          working on this code right now for another reason. This is broken
          even worse in trunk, but for a different reason. The fact that
          the mailbox option supported multiple mailboxes is completely not
          obvious from the code in the channel drivers. Anyway, I will fix
          that in another commit ...

        * apps/app_meetme.c: Fix a problem with the combination of the 'F'
          option to pass DTMF through a conference and options that use
          DTMF to activate various features. The problem was that the BEGIN
          frame would be passed through, but the END frame would get
          intercepted to activate a feature. Then, the other conference
          members would hear DTMF for forever, which they didn't seem to
          like very much. (closes issue #10400, reported by stevefeinstein,
          fixed by me)

2007-08-08 19:29 +0000 [r78646]  Jason Parker <jparker@digium.com>

        * doc/jabber.txt: Fix mogs email address.

2007-08-08 18:16 +0000 [r78575-78620]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Fixed some compiler warnings so that
          compiling with dev-mode and IMAP storage would not have any
          errors. This section of code may get changed again shortly since
          my change uncovers a rather silly bit of logic.

        * apps/app_queue.c: Changing a bit of logic so that someone will
          NEVER exit the queue on timeout unless they have enabled the 'n'
          option. This commit relates to issue #10320. Thanks to
          jfitzgibbon for detailing the idea behind this code change.

2007-08-08 13:51 +0000 [r78569]  Joshua Colp <jcolp@digium.com>

        * configs/sip.conf.sample: (closes issue #10335) Reported by:
          adamgundy Update sip.conf to include another scenario where
          directrtpsetup will fail.

2007-08-07  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.10 released.
        
2007-08-07 20:57 +0000 [r78488]  Russell Bryant <russell@digium.com>

        * res/res_config_odbc.c: Fix the build of this module on 64-bit
          platforms

2007-08-07 19:43 +0000 [r78450]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: The logic behind inboxcount's return value
          was reversed in has_voicemail and message_count. (closes issue
          #10401, reported by st1710, patched by me)

2007-08-07 19:34 +0000 [r78437]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * res/res_odbc.c: Don't free the environment handle when the
          connection fails, because other connections might be depending
          upon it

2007-08-07 19:11 +0000 [r78416]  Jason Parker <jparker@digium.com>

        * channels/chan_sip.c: Allow chan_sip to build in devmode

2007-08-07 19:09 +0000 [r78415]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_voicemail.c, res/res_config_odbc.c,
          apps/app_directory.c: Reconnection doesn't happen automatically
          when a DB goes down (fixes issue #9389)

2007-08-07 18:25 +0000 [r78375]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Properly check the capabilities count to
          avoid a segfault. (ASA-2007-019)

2007-08-07 17:45 +0000 [r78371]  Russell Bryant <russell@digium.com>

        * channels/chan_zap.c, /: Merged revisions 78370 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r78370 | russell | 2007-08-07 12:44:04 -0500 (Tue, 07 Aug 2007) |
          4 lines Revert patch committed for issue #9660. It broke E&M
          trunks. (closes issue #10360) (closes issue #10364) ........

2007-08-06 21:41 +0000 [r78275]  Joshua Colp <jcolp@digium.com>

        * main/channel.c: Add additional DTMF log messages to help when
          debugging issues.

2007-08-06 20:44 +0000 [r78184-78242]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix an issue where dynamic threads can get
          free'd, but still exist in the dynamic thread list. (closes issue
          #10392, patch from Mihai, with credit to his colleague, Pete)

        * include/asterisk/linkedlists.h: Fix the return value of
          AST_LIST_REMOVE(). This shouldn't be causing any problems,
          though, because the only code that uses the return value only
          checks to see if it is NULL. (closes issue #10390, pointed out by
          mihai)

2007-08-06 16:32 +0000 [r78182]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: It is possible for a transfer to occur
          before the remote device has our tag in which case they send none
          in the transfer. In this case we need to not fail the transfer
          dialog lookup.

2007-08-06 16:30 +0000 [r78180]  Jason Parker <jparker@digium.com>

        * main/config.c: Fix an issue with using UpdateConfig (manager
          action) where escaped semicolons in a config would be converted
          to just semicolons (\; to ;) Issue 9938

2007-08-06 15:27 +0000 [r78166-78172]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: (closes issue #10355) Reported by: wdecarne Now that
          we pass through RTP timestamp information we need to make the
          allowed timestamp skew considerably less. There are situations
          where a source may change and due to the timestamp difference the
          receiver will experience an audio gap since we did not indicate
          by setting the marker bit that the source changed.

        * configure, configure.ac: (closes issue #10383) Reported by: rizzo
          Include stdlib.h so NULL gets defined for gethostbyname_r checks.

2007-08-06 13:33 +0000 [r78164]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: Fixed a mistake I made in realtime_peer
          which caused it to return NULL every time. Thanks to Jon Fealy
          for emailing me the correction.

2007-08-05 14:18 +0000 [r78146]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * cdr/cdr_pgsql.c: Portability fix for devmode compiling (closes
          bug #10382)

2007-08-05 04:15 +0000 [r78143]  Russell Bryant <russell@digium.com>

        * include/asterisk/lock.h: Fix compilation failure when
          MALLOC_DEBUG is enabled, but DEBUG_THREADS is not

2007-08-05 03:29 +0000 [r78139]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * channels/chan_sip.c: If peer is not found, the error message is
          misleading (should be peer not found, not ACL failure)

2007-08-03 20:25 +0000 [r78103]  Mark Michelson <mmichelson@digium.com>

        * main/config.c, channels/chan_sip.c, include/asterisk/config.h:
          Changed the behavior of sip's realtime_peer function to match the
          corresponding way of matching for non-realtime peers. Now matches
          are made on both the IP address and port number, or if the
          insecure setting is set to "port" then just match on the IP
          address. In order to accomplish this, I also added a new API
          call, ast_category_root, which returns the first variable of an
          ast_category struct

2007-08-03 20:14 +0000 [r78028-78101]  Russell Bryant <russell@digium.com>

        * apps/app_voicemail.c: (closes issue #10194) Reported by:
          blitzrage Patches: bug0010194 uploaded by vovochka Tested by:
          blitzrage Fix a problem when you call Voicemail() with multiple
          mailboxes specified and ODBC_STORAGE is in use. The audio part of
          the message was only given to the first mailbox specified.

        * main/utils.c, include/asterisk/lock.h, main/astmm.c: Add some
          improvements to lock debugging. These changes take effect with
          DEBUG_THREADS enabled and provide the following: * This will keep
          track of which locks are held by which thread as well as which
          lock a thread is waiting for in a thread-local data structure. A
          reference to this structure is available on the stack in the
          dummy_start() function, which is the common entry point for all
          threads. This information can be easily retrieved using gdb if
          you switch to the dummy_start() stack frame of any thread and
          print the contents of the lock_info variable. * All of the
          thread-local structures for keeping track of this lock
          information are also stored in a list so that the information can
          be dumped to the CLI using the "core show locks" CLI command.
          This introduces a little bit of a performance hit as it requires
          additional underlying locking operations inside of every
          lock/unlock on an ast_mutex. However, the benefits of having this
          information available at the CLI is huge, especially considering
          this is only done in DEBUG_THREADS mode. It means that in most
          cases where we debug deadlocks, we no longer have to request
          access to the machine to analyze the contents of ast_mutex_t
          structures. We can now just ask them to get the output of "core
          show locks", which gives us all of the information we needed in
          most cases. I also had to make some additional changes to astmm.c
          to make this work when both MALLOC_DEBUG and DEBUG_THREADS are
          enabled. I disabled tracking of one of the locks in astmm.c
          because it gets used inside the replacement memory allocation
          routines, and the lock tracking code allocates memory. This
          caused infinite recursion.

        * channels/chan_iax2.c: Only pass through HOLD and UNHOLD control
          frames when the mohinterpret option is set to "passthrough". This
          was pointed out by Kevin in the middle of a training session.

        * channels/chan_iax2.c: Don't reuse the timespec that was set to 0
          in the previous timedwait as it will just return immediately.
          Also, fix some logic so the thread's lock isn't unlocked twice in
          the weird case of dynamic threads getting acquired right after a
          timeout. (pointed out by SteveK)

2007-08-02 21:53 +0000 [r77993-77996]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c, configs/skinny.conf.sample: Make sure we
          actually allow 6 chars to be sent. Also make note of the "A"
          option of date format. Issue 9779, modifications by DEA, wedhorn,
          and myself.

        * channels/chan_skinny.c: If a device disconnects, the session will
          go away. If this happens during call setup, we need to give up.
          Issue 10325.

2007-08-02 19:25 +0000 [r77949]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix the case where a dynamic thread times
          out waiting for something to do during the first time it runs.
          This shouldn't ever happen, but we should account for it anyway.
          (pointed out by pete, who works with mihai)

2007-08-02 18:42 +0000 [r77947]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Make sure we clear the prompt status
          message on a hangup. Also rearrange messages to better fit with
          what a wireshark trace shows it should be. Issue 10299, initial
          patch and solution by sbisker, modified by me to fit with
          wireshark trace.

2007-08-02 18:21 +0000 [r77945]  Steve Murphy <murf@digium.com>

        * main/fskmodem.c, /: Merged revisions 77942 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r77942 | murf | 2007-08-02 11:56:37 -0600 (Thu, 02 Aug 2007) | 1
          line This patch hopefully solves 10141; The user is running with
          it, and it doesn't appear to harm asterisk's operation, and may
          prevent a crash. I'll store it in 1.2, as we have shut down
          support on 1.2, but since I developed the patch before support
          finished, and it might affect 1.4 and trunk, I'm going ahead with
          it. ........

2007-08-02 18:04 +0000 [r77939-77943]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix another race condition in the handling
          of dynamic threads. If the dynamic thread timed out waiting for
          something to do, but was acquired to perform an action
          immediately afterwords, then wait on the condition again to give
          the other thread a chance to finish setting up the data for what
          action this thread should perform. Otherwise, if it immediately
          continues, it will perform the wrong action. (reported on IRC by
          mihai, patch by me) (related to issue #10289)

        * channels/chan_iax2.c: Add another sanity check to
          vnak_retransmit(). This check ensures that frames that have
          already been marked for deletion don't get retransmitted. (closes
          issue #10361, patch from mihai)

2007-08-02 15:15 +0000 [r77890-77894]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Make sure that we show the correct
          extension if dialed from a macro "From: 5555" rather than "From:
          s" Issue 10358, initial patch by DEA, reworked by me to use S_OR,
          tested by sbisker

        * channels/chan_skinny.c: Put in some additional debug information
          for softkey/stimulus messages. Issue 10291, patch by DEA.

2007-08-01 22:16 +0000 [r77887]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix some race conditions which have been
          causing weird problems in chan_iax2. The most notable problem is
          that people have been seeing storms of VNAK frames being sent due
          to really old frames mysteriously being in the retransmission
          queue and never getting removed. It was possible that a dynamic
          thread got created, but did not acquire its lock before the
          thread that created it signals it to perform an action. When this
          happens, the thread will sleep until it hits a timeout, and then
          get destroyed. So, the action never gets performed and in some
          cases, means a frame doesn't get transmitted and never gets freed
          since the scheduler never gets a chance to reschedule
          transmission. Another less severe race condition is in the
          handling of a timeout for a dynamic thread. It was possible for
          it to be acquired to perform at action at the same time that it
          hit a timeout. When this occurs, whatever action it was acquired
          for would never get performed. (patch contributed by Mihai and
          SteveK) (closes issue #10289) (closes issue #10248) (closes issue
          #10232) (possibly related to issue #10359)

2007-08-01 22:14 +0000 [r77886]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_voicemail.c: Voicemail with ODBC_STORAGE defined does
          not compile cleanly (missing def)

2007-08-01 21:08 +0000 [r77883]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Fix an issue that caused one-way audio on
          some newer devices (specifically the 7921), due to sending
          packets in the wrong order during hangup. Also make sure we clear
          tones/messages on the correct line/instance. Issue 10291, patch
          by DEA, tested by sbisker and myself.

2007-08-01 18:08 +0000 [r77863-77871]  Joshua Colp <jcolp@digium.com>

        * main/cli.c: (closes issue #10351) Reported by: ftarz Some
          platforms don't like it when you pass NULL to vsnprintf so pass
          "" instead.

        * include/asterisk/threadstorage.h, channels/chan_mgcp.c,
          apps/app_voicemail.c, main/acl.c, utils/smsq.c,
          channels/chan_iax2.c: Add some fixes for building on Solaris.

        * main/utils.c: Whoops, I meant R_5 not R5.

        * configure, configure.ac: And for my last trick... make sure that
          if gethostbyname_r is exported by a library that it is used.

        * configure, include/asterisk/autoconfig.h.in, configure.ac,
          main/utils.c: Extend autoconf logic to determine which version of
          gethostbyname_r is on the system.

2007-08-01 14:08 +0000 [r77852-77854]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Fixes an issue I introduced to queues wherein a
          queue with joinempty=yes would kick people out of the queue
          because of erroneously thinking the 'n' option was in use.
          (closes issue #10320, reported by jfitzgibbon, patched by me,
          tested by blitzrage and me) Thank you blitzrage for all the
          testing you've done lately with queues! It's much appreciated!

        * apps/app_queue.c: If a queue uses dynamic realtime members, then
          the member list should be updated after each attempt to call the
          queue. This fixes an issue where if a caller calls into a queue
          where no one is logged in, they would wait forever even if a
          member logged in at some point. (closes issue #10346, reported by
          and tested by blitzrage, patched by me)

2007-07-31 21:09 +0000 [r77845-77846]  Jim Dixon <telesistant@hotmail.com>

        * apps/app_rpt.c: Much newer version, 0.70 with much additions

        * main/dsp.c, channels/chan_zap.c: Made VAST improvements in DTMF
          receiver in RADIO_RELAX mode (thanx Steve W9SH), and oversight in
          logic in TONE_VERIFY/RELAX mode in chan_zap.

2007-07-31 20:59 +0000 [r77844]  Steve Murphy <murf@digium.com>

        * /, contrib/scripts/ast_grab_core: Merged revisions 77842 via
          svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r77842 | murf | 2007-07-31 13:19:35 -0600 (Tue, 31 Jul 2007) | 1
          line This probably isn't super-general, but it's a first stab at
          using kill -11 to generate a core file instead of gcore. ........

2007-07-31 16:17 +0000 [r77831]  Joshua Colp <jcolp@digium.com>

        * res/res_speech.c, include/asterisk/speech.h: Add a flag to the
          speech API that allows an engine to set whether it received
          results or not.

2007-07-31 15:53 +0000 [r77827]  Kevin P. Fleming <kpfleming@digium.com>

        * build_tools/cflags.xml: DETECT_DEADLOCKS can't be enabled without
          DEBUG_THREADS or it does nothing

2007-07-31 15:21 +0000 [r77824]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: This patch makes Asterisk send 100 Trying
          provisional responses upon receipt of re-invites. This makes it
          so that if there are two or more Asterisk servers between
          endpoints, the Asterisk servers will not keep retransmitting the
          re-invites. (closes issue #10274, reported by cstadlmann, patched
          by me with approval from file)

2007-07-30 20:17 +0000 [r77795]  Jason Parker <jparker@digium.com>

        * main/say.c: Applications like SayAlpha() should not hang up the
          channel if you request an "unknown" character such as a comma.
          Instead, skip the character and move on. Issue 10083, initial
          patch by jsmith, modified by me.

2007-07-30 20:16 +0000 [r77785-77794]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix an issue that could potentially cause
          corruption of the global iax frame queue. In the network_thread()
          loop, it traverses the list using the AST_LIST_TRAVERSE_SAFE
          macro. However, to remove an element of the list within this
          loop, it used AST_LIST_REMOVE, instead of
          AST_LIST_REMOVE_CURRENT, which I believe could leave some of the
          internal variables of the SAFE macro invalid. Mihai says that he
          already made this change in his local copy and it didn't help his
          VNAK storm issues, but I still think it's wrong. :)

        * res/res_agi.c: (closes issue #10279) Reported by: seanbright
          Patches: res_agi.carefulwrite.1.4.07252007.patch uploaded by
          seanbright (license 71) res_agi.carefulwrite.trunk.07252007.patch
          uploaded by seanbright (license 71) Allow the "agi_network: yes"
          line to be printed out in the AGI debug output. Also, allow
          partial writes to be handled when writing out this line just like
          it is for all of the others.

        * main/channel.c: file and I both committed changes for issue
          #10301. Remove a duplicated assignment to restore the original
          value of the previous channel.

2007-07-30 18:43 +0000 [r77783]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * /, res/res_agi.c: Merged revisions 77782 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r77782 | tilghman | 2007-07-30 13:40:54 -0500 (Mon, 30 Jul 2007)
          | 2 lines Revert change in revision 71656, even though it fixed a
          bug, because many people were depending upon the (broken)
          behavior. ........

2007-07-30 17:29 +0000 [r77780]  Russell Bryant <russell@digium.com>

        * main/channel.c: (closes issue #10301) Reported by: fnordian
          Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian
          (license 110) Additional changes by me Fix some problems in
          channel_find_locked() which can cause an infinite loop. The
          reference to the previous channel is set to NULL in some cases.
          These changes ensure that the reference to the previous channel
          gets restored before needing it again. I'm not convinced that the
          code that is setting it to NULL is really the right thing to do.
          However, I am making these changes to fix the obvious problem and
          just leaving an XXX comment that it needs a better explanation
          that what is there now.

2007-07-30 17:11 +0000 [r77768-77778]  Joshua Colp <jcolp@digium.com>

        * res/res_features.c: (closes issue #10327) Reported by: kkiely
          Instead of directly mucking with the extension/context/priority
          of the channel we are transferring when it has a PBX simply call
          ast_async_goto on it. This will ensure that the channel gets
          handled properly and sent to the right place.

        * main/channel.c: (closes issue #10301) Reported by: fnordian
          Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian
          (license 110) Restore previous behavior where if we failed to
          lock the channel we wanted we would return to exactly the same
          point as if we had just reentered the function.

        * /, apps/app_macro.c: Merged revisions 77767 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r77767 | file | 2007-07-30 11:50:02 -0300 (Mon, 30 Jul 2007) | 4
          lines (closes issue #10334) Reported by: ramonpeek Pass through
          the return value from macro_exec through the MacroIf application.
          ........

2007-07-27 18:15 +0000 [r77571]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * res/res_odbc.c: Missing newline

2007-07-27 17:04 +0000 [r77536-77540]  Joshua Colp <jcolp@digium.com>

        * cdr/cdr_pgsql.c: (closes issue #10310) Reported by: prashant_jois
          Patches: cdr_pgsql.patch uploaded by prashant (license 114)
          Finish the Postgresql connection after the log messages are
          printed so we don't access invalid memory.

        * channels/chan_sip.c: (closes issue #10323) Reported by: julianjm
          Patches: chan_sip_device_state_hold_fix.v1.diff.txt uploaded by
          julianjm (license 99) Clear ONHOLD flag when decrementing the
          onHold peer count. If we did not do this the count may keep
          decreasing.

2007-07-27 14:30 +0000 [r77490]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_sip.c: "re-invite" was misspelled

2007-07-26 23:19 +0000 [r77460]  Joshua Colp <jcolp@digium.com>

        * main/channel.c: (closes issue #10302) Reported by: litnialex If a
          DTMF end frame comes from a channel without a begin and it is
          going to a technology that only accepts end frames (aka INFO)
          then use the minimum DTMF duration if one is not in the frame
          already.

2007-07-26 22:16 +0000 [r77424-77429]  Kevin P. Fleming <kpfleming@digium.com>

        * doc/mp3.txt: change protocol for downloads as well

        * doc/mp3.txt, sounds/Makefile: use new canonical name for download
          server

2007-07-26 21:23 +0000 [r77410]  Russell Bryant <russell@digium.com>

        * Makefile, build_tools/make_buildopts_h: AST_DEVMODE was defined
          in trunk, but not in 1.4. When Asterisk is compiled under dev
          mode, AST_DEVMODE will get defined in buildopts.h. Change 1.4 to
          define it in the same way that trunk does. Also, revert the
          change that added this define in the Makefile The advantage to
          doing it this way is that buildopts.h gets installed when you
          install Asterisk. Then, when building any out of tree modules, or
          building asterisk-addons, these modules know which options the
          rest of Asterisk was built with.

2007-07-26 20:35 +0000 [r77380]  Mark Michelson <mmichelson@digium.com>

        * Makefile, main/logger.c: Fixes to get ast_backtrace working
          properly. The AST_DEVMODE macro was never defined so the majority
          of ast_backtrace never attempted compilation. The makefile now
          defines AST_DEVMODE if configure was run with --enable-dev-mode.
          Also, changes were made to acccomodate 64 bit systems in
          ast_backtrace. Thanks to qwell, kpfleming, and Corydon76 for
          their roles in allowing me to get this committed

2007-07-26 19:32 +0000 [r77348-77350]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * main/logger.c: Missed one

        * main/logger.c: Oops, that builtin define should be all-lowercase.

2007-07-26 18:30 +0000 [r77318]  Mark Michelson <mmichelson@digium.com>

        * cdr/cdr_pgsql.c: Two consecutive calls to PQfinish could occur,
          meaning free gets called on the same variable twice. This patch
          sets the connection to NULL after calls to PQfinish so that the
          problem does not occur. Also in this patch, prashant_jois
          informed me that it is safe to pass a null pointer to PQfinish,
          so I have removed the check for conn's existence from
          my_unload_module. (closes issue 10295, reported by junky, patched
          by me with input from prashant_jois)

2007-07-25 22:39 +0000 [r77191]  Steve Murphy <murf@digium.com>

        * apps/app_meetme.c: This fix solves problem with intense squelch
          noise when someone joins conf in bug 9430; We repro'd the problem
          with meetme opts of 'CciMo'; Josh Colp supplied this patch, and
          I'm applying it. It looks like playing the recorded username will
          louse up the next thing played into the channel. Josh rearranged
          the code so as to start things over before playing data directly
          into the conference.

2007-07-25 22:16 +0000 [r77176]  Joshua Colp <jcolp@digium.com>

        * apps/app_speech_utils.c: (closes issue #10303) Reported by: jtodd
          Add SPEECH_DTMF_TERMINATOR variable so the user can specify the
          digit to terminate a DTMF string with. If none is specified then
          no terminator will be used.

2007-07-25 21:52 +0000 [r77154]  Mark Michelson <mmichelson@digium.com>

        * main/channel.c: chan->emulate_dtmf_duration is an unsigned int,
          not a signed int, so use %u instead of %d in the format string

2007-07-25 20:23 +0000 [r77116-77136]  Jason Parker <jparker@digium.com>

        * /: so are my fingers...

        * /: autotagexternals script is still obviously misbehaving...

        * /: use autotagged externals

2007-07-25 17:14 +0000 [r77071]  Joshua Colp <jcolp@digium.com>

        * configure, acinclude.m4: Fix autoconf logic for finding OpenH323
          when it is not in the first place searched (/usr/share/openh323).

2007-07-25 09:34 +0000 [r77022]  Luigi Rizzo <rizzo@icir.org>

        * main/rtp.c: set the sequence number in a frame for all frame
          types

2007-07-25 00:18 +0000 [r76983]  Steve Murphy <murf@digium.com>

        * channels/chan_zap.c, /: Merged revisions 76978 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1
          line this fixes bug 10293, where the error message because
          defaultzone or loadzone was not defined was confusing ........

2007-07-24 22:12 +0000 [r76891-76937]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * /, include/asterisk/lock.h: Merged revisions 76934 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r76934 | tilghman | 2007-07-24 17:11:33 -0500 (Tue, 24
          Jul 2007) | 2 lines Oops, res contains the error code, not errno.
          I was wondering why a mutex was reporting "No such file or
          directory"... ........

        * main/app.c: Found another place where we should be using the
          umask (thanks jcmoore)

2007-07-24  Jason Parker <jparker@digium.com>

        * Asterisk 1.4.9 released.

2007-07-24 16:42 +0000 [r76803-76805]  Jason Parker <jparker@digium.com>

        * channels/chan_iax2.c: Don't create the Asterisk channel until we
          are starting the PBX on it. (ASA-2007-018)

2007-07-24 16:26 +0000 [r76801]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Added a membercount variable to call_queue
          struct which keeps track of the number of logged in members in a
          particular queue. This makes it so that the 'n' option for
          Queue() can act properly depending on which strategy is used. If
          the strategy is roundrobin, rrmemory, or ringall, we want to ring
          each phone once before moving on in the dialplan. However, if any
          other strategy is used, we will only ring one phone since it
          cannot be guaranteed that a different phone will ring on
          subsequent attempts to ring a phone. As a side effect of this,
          the QUEUE_MEMBER_COUNT dialplan function now just reads the
          membercount variable instead of traversing through the member
          list to figure out how many members there are. Special thanks to
          blitzrage for helping to test this out. (closes issue #10127,
          reported by bcnit, patched by me, tested by blitzrage)

2007-07-23 22:38 +0000 [r76708]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_voicemail.c: It was our stated intention for 1.4 that
          files created in app_voicemail should depend upon the umask.
          Unfortunately, mkstemp() creates files with mode 0600, regardless
          of the umask. This corrects that deficiency.

2007-07-23 18:59 +0000 [r76656]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Fix some incorrect softkey labels in
          messages. Don't try to play dialtone in some unimplemented
          features.

2007-07-23 18:29 +0000 [r76654]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_agent.c: Merged revisions 76653 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul
          2007) | 4 lines (closes issue #5866) Reported by: tyler Do not
          force channel format changes when a generator is present. The
          generator may have changed the formats itself and changing them
          back would cause issues. ........

2007-07-23 17:57 +0000 [r76620]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Don't try to queue up hold/unhold frames
          on a non-existent channel. Issue 10276.

2007-07-23 17:48 +0000 [r76519-76618]  Joshua Colp <jcolp@digium.com>

        * apps/app_morsecode.c: Allow app_morsecode to build on PPC Linux
          by putting the value of the digit char in an int.

        * /, channels/chan_sip.c: Merged revisions 76560 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6
          lines (closes issue #10236) Reported by: homesick Patches:
          rpid_1.4_75840.patch uploaded by homesick (license 91) Accept
          Remote Party ID on guest calls. ........

        * channels/chan_skinny.c: (closes issue #10268) Reported by:
          mvanbaak Patches: chan_skinny_openbsd.diff uploaded by mvanbaak
          (license 7) Add another OS that has to use the Macros for byte
          ordering.

2007-07-23 12:25 +0000 [r76485]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Use a signed integer for storing the number
          of bytes in the packet read from the network. Using an unsigned
          value here made it impossible to handle an error returned from
          recvfrom(). Furthermore, in the case that recvfrom() did return
          an error, this would cause a crash due to a heap overflow.
          (closes issue #10265, reported by and fix suggested by
          timrobbins)

2007-07-21 02:02 +0000 [r76227]  Russell Bryant <russell@digium.com>

        * /, channels/chan_sip.c: Merged revisions 76226 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r76226 | russell | 2007-07-20 21:01:46 -0500 (Fri, 20 Jul 2007) |
          4 lines Backport a fix for a memory leak that was fixed in trunk
          in reivision 76221 by rizzo. The memory used for the localaddr
          list was not freed during a configuration reload. ........

2007-07-20 21:36 +0000 [r76211]  Steve Murphy <murf@digium.com>

        * sounds/Makefile: This patch from 10249 is worth applying! It
          prevents downloading sound files if they are already downloaded.
          Darn Practical, if you ask me

2007-07-20 21:03 +0000 [r76174-76178]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Allow getting a call from an existing
          "sub" channel. Cancel ringing if endpoint hangs up before
          answering. Fixes were backported from trunk (there was apparently
          a bit of confusion during merge of a previous patch). (closes
          issue #10241)

        * main/manager.c: Eliminate a compiler warning with gcc 4.2 by
          constifying a char *

        * channels/chan_skinny.c: It's possible for sub->owner to be NULL
          here if you cancel the call immediately after/during sending a
          digit.

2007-07-20 18:42 +0000 [r76139]  Mark Michelson <mmichelson@digium.com>

        * apps/app_directory.c: When using users.conf for the entries in
          the directory, if multiple users had the same last name, only the
          first user listed would be available in the directory. (closes
          issue #10200, reported by mrskippy, patched by me)

2007-07-20 18:22 +0000 [r76132]  Russell Bryant <russell@digium.com>

        * main/channel.c: Use the define that specifies the default length
          of an artificially created DTMF digit in the ast_senddigit()
          function. The define is set to 100ms by default, which is the
          same thing that this function was using. But, using the define
          lets changes take effect in this case, as well as the others
          where it was already used.

2007-07-20 17:20 +0000 [r76054-76087]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_sip.c: Merged revisions 76080 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r76080 | file | 2007-07-20 14:16:48 -0300 (Fri, 20 Jul 2007) | 6
          lines (closes issue #10247) Reported by: fkasumovic Patches:
          chan_sip.patch uploaded by fkasumovic (license #101) Drop any
          peer realm authentication entries when reloading so multiple
          entries do not get added to the peer. ........

        * res/res_convert.c: (closes issue #10246) Reported by: fkasumovic
          Patches: res_conver.patch uploaded by fkasumovic (license #101)
          Use the last occurance of . to find the extension, not the first
          occurance.

        * apps/app_queue.c: Move makeannouncement variable declaration to
          proper place.

2007-07-19 20:36 +0000 [r75980]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Remove some duplicate code.

2007-07-19 18:59 +0000 [r75969-75978]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: The diff on this looks pretty big but all I did
          was remove a pointless if statement (always evaluates true).

        * apps/app_queue.c: Changes in handling return values of several
          functions in app_queue. This all started as a fix for issue
          #10008 but now includes all of the following changes: 1.
          Simplifying the code to handle positive return values from ast
          API calls. 2. Removing the background_file function. 3. The fix
          for issue #10008 (closes issue #10008, reported and patched by
          dimas)

2007-07-19 15:53 +0000 [r75928]  Russell Bryant <russell@digium.com>

        * /, channels/chan_iax2.c: Merged revisions 75927 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r75927 | russell | 2007-07-19 10:49:42 -0500 (Thu, 19 Jul 2007) |
          6 lines When processing full frames, take sequence number
          wraparound into account when deciding whether or not we need to
          request retransmissions by sending a VNAK. This code could cause
          VNAKs to be sent erroneously in some cases, and to not be sent in
          other cases when it should have been. (closes issue #10237,
          reported and patched by mihai) ........

2007-07-18 22:59 +0000 [r75807]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Need to make sure we set milliseconds and
          timestamp - pointed out by the recent ast_ time stuff from
          Tilghman

2007-07-18 21:09 +0000 [r75759]  Russell Bryant <russell@digium.com>

        * /, channels/chan_iax2.c: Merged revisions 75757 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r75757 | russell | 2007-07-18 16:09:13 -0500 (Wed, 18 Jul 2007) |
          5 lines When traversing the queue of frames for possible
          retransmission after receiving a VNAK, handle sequence number
          wraparound so that all frames that should be retransmitted
          actually do get retransmitted. (issue #10227, reported and
          patched by mihai) ........

2007-07-18 20:40 +0000 [r75749]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_voicemail.c, /: Merged revisions 75748 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r75748 | tilghman | 2007-07-18 15:31:36 -0500 (Wed, 18 Jul 2007)
          | 2 lines Store prior to copy (closes issue #10193) ........

2007-07-18 20:17 +0000 [r75732]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Umm, why are we transmitting dialtone on
          cfwdall?

2007-07-18 20:00 +0000 [r75712]  Joshua Colp <jcolp@digium.com>

        * apps/app_voicemail.c, channels/chan_sip.c, channels/chan_agent.c,
          pbx/pbx_realtime.c: Backport GCC 4.2 fixes. Without these
          Asterisk won't build under devmode using GCC 4.2.

2007-07-18 19:54 +0000 [r75707-75711]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Fixes for 7935/7936 conference phones.
          Issue 9245, patch by slimey.

        * channels/chan_skinny.c: Fix issues with new 79x1 phones. Issue
          9887, patches by DEA

2007-07-18 17:56 +0000 [r75658]  Dwayne M. Hubbard <dhubbard@digium.com>

        * /, apps/app_queue.c: Merged revisions 75657 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r75657 | dhubbard | 2007-07-18 12:48:33 -0500 (Wed, 18 Jul 2007)
          | 1 line removed the word 'pissed' from ast_log(...) function
          call for BE-90 ........

2007-07-18 15:44 +0000 [r75583-75623]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Few more places that needs to check for
          onhold state.

        * channels/chan_sip.c: (closes issue #10165) Reported by: elandivar
          It is possible for hold status to exist without call limits set,
          so we need to ensure update_call_counter is executed regardless.

        * channels/chan_h323.c: Don't bother reloading chan_h323 if it did
          not load successfully in the first place. This would otherwise
          cause a crash.

        * pbx/pbx_dundi.c: (closes issue #10224) Reported by: irroot Record
          the threadid of each running thread before shutting them down as
          the thread themselves may change the value.

2007-07-18 12:29 +0000 [r75529]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_meetme.c: Using a freed frame causes crashes (closes
          issue #9317)

2007-07-17  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.8 released.

2007-07-17 20:57 +0000 [r75441-75450]  Russell Bryant <russell@digium.com>

        * /, channels/chan_skinny.c: Merged revisions 75449 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r75449 | russell | 2007-07-17 15:57:09 -0500 (Tue, 17
          Jul 2007) | 3 lines Properly check for the length in the skinny
          packet to prevent an invalid memcpy. (ASA-2007-016) ........

        * main/rtp.c: cast arguments to ast_log so that it builds without
          warnings for me

        * channels/iax2-parser.c, channels/iax2-parser.h, /,
          channels/chan_iax2.c: Merged revisions 75444 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r75444 | russell | 2007-07-17 15:45:27 -0500 (Tue, 17 Jul 2007) |
          5 lines Ensure that when encoding the contents of an ast_frame
          into an iax_frame, that the size of the destination buffer is
          known in the iax_frame so that code won't write past the end of
          the allocated buffer when sending outgoing frames. (ASA-2007-014)
          ........

        * /, channels/chan_iax2.c: Merged revisions 75440 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r75440 | russell | 2007-07-17 15:41:41 -0500 (Tue, 17 Jul 2007) |
          4 lines After parsing information elements in IAX frames, set the
          data length to zero, so that code later on does not think it has
          data to copy. (ASA-2007-015) ........

2007-07-17 20:40 +0000 [r75439]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: Ensure that the pointer to STUN data does not go to
          unaccessible memory. (ASA-2007-017)

2007-07-17 20:33 +0000 [r75437]  Russell Bryant <russell@digium.com>

        * res/res_agi.c: (issue #10210) Reported by: juggie Patches:
          10210-1.4-grr.patch uploaded by juggie (license #24) Tested by:
          juggie, blitzrage Log a warning if someone uses DeadAGI on a live
          channel.

2007-07-17 20:03 +0000 [r75405]  Mark Michelson <mmichelson@digium.com>

        * apps/app_dial.c: Fixing an error I made earlier. ast_fileexists
          can return -1 on failure, so I need to be sure that we only enter
          the if statement if it is successful. Related to my fix to issue
          #10186

2007-07-17 20:01 +0000 [r75401-75403]  Russell Bryant <russell@digium.com>

        * main/pbx.c: (closes issue #10209) Reported by: juggie Patches:
          10209-trunk-2.patch uploaded by juggie Tested by: juggie,
          blitzrage In ast_pbx_run(), mark a channel as hung up after an
          application returned -1, or when it runs out of extensions to
          execute. This is so that code can detect that this channel has
          been hung up for things like making sure DeadAGI is used on
          actual dead channels, and is beneficial for other things, like
          making sure someone doesn't try to start spying on a channel that
          is about to go away.

        * res/res_agi.c: Remove a duplicated newline character in AGI debug
          output. (closes issue #10207, patch by seanbright)

2007-07-16 20:53 +0000 [r75258-75306]  Kevin P. Fleming <kpfleming@digium.com>

        * main/dns.c, /: Merged revisions 75304 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r75304 | kpfleming | 2007-07-16 15:46:58 -0500 (Mon, 16 Jul 2007)
          | 3 lines provide proper copyright/license attribution for this
          structure that was copied from a BSD-licensed header file long,
          long ago... ........

        * /: another fix that is not needed here (finishing up 75251)

2007-07-16 18:16 +0000 [r75253]  Mark Michelson <mmichelson@digium.com>

        * apps/app_dial.c: Restoring functionality from 1.2 wherein
          Retrydial will not exit if there is no announce file specified.
          This change makes it so that if there is no announce file
          specified, the application will continue until finished (or
          caller hangs up). If a bogus announce file is specified, then a
          warning message will be printed saying that the file could not be
          found, but execution will still continue. (closes issue #10186,
          reported by jon, patched by me)

2007-07-16 18:12 +0000 [r75252]  Kevin P. Fleming <kpfleming@digium.com>

        * /: block change that is not relevant here

2007-07-13 20:36 +0000 [r75108]  Russell Bryant <russell@digium.com>

        * /, res/res_musiconhold.c: Merged revisions 75107 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r75107 | russell | 2007-07-13 15:35:22 -0500 (Fri, 13
          Jul 2007) | 3 lines Fix a couple potential minor memory leaks.
          load_moh_classes() could return without destroying the loaded
          configuration. ........

2007-07-13 20:15 +0000 [r75078]  Mark Michelson <mmichelson@digium.com>

        * apps/app_chanspy.c, /: Merged revisions 75066 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r75066 | mmichelson | 2007-07-13 15:10:39 -0500 (Fri, 13 Jul
          2007) | 5 lines Fixed an issue where chanspy flags were
          uninitialized if no options were passed. What triggered this
          investigation was an IRC chat where some people's quiet flags
          were set while others' weren't even though none of them had
          specified the q option. ........

2007-07-13 20:10 +0000 [r75053-75067]  Russell Bryant <russell@digium.com>

        * /, res/res_musiconhold.c: Merged revisions 75059 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r75059 | russell | 2007-07-13 15:07:21 -0500 (Fri, 13
          Jul 2007) | 6 lines Ensure that adding a user to the list of
          users of a specific music on hold class is not done at the same
          time as any of the other operations on this list to prevent list
          corruption. Using the global moh_data lock for this is not ideal,
          but it is what is used to protect these lists everywhere else in
          the module, and I am only changing what is necessary to fix the
          bug. ........

        * channels/chan_zap.c, /: Merged revisions 75052 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r75052 | russell | 2007-07-13 14:10:00 -0500 (Fri, 13 Jul 2007) |
          12 lines (closes issue #9660) Reported by: mmacvicar Patches
          submitted by: bbryant, russell Tested by: mmacvicar, marco,
          arcivanov, jmhunter, explidous When using a TDM400P (and probably
          other analog cards) there was a chance that you could hang up and
          pick the phone back up where it has been long enough to be not
          considered a flash hook, but too soon such that the device
          reports that it is busy and the person on the phone will only
          hear silence. This patch makes chan_zap more tolerant of this and
          gives the device a couple of seconds to succeed so the person on
          the phone happily gets their dialtone. ........

2007-07-12 23:00 +0000 [r74998]  Mark Michelson <mmichelson@digium.com>

        * channels/chan_agent.c: Change to my previous fix regarding agent
          logoff soft. Now uses deferlogoff instead of loginstart since
          loginstart is used after logoff. Thanks to makoto for pointing
          this out and suggesting the fix. (closes issue #10178, reported
          and patched by makoto, with modification by me)

2007-07-12 20:42 +0000 [r74955]  Steve Murphy <murf@digium.com>

        * channels/chan_sip.c: This patch resolves 10143; thanks to irroot
          for the patch; looked acceptable. Let the community decide if it
          messes things up

2007-07-12 19:17 +0000 [r74888-74922]  Joshua Colp <jcolp@digium.com>

        * main/channel.c: Whoops... didn't want this to be returned to 0
          each iteration.

        * main/channel.c: When waiting for a digit ensure that a begin
          frame was received with it, not just an end frame. (issue #10084
          reported by rushowr)

2007-07-12 16:53 +0000 [r74839-74866]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: It helps if I actually add this stuff for
          the 7921 too - otherwise it won't actually do much of anything.

        * channels/chan_skinny.c: Add device ID for 7921 wireless skinny
          phone

        * channels/chan_skinny.c: Fix dialing in skinny that was broken in
          some cases. Issue 10136, fix provided by DEA.

2007-07-12 15:53 +0000 [r74815]  Joshua Colp <jcolp@digium.com>

        * /, res/res_musiconhold.c: Merged revisions 74814 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r74814 | file | 2007-07-12 12:51:24 -0300 (Thu, 12 Jul
          2007) | 2 lines Only print out a warning for situations where it
          is actually helpful. (issue #10187 reported by denke) ........

2007-07-11 22:57 +0000 [r74767]  Russell Bryant <russell@digium.com>

        * /, channels/chan_iax2.c: Merged revisions 74766 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r74766 | russell | 2007-07-11 17:53:26 -0500 (Wed, 11 Jul 2007) |
          5 lines The function make_trunk() can fail and return -1 instead
          of a valid new call number. Fix the uses of this function to
          handle this instead of treating it as the new call number. This
          would cause a deadlock and memory corruption. (possible cause of
          issue #9614 and others, patch by me) ........

2007-07-11 21:14 +0000 [r74722]  Mark Michelson <mmichelson@digium.com>

        * /, channels/chan_agent.c: Merged revisions 74719 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r74719 | mmichelson | 2007-07-11 16:12:30 -0500 (Wed, 11
          Jul 2007) | 5 lines The cli command "agent logoff Agent/x soft"
          did not work...at all. Now it does. (closes issue #10178,
          reported and patched by makoto, with slight modification for 1.4
          and trunk by me) ........

2007-07-11 18:34 +0000 [r74657]  Russell Bryant <russell@digium.com>

        * res/res_config_odbc.c, /: Merged revisions 74656 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r74656 | russell | 2007-07-11 13:33:23 -0500 (Wed, 11
          Jul 2007) | 4 lines Make sure that the ESCAPE immediately follows
          the condition that uses LIKE. This fixes realtime extensions with
          ODBC. (closes issue #10175, reported by stuarth, patch by me)
          ........

2007-07-11 18:18 +0000 [r74628-74642]  Steve Murphy <murf@digium.com>

        * Makefile: This fixes 10172, where the entire man8 dir gets
          removed during an uninstall of asterisk

        * utils/expr2.testinput, doc/channelvariables.txt, UPGRADE.txt:
          further reversion of previously applied floating point stuff for
          expr2

2007-07-11 17:16 +0000 [r74515-74590]  Joshua Colp <jcolp@digium.com>

        * channels/chan_phone.c, configure,
          include/asterisk/autoconfig.h.in, configure.ac: Instead of
          figuring out kernel versions that have compiler.h and not...
          let's just use autoconf to check for it's presence. (issue #10174
          reported by francesco_r)

        * channels/chan_phone.c: Only check if we need to do a SIGMA based
          tone generation if we have a card. (issue #10179 reported by
          mikowhy)

2007-07-10 23:32 +0000 [r74476]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Forwarding a message with IMAP storage was
          storing the message in the sender's box instead of the forwarded
          mailbox. (closes issue #10138, reported and patched by jaroth)

2007-07-10 19:58 +0000 [r74374-74428]  Jason Parker <jparker@digium.com>

        * /, apps/app_queue.c: Merged revisions 74427 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r74427 | qwell | 2007-07-10 14:57:20 -0500 (Tue, 10 Jul 2007) | 6
          lines Fix an issue where it was possible to have a service level
          of over 100% Between the time recalc_holdtime and update_queue
          was called, it was possible that the call could have been hungup.
          Move both additions to the same place, so this won't happen.
          Issue 10158, initial patch by makoto, modified by me. ........

        * main/dns.c: Don't use #if to check if something is defined - use
          #ifdef instead. Pointed out by kpfleming

        * /, channels/chan_agent.c: Merged revisions 74376 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r74376 | qwell | 2007-07-10 14:03:45 -0500 (Tue, 10 Jul
          2007) | 4 lines Fix an issue with wrapuptime not working when
          using AgentLogin. Issue 10169, patch by makoto, with a minor mod
          by me to not re-break issue 9618 ........

        * main/dns.c, /, configure, include/asterisk/autoconfig.h.in,
          configure.ac: Merged revisions 74373 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5
          lines Use res_ndestroy on systems that have it. Otherwise, use
          res_nclose. This prevents a memleak on NetBSD - and possibly
          others. Issue 10133, patch by me, reported and tested by scw
          ........

2007-07-10  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.7.1 released.

2007-07-10 16:00 +0000 [r74323]  Russell Bryant <russell@digium.com>

        * res/res_musiconhold.c: fix an uninitialized variable

2007-07-10 15:38 +0000 [r74317]  Jason Parker <jparker@digium.com>

        * apps/app_voicemail.c, /: Merged revisions 74316 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r74316 | qwell | 2007-07-10 10:37:54 -0500 (Tue, 10 Jul 2007) | 4
          lines Fix a small typo in description in of Voicemail()
          application. Issue 10170, patch by casper. ........

2007-07-10 15:31 +0000 [r74314]  Russell Bryant <russell@digium.com>

        * res/res_config_odbc.c, /: Merged revisions 74313 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r74313 | russell | 2007-07-10 10:30:20 -0500 (Tue, 10
          Jul 2007) | 3 lines Only use ESCAPE when LIKE is used. (issue
          #10075, this part reported by jmls on IRC, patch by me) ........

2007-07-10 14:50 +0000 [r74262-74265]  Joshua Colp <jcolp@digium.com>

        * /, main/app.c: Merged revisions 74264 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r74264 | file | 2007-07-10 11:48:00 -0300 (Tue, 10 Jul 2007) | 2
          lines Ensure the group information category exists before trying
          to do a string comparison with it. (issue #10171 reported by
          mlegas) ........

        * channels/chan_sip.c: Only spit out an inringing warning message
          when it is applicable. Since call limits are already toast in
          realtime let's not scare the user if they are using it. (issue
          #10166 reported by bcnit)

2007-07-09  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.7 released.

2007-07-09 21:31 +0000 [r74162-74211]  Russell Bryant <russell@digium.com>

        * configure, configure.ac: Update the configure script to check for
          a required function that is not present in the 1.2 version of
          libpri. This will prevent the configure script from thinking that
          it has compatible libpri support for Asterisk 1.4, when it
          actually does not because the installed version is from 1.2.

        * res/res_musiconhold.c: (closes issue #10123) Reported by:
          blitzrage Patches submitted by: juggie, qwell, me Tested by:
          blitzrage When trying to find a music on hold class to use, try
          all of the options, instead of only the first one that is set.
          Also, change the MusicOnHold applications to not hang up on the
          channel when a class can not be found.

2007-07-09 20:19 +0000 [r74159]  Jason Parker <jparker@digium.com>

        * channels/chan_zap.c, /: Merged revisions 74158 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul 2007) | 8
          lines Several chan_zap options were not working on reload because
          they were arbitrarily disallowed when reloading some/most PRI
          options (such as signalling) was disallowed. Options such as
          polarityonanswerdelay and answeronpolarityswitch can safely be
          changed on a reload. This corrects that behavior. Issue 9186,
          patch by tzafrir. ........

2007-07-09 18:38 +0000 [r74120-74122]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Forgot to get rid of an extraneous debug
          message.

        * apps/app_queue.c: The n option for Queue should make the queue
          exit immediately after failure to reach any members and should
          not be dependent on the timeout value passed to Queue (closes
          issue #10127, reported by bcnit, repaired by me)

2007-07-09 15:32 +0000 [r74082]  Joshua Colp <jcolp@digium.com>

        * channels/chan_skinny.c: Only destroy the scheduler context if it
          was allocated. (issue #10124 reported by gzero)

2007-07-09 14:57 +0000 [r74047]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Fixed a logic error in leave_voicemail.
          Pass the mailbox instead of the context to inbox_count when the
          context is "default." (closes issue #10135, reported by yannj,
          repaired by me)

2007-07-09 14:49 +0000 [r74043-74045]  Joshua Colp <jcolp@digium.com>

        * channels/chan_skinny.c, pbx/pbx_dundi.c: Few minor thread
          synchronization tweaks. (issue #10124 reported by gzero)

        * configure, acinclude.m4: Use AC_CHECK_HEADER to check for
          ptlib/openh323 to allow for cross compiling. (issue #9675
          reported by zandbelt)

2007-07-09 04:03 +0000 [r73985]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * main/ast_expr2f.c: Doxygen formatting fixes; fixes errors while
          'make progdocs'. (Closes issue #10104)

2007-07-09 03:13 +0000 [r73930-73980]  Joshua Colp <jcolp@digium.com>

        * main/cdr.c: Give Agent channel names priority when doing CDR
          merging. (issue #10011 reported by krtorio)

        * pbx/pbx_config.c: Add a few sanity checks when writing out the
          dialplan. (issue #10157 reported by dome)

2007-07-08 09:47 +0000 [r73849]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: While tracking down a bug, I need some more
          history. Dumphistory is very useful, indeed.

2007-07-06 23:02 +0000 [r73769]  Russell Bryant <russell@digium.com>

        * /, channels/chan_sip.c: Merged revisions 73768 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 Jul 2007) |
          4 lines If a sip_pvt struct has already registered an extension
          state callback, remove the old one before adding a new one. If
          this isn't done, Asterisk will crash. (issue #10120) ........

2007-07-06 16:36 +0000 [r73727]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Fixing a rare case which causes voicemail
          to crash when compiled with IMAP storage. inboxcount has the
          possibility of finding an "interactive" vm_state when no
          persistent "non-interactive" vm_state exists for that mailbox. If
          this should happen when someone attempts to leave a message, it
          results in a crash. This patch, along with my commit in revision
          72670 fix issue 10053, reported by jaroth. closes issue #10053

2007-07-06 16:12 +0000 [r73679-73696]  Russell Bryant <russell@digium.com>

        * res/res_config_odbc.c, /: Merged revisions 73684 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r73684 | russell | 2007-07-06 11:06:27 -0500 (Fri, 06
          Jul 2007) | 8 lines (closes issue #10075) Reported by: apsaras
          Patches submitted by: Corydon76 Tested by: apsaras Fix a problem
          with MSSQL 2005 by explicitly stating that '\' is being used as
          an escape character. ........

        * /, channels/chan_sip.c: Merged revisions 73678 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) |
          7 lines (closes issue #10125) Reported by: makoto Patches
          submitted by: makoto This fixes a crash in chan_sip that happens
          when the bindaddr setting is not valid on Asterisk startup, gets
          fixed, and then a reload gets issued. ........

2007-07-06 15:27 +0000 [r73675]  Mark Michelson <mmichelson@digium.com>

        * /, channels/chan_agent.c: Merged revisions 73674 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r73674 | mmichelson | 2007-07-06 10:26:40 -0500 (Fri, 06
          Jul 2007) | 5 lines Fixed a bug wherein agents get stuck busy.
          (issue 9618, reported by jiddings, patched by moi) closes issue
          #9618 ........

2007-07-06 03:34 +0000 [r73551-73629]  Russell Bryant <russell@digium.com>

        * BUGS: fix a little spelling error

        * channels/chan_sip.c: Fix a crash in chan_sip. Don't try to stop
          the monitor thread if it was never started. (closes issue #10124,
          reported by gzero, fixed by me)

        * channels/chan_iax2.c: copy from the correct buffer when deferring
          a full frame (related to issue #9937)

        * channels/chan_iax2.c: * Store the call number that a thread is
          processing without the full frame bit set to ease debugging *
          When deferring a full frame for processing, stick it into the
          queue for the thread that is processing frames for that call, not
          the one that read the current frame and is about to go back into
          the idle list (related to issue #9937)

2007-07-05 22:20 +0000 [r73548]  Kevin P. Fleming <kpfleming@digium.com>

        * /, channels/chan_sip.c: Merged revisions 73547 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05 Jul 2007)
          | 2 lines we shouldn't allow G.723.1 endpoints to use VAD, just
          like we don't support it for G.729 ........

2007-07-05 20:50 +0000 [r73512]  Russell Bryant <russell@digium.com>

        * res/res_features.c: Pass HOLD and UNHOLD frames to the other
          channel when they are returned from a native bridge function.
          This fixes a problem where when two zap channels are natively
          bridged and one does a flash hook, the other channel did not
          receive music on hold. (Reported to me directly by Doug Bailey at
          Digium)

2007-07-05 19:18 +0000 [r73467]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_sip.c: Merged revisions 73466 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2
          lines Copy language information to the dialog structure when
          calling a peer for situations where a PBX may be started on the
          dialed channel. (issue #10121 reported by clegall_proformatique)
          ........

2007-07-05 15:59 +0000 [r73400]  Mark Michelson <mmichelson@digium.com>

        * apps/app_queue.c: Correcting a minor CLI bug I found. When
          issuing the queue show command, if you type queue show and then
          press tab, you can continue pressing tab and it will keep
          auto-completing queue names even though only 1 queue can be used
          as an argument.

2007-07-05 15:28 +0000 [r73398]  Russell Bryant <russell@digium.com>

        * channels/chan_vpb.cc, channels/Makefile: Make this module build
          for me in dev-mode

2007-07-05 14:21 +0000 [r73316-73355]  Joshua Colp <jcolp@digium.com>

        * apps/app_chanspy.c, main/channel.c, /: Merged revisions 73349 via
          svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r73349 | file | 2007-07-05 11:19:14 -0300 (Thu, 05 Jul 2007) | 2
          lines Tweak spy locking. (issue #9951 reported by welles)
          ........

        * channels/chan_local.c, /: Merged revisions 73318 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul
          2007) | 2 lines Actually check to make sure a PBX was started on
          one of the Local channels instead of blindly assuming it was.
          (issue #10112 reported by makoto) ........

        * /, apps/app_queue.c: Merged revisions 73315 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r73315 | file | 2007-07-05 10:19:17 -0300 (Thu, 05 Jul 2007) | 2
          lines Reset ServicelevelPerf variable back to 0 if we are unable
          to calculate it each time... otherwise we will get previous
          values. (issue #10117 reported by noriyuki) ........

2007-07-04 14:53 +0000 [r73208-73253]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, /: Merged revisions 73252 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r73252 | crichter | 2007-07-04 16:50:58 +0200 (Mi, 04
          Jul 2007) | 1 line bchannel configurations like echocancel and
          volume control, need to be setuped on inbound calls too. ........

        * channels/chan_misdn.c, /: Merged revisions 73207 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r73207 | crichter | 2007-07-04 10:20:54 +0200 (Mi, 04
          Jul 2007) | 1 line bad bug in overlapdial case, we called
          start_pbx multiple times, because the state wasn't changed..
          ........

2007-07-03 20:17 +0000 [r73143]  Steve Murphy <murf@digium.com>

        * main/ast_expr2.fl, main/ast_expr2.c, main/Makefile,
          main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c: Removing
          expr floating patch from 1.4; too much of a behavior change. If
          you want this fix, try trunk instead. bug 9508.

2007-07-03 15:42 +0000 [r73104-73106]  Jason Parker <jparker@digium.com>

        * /: What the heck. This should not have happened.

        * /: use autotagged externals

2007-07-03 12:38 +0000 [r73053]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_dial.c, /: Merged revisions 73052 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r73052 | tilghman | 2007-07-03 07:34:14 -0500 (Tue, 03 Jul 2007)
          | 2 lines RetryDial should accept a 0 argument, but it does not,
          because atoi does not distinguish between 0 and error (closes
          issue #10106) ........

2007-07-03 08:17 +0000 [r73005]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c, /: Merged revisions 73004 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r73004 | crichter | 2007-07-03 10:04:35 +0200 (Di, 03
          Jul 2007) | 1 line fixed issue, that misdn_l2l1_check could only
          be called from mISDN Source channels.. #9449 ........

2007-07-02 20:16 +0000 [r72933]  Steve Murphy <murf@digium.com>

        * main/ast_expr2.fl, main/ast_expr2.c, utils/expr2.testinput,
          main/Makefile, main/ast_expr2.h, main/ast_expr2.y,
          main/ast_expr2f.c, doc/channelvariables.txt, UPGRADE.txt: support
          for floating point numbers added to ast_expr2 $\[...\] exprs.
          Fixes bug 9508, where the expr code fails with fp numbers. The
          MATH function returns fp numbers by default, so this fix is
          considered necessary.

2007-07-02 18:18 +0000 [r72926]  Russell Bryant <russell@digium.com>

        * main/manager.c: Remove a bogus comment and add proper locking to
          the handler function for the CLI command to show information on
          manager actions.

2007-07-02 14:32 +0000 [r72888]  Joshua Colp <jcolp@digium.com>

        * main/channel.c: Added additional DTMF debug messages for when
          emulation occurs.

2007-07-02 08:41 +0000 [r72850-72852]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
          revisions 72585 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r72585 | crichter | 2007-06-29 15:08:26 +0200 (Fr, 29 Jun 2007) |
          1 line check if the bchannel stack id is already used, if so
          don't use it a second time. Also added a release_chan lock, so
          that the same chan_list object cannot be freed twice. chan_misdn
          does not crash anymore on heavy load with these changes. ........

        * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
          channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
          Merged revisions 72099 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r72099 | crichter | 2007-06-27 15:22:37 +0200 (Mi, 27 Jun 2007) |
          1 line simplified generation for dummy bchannels, also we mark
          them as dummies, so they are not used later as real-bchannels,
          optimized the RESTART mechanisms, we block a channel now on
          cause:44, and send out a RESTART automatically, then on reception
          of RESTART_ACKNOWLEDGE we unblock the channel again. ........

        * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged
          revisions 72087 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r72087 | crichter | 2007-06-27 11:26:53 +0200 (Mi, 27 Jun 2007) |
          1 line simplified channel finding and locking a lot. removed
          unnecessary #ifdefed areas. ........

2007-07-01 23:52 +0000 [r72806]  Russell Bryant <russell@digium.com>

        * pbx/pbx_spool.c, /: Merged revisions 72805 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r72805 | russell | 2007-07-01 18:51:34 -0500 (Sun, 01 Jul 2007) |
          5 lines When appending lines to call files to keep track of
          retries, write a leading newline just in case the original call
          file did not have a newline at the end. This fix is in response
          to a problem I saw reported on the asterisk-users mailing list.
          ........

2007-06-30 16:50 +0000 [r72705-72766]  Russell Bryant <russell@digium.com>

        * configure, configure.ac: Tweak the configure script so that error
          output isn't spewed to the console when searching for GTK2 libs,
          and they aren't found.

        * formats/format_pcm.c: give format_pcm a more concise destription

2007-06-29 19:07 +0000 [r72665]  Luigi Rizzo <rizzo@icir.org>

        * main/utils.c: Use !defined(HAVE_GETHOSTBYNAME_R) to check for
          absence of the function. This was already done in trunk.

2007-06-29  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.6 released.

2007-06-29 14:26 +0000 [r72597-72599]  Joshua Colp <jcolp@digium.com>

        * main/cdr.c: Minor change for older GCC versions.

        * Makefile, configure, configure.ac, makeopts.in: Backport fix for
          GCC versions without support for declaration-after-statement.

2007-06-29 04:47 +0000 [r72554-72556]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * main/manager.c: Issue 10055 - Change memory allocation to use the
          heap for a command, since the output has the potential to
          overflow the stack (as it did here)

        * res/res_jabber.c: Fix 1.4 breakage

2007-06-28 19:44 +0000 [r72493]  Russell Bryant <russell@digium.com>

        * configure, include/asterisk/autoconfig.h.in: regenerate the
          configure script for rizzo

2007-06-28 19:29 +0000 [r72453-72489]  Luigi Rizzo <rizzo@icir.org>

        * configure.ac: add a check for gethostbyname_r so we can simplify
          the handling e.g. in utils.c Also add comments on a couple of
          features which are not working on FreeBSD. All the above has been
          already done in trunk so the merge must be blocked. Can someone
          please regenerate ./configure ?

        * Makefile, channels/chan_zap.c, main/say.c: Add
          -Wdeclaration-after-statement to AST_DEVMODE flags to catch
          variable declarations in the middle of a block. Fix the few
          instances of the above spotted out by the compiler. All of this
          has been already done or is not applicable in trunk, so the merge
          of this change will be blocked.

        * apps/app_meetme.c: cast a time_t so that it does not conflict
          with the print format. This change was already done on trunk so
          this change needs to be blocked from merging.

2007-06-27 23:29 +0000 [r72383]  Brett Bryant <bbryant@digium.com>

        * main/asterisk.c, /: Merged revisions 72373 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r72373 | bbryant | 2007-06-27 18:22:13 -0500 (Wed, 27 Jun 2007) |
          3 lines Reinstating patch. This actually fixes the problem,
          however I was running a development branch without it and
          mistakenly thought it wasn't fixed. Fixes issue #10010, and
          #9654: 100% CPU usage caused by an asterisk console losing it's
          controlling terminal. ........

2007-06-27 23:25 +0000 [r72381]  Joshua Colp <jcolp@digium.com>

        * apps/app_mixmonitor.c, /: Merged revisions 72378 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun
          2007) | 2 lines Update documentation to clarify variable usage
          with MixMonitor. (issue #9494 reported by netoguy) ........

2007-06-27 23:03 +0000 [r72335]  Brett Bryant <bbryant@digium.com>

        * main/asterisk.c, /: Merged revisions 72333 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r72333 | bbryant | 2007-06-27 17:58:53 -0500 (Wed, 27 Jun 2007) |
          2 lines Reverted changes for earlier revisions 72259 to 72261.
          Issue #9654, #10010 ........

2007-06-27 22:58 +0000 [r72328-72331]  Joshua Colp <jcolp@digium.com>

        * channels/chan_gtalk.c: Make payload IDs for iLBC/Speex match to
          our list. Since these are dynamic payloads the other side
          shouldn't care. (issue #9426 reported by irroot)

        * /, apps/app_queue.c: Merged revisions 72327 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r72327 | file | 2007-06-27 18:43:11 -0400 (Wed, 27 Jun 2007) | 2
          lines Fix issue where queue log events might be missing. (issue
          #7765 reported by mtryfoss) ........

2007-06-27 21:08 +0000 [r72272]  Russell Bryant <russell@digium.com>

        * /, pbx/pbx_config.c: Merged revisions 72267 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r72267 | russell | 2007-06-27 16:06:45 -0500 (Wed, 27 Jun 2007) |
          5 lines Fix a minor issue with parsing the priority number. You
          could have as much whitespace as you want around a numeric
          priority, but you couldn't have any whitespace around a special
          priority like "n" or "hint". (issue #10039, reported by mitheloc,
          fixed by me) ........

2007-06-27 20:46 +0000 [r72260]  Brett Bryant <bbryant@digium.com>

        * main/asterisk.c, /: Merged revisions 72259 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r72259 | bbryant | 2007-06-27 15:43:53 -0500 (Wed, 27 Jun 2007) |
          4 lines Fixes 100% load when controlling terminal disappears.
          Issue #9654, #10010 ........

2007-06-27 20:25 +0000 [r72257]  Joshua Colp <jcolp@digium.com>

        * main/channel.c, /: Merged revisions 72256 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2
          lines I may possibly get shot for doing this... but... defer CDR
          processing until after the channel has been dealt with. This
          should eliminate all of the issues with channels going funky
          (SIP/PRI) when you are posting CDRs to a database that is either
          slow or unavailable and do not want to enable batching. ........

2007-06-27 19:13 +0000 [r72205]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_zap.c: use the proper type for storing group number
          bits so that if someone specifies 'group=42' it will actually
          work instead of being silently ignored

2007-06-27 18:40 +0000 [r72182-72185]  Jason Parker <jparker@digium.com>

        * apps/app_voicemail.c: Fix another problem in voicemail with
          missing symbols. Issue 10074, patch by kryptolus, extended to
          include #if 0'd blocks (just in case)

2007-06-27 17:31 +0000 [r72148]  Joshua Colp <jcolp@digium.com>

        * main/channel.c: Make the ast_read_noaudio API call behave better
          under circumstances where DTMF emulation was happening and a
          generator was setup. (issue #10065 reported by stevefeinstein)

2007-06-27 17:10 +0000 [r72125]  Jason Parker <jparker@digium.com>

        * channels/chan_gtalk.c: Don't modify a variable that we don't want
          modified. Make a copy of it instead. Issue 10029, patch by
          phsultan with slight modifications by me (to remove needless
          casts).

2007-06-27 16:34 +0000 [r72112]  Russell Bryant <russell@digium.com>

        * main/rtp.c: Only output debug information related to RTCP
          timestamps when RTCP debug is turned on (issue #10066, patch by
          me)

2007-06-27 07:58 +0000 [r72042]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, /: Merged revisions 72040-72041 via
          svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r72040 | crichter | 2007-06-27 09:49:27 +0200 (Mi, 27 Jun 2007) |
          1 line for inbound TE calls, we setup the bchannel when we get
          the CONNECT_ACKNOWLEDGE, to make sure mISDN has everything ready.
          removed some #if 0 areas which weren't used anymore. ........
          r72041 | crichter | 2007-06-27 09:54:30 +0200 (Mi, 27 Jun 2007) |
          1 line isdn_lib.c didn't compile ........

2007-06-27 00:58 +0000 [r72006]  Joshua Colp <jcolp@digium.com>

        * pbx/pbx_dundi.c: Make unloading of pbx_dundi actually work.

2007-06-26 23:02 +0000 [r71953]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Removing a pointless line. This variable
          was already set earlier and between then and this line, there is
          no way that the values on the right side of the assignment could
          have changed.

2007-06-26 20:36 +0000 [r71915]  Jason Parker <jparker@digium.com>

        * main/rtp.c: Don't dereference a pointer that may be NULL here.
          Issue 10017.

2007-06-26 19:00 +0000 [r71877]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: A few changes, the ultimate goal of which
          is to keep better track of the number of messages that a mailbox
          currently has. A description of the changes: 1. Changed the
          "updated" field of the vm_state struct to act more as a binary
          semaphore than a counting semaphore, since its current
          implementation made the inboxcount function not work properly.
          This change falls in line with a change made by UPenn with their
          IMAP setup and helps to sync our changes with theirs. 2.
          Eliminated some redundant calls to get_vm_state_by_mailbox inside
          leave_voicemail 3. Use the play_folder variable to keep track of
          the number of old and new messages in a mailbox as the messages
          are deleted 4. Added an increment to the number of new messages
          that was not there previously in the leave_voicemail function

2007-06-26 15:47 +0000 [r71796]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Fixing bug where the authuser was
          mistakenly pulled from the mailbox string instead of the IMAP
          user. (closes issue 10054, reported and patched by jaroth)

2007-06-26 12:27 +0000 [r71657-71751]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_voicemail.c, /: Merged revisions 71750 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r71750 | tilghman | 2007-06-26 07:25:58 -0500 (Tue, 26 Jun 2007)
          | 2 lines Issue 10062 - Trying to move a message without
          selecting one first results in memory corruption ........

        * /, res/res_agi.c: Merged revisions 71656 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r71656 | tilghman | 2007-06-25 13:12:37 -0500 (Mon, 25 Jun 2007)
          | 2 lines Issue 10035 - handle_exec returns a result inconsistent
          with all of the other AGI commands ........

2007-06-25 14:13 +0000 [r71522-71576]  Joshua Colp <jcolp@digium.com>

        * channels/chan_h323.c: Build a peer as well when hash323 is
          enabled in users.conf (issue #9599 reported by asagage)

        * channels/chan_agent.c: Minor tweak for queueing up the unhold
          frame... this will teach me to do bugs while half asleep. (issue
          #10046 reported by dimas)

2007-06-25 12:40 +0000 [r71519]  Russell Bryant <russell@digium.com>

        * doc/asterisk-mib.txt: Fix a typo in the Asterisk mib. (issue
          #10048, Matti)

2007-06-25 01:10 +0000 [r71412-71430]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_sip.c: Merged revisions 71414 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2
          lines Ignore other URIs after the first in a 300 Multiple Choice
          response. (issue #10041 reported by homesick) ........

        * main/cdr.c: Fix it so 1.4 actually compiles on my box.

        * channels/chan_agent.c: Check to make sure the channel pointer is
          present before queueing up an unhold frame on it. (issue #10046
          reported by dimas)

2007-06-24 20:16 +0000 [r71362-71371]  Russell Bryant <russell@digium.com>

        * build_tools/prep_tarball: Include the menuselect-tree file in
          tarballs to make builds from tarballs a little bit faster

        * main/asterisk.c, /: Merged revisions 71358 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r71358 | russell | 2007-06-24 15:04:21 -0500 (Sun, 24 Jun 2007) |
          2 lines Revert the patch from issue 9654 due to an unexpected
          side effect ........

2007-06-24 17:50 +0000 [r71289-71291]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * res/res_features.c: Issue 10044 - chan->cdr is NULL here, so
          peer->cdr is what we really wanted to use

        * main/db.c, main/manager.c, /: Merged revisions 71288 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r71288 | tilghman | 2007-06-24 12:32:21 -0500 (Sun, 24
          Jun 2007) | 2 lines Issue 10043 - There is a legitimate need to
          be able to set variables to the empty string. ........

2007-06-23 03:29 +0000 [r71230]  Steve Murphy <murf@digium.com>

        * main/cdr.c, res/res_features.c: This patch is meant to fix 8433;
          where clid and src are lost via bridging.

2007-06-22 22:44 +0000 [r71214]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
          channels/chan_misdn.c, /: Merged revisions 70341 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r70341 | crichter | 2007-06-20 17:29:09 +0200 (Mi, 20
          Jun 2007) | 1 line fixed a bug that was introduced by copy and
          paste in the last commit ..bchannels weren't cleaned properly.
          ........

2007-06-22 15:38 +0000 [r71096-71123]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
          revisions 70672 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r70672 | crichter | 2007-06-21 15:11:29 +0200 (Do, 21 Jun 2007) |
          1 line we activate the bchannels in TE mode on incoming calls
          only when we want to connect the call. ........

        * channels/misdn/isdn_lib.c, /: Merged revisions 70342 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r70342 | crichter | 2007-06-20 17:42:39 +0200 (Mi, 20
          Jun 2007) | 1 line forgot one place .. ........

        * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
          channels/chan_misdn.c, /: Merged revisions 70311 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r70311 | crichter | 2007-06-20 16:47:59 +0200 (Mi, 20
          Jun 2007) | 1 line on receiption of cause:44 we mark the channel
          as in use and inform the user about the situation, we need to
          test the RESTART stuff then. Also shuffled the
          empty_chan_in_stack function after the bchannel cleaning
          functions, to avoid race conditions. ........

        * channels/chan_misdn.c, /: Merged revisions 69887 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r69887 | crichter | 2007-06-19 15:23:04 +0200 (Di, 19
          Jun 2007) | 1 line when we send out a SETUP, but get no response,
          we should cleanup everything after reception of a hangup.
          ........

        * /, channels/misdn/isdn_msg_parser.c: Merged revisions 69053 via
          svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r69053 | crichter | 2007-06-13 11:55:54 +0200 (Mi, 13 Jun 2007) |
          1 line restart indicator 0x80 is correct, at least that's what
          libpri does. ........

        * channels/chan_misdn.c, /: Merged revisions 68887 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r68887 | crichter | 2007-06-12 10:35:22 +0200 (Di, 12
          Jun 2007) | 1 line if the bridged partner is mISDN too we should
          not send dtmf tones, they are transmitted inband always ........

        * channels/chan_misdn.c, /: Merged revisions 68874 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r68874 | crichter | 2007-06-12 09:48:52 +0200 (Di, 12
          Jun 2007) | 1 line if we have already some digits, we just stop
          the tones. ........

2007-06-22 15:00 +0000 [r71068]  Jason Parker <jparker@digium.com>

        * apps/app_speech_utils.c, /, res/res_agi.c, main/file.c: Merged
          revisions 71065 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4
          lines Fix a few silly usages of ast_playstream() - it only ever
          returns 0... Issue 10035 ........

2007-06-22 14:53 +0000 [r71066]  Brett Bryant <bbryant@digium.com>

        * main/asterisk.c, /: Merged revisions 71064 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r71064 | bbryant | 2007-06-22 09:39:34 -0500 (Fri, 22 Jun 2007) |
          10 lines Fixed infinite loop when controlling terminal was lost
          and return value of input function wasn't checked for errors.
          This would cause 100% cpu to be taken up. (closes issue #9654,
          issue #10010) Reported by: mnicholson, and eserra Idea for the
          patch from mnicholson, patched by me ........

2007-06-22 14:10 +0000 [r71063]  Steve Murphy <murf@digium.com>

        * main/cdr.c: My conditions for merging amaflags info was naive;
          DOCUMENTATION is the default, although null is possible; theft of
          user-settable fields is not good. Just copy them, leave them
          alone.

2007-06-22 03:14 +0000 [r71003]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix a small typo which ... well ...
          completely broke chan_iax2. oops! (issue #9937, patch by me)

2007-06-21 22:34 +0000 [r70949]  Steve Murphy <murf@digium.com>

        * main/cdr.c, /: Merged revisions 70948 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r70948 | murf | 2007-06-21 16:29:50 -0600 (Thu, 21 Jun 2007) | 1
          line This little fix is in response to bug 10016, but may not
          cure it. The code is wrong, clearly. In a situation where you set
          the CDR's amaflags, and then ForkCDR, and then set the new CDR's
          amaflags to some other value, you will see that all CDRs have had
          their amaflags changed. This is not good. So I fixed it. ........

2007-06-21 21:40 +0000 [r70899]  Joshua Colp <jcolp@digium.com>

        * apps/app_voicemail.c, /: Merged revisions 70898 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r70898 | file | 2007-06-21 17:37:55 -0400 (Thu, 21 Jun 2007) | 2
          lines Don't explode if the gain option is specified without a
          value. (issue #9274 reported by mfarver) ........

2007-06-21 21:14 +0000 [r70866-70883]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Put the thread reading from the socket back
          in the idle list if it deferred the processing of a full frame to
          another thread

        * channels/chan_iax2.c: If a full frame is received while one of
          the iax2 threads is in the middle of handling a full frame for
          the same call, queue it up for processing by that same thread
          later instead of dropping it. (issue #9937, patch by me)

2007-06-21 20:19 +0000 [r70841]  Steve Murphy <murf@digium.com>

        * cdr/cdr_custom.c, /: Merged revisions 70804 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r70804 | murf | 2007-06-21 13:13:17 -0600 (Thu, 21 Jun 2007) | 1
          line it was pointed out that the cdr_custom config load could get
          a lock, and under certain circumstances, would never release it.
          I also noted that the situation where more than one mapping spec
          was warned about, but did not ignore further mappings as it had
          promised. I think I have fixed both situations. ........

2007-06-21 19:49 +0000 [r70808]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: When volgain is used don't leave a
          temporary file behind. (Closes Issue 8514, Reported and patched
          by ulogic, code reviewed by Jason Parker)

2007-06-21 15:22 +0000 [r70727]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: Do not Packet2Packet bridge if packetization settings
          do not allow it. (issue #9117 reported by phsultan)

2007-06-21 15:21 +0000 [r70726]  Russell Bryant <russell@digium.com>

        * apps/app_meetme.c: Remove a couple of duplicate unlocks

2007-06-21 13:58 +0000 [r70677]  Joshua Colp <jcolp@digium.com>

        * apps/app_voicemail.c: Fix building with ODBC storage enabled.
          (issue #10025 reported by denisgalvao)

2007-06-21 13:00 +0000 [r70656]  Steve Murphy <murf@digium.com>

        * main/cdr.c: Via complaints aired in asterisk-users, I submit
          these changes, which allow cdr updates to see macro
          context/exten, whether hung up or not

2007-06-20 23:32 +0000 [r70554-70612]  Jason Parker <jparker@digium.com>

        * cdr/cdr_pgsql.c: Fix some potential memory leaks in cdr_pgsql.
          Issue 10020, patch by my, with credit to prashant_jois for
          pointing out the problem.

        * cdr/cdr_pgsql.c: Fix a stupid mistake in my last cdr_pgsql race
          condition fix

        * cdr/cdr_pgsql.c: Fix a race condition in cdr_pgsql that can occur
          when reloading the module. Issue 10022, patch by me, with credit
          to prashant_jois for finding the bug.

2007-06-20 22:22 +0000 [r70552]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_sip.c: Merged revisions 70551 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2
          lines Don't overwrite the configured username setting upon a
          REGISTER. (issue #8565 reported by jsmith) ........

2007-06-20 20:53 +0000 [r70494]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Make sure we clear the previously dialed
          number if it did not exist. Issue 9958.

2007-06-20 19:29 +0000 [r70445]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_dial.c, /: Merged revisions 70444 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r70444 | tilghman | 2007-06-20 14:25:54 -0500 (Wed, 20 Jun 2007)
          | 2 lines Issue 9997 - Timelimit times out the wrong channel
          ........

2007-06-20 18:46 +0000 [r70397]  Russell Bryant <russell@digium.com>

        * channels/chan_zap.c, /: Merged revisions 70396 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r70396 | russell | 2007-06-20 13:45:38 -0500 (Wed, 20 Jun 2007) |
          5 lines Fix a problem where an established call would not be
          properly disconnected when a PRI disconnect is received depending
          on which cause code was received. (issue #9588, original patch by
          softins, updated patch from jtexter3, and some additional
          feedback from mhardeman) ........

2007-06-20 17:52 +0000 [r70198-70360]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c, main/frame.c: Put the speex packetization values back
          in but disable it when setting up the smoother.

        * main/frame.c: Don't do packetization/smoother stuff with speex,
          it doesn't work.

2007-06-20 00:03 +0000 [r70084-70164]  Russell Bryant <russell@digium.com>

        * contrib/scripts/ast_grab_core: don't delete the backtrace in
          ast_grab_core

        * channels/chan_gtalk.c: Only attempt to queue a hangup on the
          owner channel if it actually exists. (issue #9795, patch from
          zandbelt)

2007-06-19 18:23 +0000 [r70062]  Steve Murphy <murf@digium.com>

        * main/channel.c, /: Merged revisions 70053 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1
          line This fixes 9246, where channel variables are not available
          in the 'h' exten, on a 'ZOMBIE' channel. The fix is to
          consolidate the channel variables during a masquerade, and then
          copy the merged variables back onto the clone, so the zombie has
          the same vars that the 'original' has. ........

2007-06-19 17:07 +0000 [r70003]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c, /: Merged revisions 69992 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2
          lines Handle the CC field in the RTP header. (issue #9384
          reported by DoodleHu) ........

2007-06-19 16:24 +0000 [r69987]  Joshua Colp <jcolp@digium.com>

        * main/channel.c, /: Merged revisions 69986 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r69986 | file | 2007-06-19 12:21:29 -0400 (Tue, 19 Jun 2007) | 2
          lines Update BRIDGEPEER variable if set to the new channel name
          when a masquerade happens. (issue #9699 reported by dimas)
          ........

2007-06-19 15:22 +0000 [r69944]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: Fix a crash that could occur when handing
          device state changes. When the state of a device changes, the
          device state thread tells the extension state handling code that
          it changed. Then, the extension state code calls the callback in
          chan_sip so that it can update subscriptions to that extension. A
          pointer to a sip_pvt structure is passed to this function as the
          call which needs a NOTIFY sent. However, there was no locking
          done to ensure that the pvt struct didn't disappear during this
          process. (issue #9946, reported by tdonahue, patch by me, patch
          updated to trunk to use the sip_pvt lock wrappers by eliel)

2007-06-19 13:55 +0000 [r69805-69895]  Joshua Colp <jcolp@digium.com>

        * /, apps/app_meetme.c: Merged revisions 69894 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r69894 | file | 2007-06-19 09:54:03 -0400 (Tue, 19 Jun 2007) | 2
          lines Perform an extra hangup check just in case. (issue #9589
          reported by bcnit) ........

        * /, res/res_features.c: Merged revisions 69846 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r69846 | file | 2007-06-19 08:57:55 -0400 (Tue, 19 Jun 2007) | 2
          lines Add parked call extension AFTER the parking slot has been
          announced, otherwise two threads will try to handle the same
          channel and it will go kaboom. (issue #9191 reported by japple)
          ........

        * main/callerid.c: Fix for building on PowerPC under Linux.

2007-06-18 19:48 +0000 [r69796]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * channels/chan_sip.c: Issue 10005 - Segfault with missing
          arguments, plus fix a missing define for SIP INFO channels

2007-06-18 19:00 +0000 [r69775-69794]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Don't count RTP timeout when involved in a
          T38 fax session. (issue #9222 reported by ivoc)

        * /, channels/chan_sip.c: Merged revisions 69765 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2
          lines Set the peer name on the dialog to the one configured in
          sip.conf and NOT the username to be used for authentication
          attempts. (issue #9967 reported by achauvin) ........

2007-06-18 17:46 +0000 [r69744]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * contrib/scripts/safe_asterisk, /: Merged revisions 69743 via
          svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r69743 | tilghman | 2007-06-18 12:45:15 -0500 (Mon, 18 Jun 2007)
          | 2 lines Issue 9998 - Remove SIG prefix, since it's not
          supported by ksh ........

2007-06-18 16:51 +0000 [r69708]  Joshua Colp <jcolp@digium.com>

        * main/dnsmgr.c: Remember the DNS lookup done when dnsmgr is called
          for the first time so that it does not needlessly spit out
          changed messages when the host really didn't change.

2007-06-18 16:35 +0000 [r69689-69702]  Russell Bryant <russell@digium.com>

        * res/res_odbc.c, apps/app_voicemail.c, res/res_config_odbc.c,
          build_tools/menuselect-deps.in, configure, funcs/func_odbc.c,
          include/asterisk/autoconfig.h.in, configure.ac, cdr/cdr_odbc.c:
          To prevent 92138749238754 more reports of "I have unixodbc
          installed, but still can't build *_odbc.so!", check for ltdl
          directly, instead of just listing it as another library to
          include in the unixodbc check in the configure script. This also
          makes ltdl show up as a dependency in menuselect so people know
          what to go install. (related to issue #9989, patch by me)

        * build_tools/prep_moduledeps: Change the use of "echo -e" to
          "printf". On systems where /bin/sh is not bash, most of the lines
          in menuselect-tree were getting a "-e" at the beginning of every
          line. I'm surprised nobody noticed this, but I think the XML
          parser was being very nice and ignoring them.

2007-06-18 16:04 +0000 [r69661-69668]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Don't defer the BYE till later on a transfer
          when the transfer itself goes kaboom and has no hope of working.

        * channels/chan_sip.c: Few minor transfer tweaks. We can't unlock
          something we never locked, and better handle a specific scenario
          with doing an attended transfer between two non-bridged calls.

2007-06-18 15:46 +0000 [r69660]  Russell Bryant <russell@digium.com>

        * Makefile: Tweak paths for BSD systems (issue #10001, stuarth)

2007-06-18 13:55 +0000 [r69625]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Fix issue where it would be possible for the
          negotiated codecs to get set back to nothing. (issue #9992
          reported by yehavi)

2007-06-15  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.5 released.

2007-06-15 20:18 +0000 [r69579]  Russell Bryant <russell@digium.com>

        * res/res_features.c: Fix a silly deadlock in res_features that I
          found while debugging on one of blitzrage's test machines. It was
          one of the situations where he was seeing hung channels, and may
          be the cause of some of the reports from other people. (related
          to issue #9235)

2007-06-15 19:23 +0000 [r69558]  Joshua Colp <jcolp@digium.com>

        * apps/app_speech_utils.c: Add support for setting the maximum
          length of acceptable DTMF in SpeechBackground.

2007-06-15 15:27 +0000 [r69518]  Russell Bryant <russell@digium.com>

        * apps/app_meetme.c: The SLATRUNK_STATUS variable indicated
          "SUCCESS" for both an answer of the incoming call on the trunk,
          or if the trunk reached its ring timeout. This patch changes the
          variable to say "RINGTIMEOUT" in that case. (issue #9973,
          reported by n00dle, patch by me)

2007-06-14 23:22 +0000 [r69434-69470]  Jason Parker <jparker@digium.com>

        * main/config.c, /: Merged revisions 69469 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4
          lines Fix an issue where the line number in an unterminated
          comment block error message would show the wrong line number.
          "Reported" to me on #asterisk (somebody posted an error message,
          and I happened to catch it) ........

        * sounds/Makefile: Update to latest versions of sound files.

2007-06-14 21:50 +0000 [r69392]  Kevin P. Fleming <kpfleming@digium.com>

        * cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c, channels/chan_phone.c,
          cdr/cdr_sqlite.c, main/logger.c, main/callerid.c, cdr/cdr_odbc.c,
          main/asterisk.c, channels/chan_mgcp.c, cdr/cdr_manager.c,
          apps/app_voicemail.c, include/asterisk/utils.h, main/pbx.c,
          main/say.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c,
          channels/chan_iax2.c: use ast_localtime() in every place
          localtime_r() was being used

2007-06-14 21:08 +0000 [r69358]  Russell Bryant <russell@digium.com>

        * main/say.c: Fix some problems with saying dates and times for the
          "tw" langauge (issue #9964, ljmid)

2007-06-14 15:21 +0000 [r69259]  Jason Parker <jparker@digium.com>

        * funcs/func_groupcount.c, /: Merged revisions 69258 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r69258 | qwell | 2007-06-14 10:15:53 -0500 (Thu, 14 Jun
          2007) | 4 lines Change a quite broken while loop to a for loop,
          so "continue;" works as expected instead of eating 99% CPU...
          Issue 9966, patch by me. ........

2007-06-13 21:19 +0000 [r69184-69222]  Joshua Colp <jcolp@digium.com>

        * channels/chan_iax2.c: Whoops...

        * channels/chan_iax2.c: Let's make chan_iax2 media only native
          transfers actually work. (issue #9376 reported by simone
          cittadini)

        * channels/iax2-parser.c: Add TXMEDIA to list so that it is
          properly displayed during iax2 packet output.

2007-06-13 19:57 +0000 [r69183]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: Move the logic for destroying a call when no
          response is received to a BYE outside of the block that checks
          for FLAG_FATAL to be set. This flag is only set when the packet
          is transmitted with the reliability set to XMIT_CRITICAL when the
          original packet is transmitted. A BYE is always sent with it set
          to XMIT_RELIABLE, meaning this code could never be encountered.
          This resulted in seeing some SIP channels that would never go
          away with the last packet sent being a BYE. (part of issue #9235,
          patch from jcmoore)

2007-06-13 19:41 +0000 [r69181]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Contains a patch for fixing an encoding
          problem when using Outlook to view voicemail emails and
          attachments. This fix has also been tested on Thunderbird,
          Evolution, Pine, and Mutt. (Issue 9336, reported by marwick,
          patched by mutterc)

2007-06-13 19:08 +0000 [r69128-69144]  Joshua Colp <jcolp@digium.com>

        * apps/app_meetme.c: Really ignore NULL frames and check whether
          the channel hungup or not. (issue #9912 reported by junky)

        * /, main/app.c: Merged revisions 69127 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r69127 | file | 2007-06-13 14:12:48 -0400 (Wed, 13 Jun 2007) | 2
          lines Return group counting to previous behavior where you could
          only have one group per category. (issue #9711 reported by
          irroot) ........

2007-06-13 16:56 +0000 [r69016-69071]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: Clarify a bit of logic. This doesn't change
          behavior in any way, but it is helpful when following the logic
          to debug problems like 9235.

        * channels/chan_iax2.c: Fix a place where a chan_iax2 pvt struct
          was accessed without the lock held. This issue was reported to me
          via email by Dmitry Mishchenko. Thanks!

        * cdr/cdr_pgsql.c: Fix a memory leak pointed out by prashant_jois
          in #asterisk-bugs. PQclear() was not called on the result
          structure after doing a PQexec(). Also, fix up some formatting in
          passing.

2007-06-12 19:36 +0000 [r69012-69014]  Joshua Colp <jcolp@digium.com>

        * channels/chan_iax2.c: Change the full frame dropping log message
          to debug to avoid future bug reports.

        * channels/chan_iax2.c: Schedule the sending of a PING packet a
          second later than previously so that it does not collide with the
          LAGRQ.

2007-06-12 19:13 +0000 [r69010]  Russell Bryant <russell@digium.com>

        * main/channel.c: In ast_channel_make_compatible(), just return if
          the channels' read and write formats already match up. There are
          code paths that call this function on a pair of channels multiple
          times. This made calls fail that were using g729 in some cases.
          The reason is that codec_g729a will unregister itself from the
          list of available translators will all licenses are in use. So,
          the first time the function got called, the right translation
          path was allocated. However, the second time it got called, the
          code would not find a translation path to/from g729 and make the
          call fail, even if the channel actually already had a g729
          translation path allocated. (SPD-32)

2007-06-12 14:23 +0000 [r68922]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c, /: Merged revisions 68921 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2
          lines Bring RTP back to Asterisk at the end of a native bridge no
          matter what. ........

2007-06-11 21:20 +0000 [r68814]  Jason Parker <jparker@digium.com>

        * include/asterisk/time.h: Solaris 10 sometimes (?) needs this
          include in order to have NULL defined.

2007-06-11 20:45 +0000 [r68781]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_directory.c: Issue 9947 - fn2 was unused / incorrectly
          used

2007-06-11 16:57 +0000 [r68733]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
          Merged revisions 68732 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11 Jun 2007) |
          1 line added check for NULL Pointer when calling misdn_new.
          Asterisk does not allow us to create channels anymore when stop
          gracefully is used :). also modified the restart_indicator to 0
          ........

2007-06-11 14:33 +0000 [r68683]  Joshua Colp <jcolp@digium.com>

        * main/channel.c, /: Merged revisions 68682 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2
          lines Improve deadlock handling of the channel list. (issue #8376
          reported by one47) ........

2007-06-11 10:29 +0000 [r68644]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
          channels/chan_misdn.c, /, channels/misdn/ie.c,
          channels/misdn/isdn_msg_parser.c: Merged revisions 68631 via
          svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11 Jun 2007) |
          1 line fixed problem that the dummybc chanels had no lock,
          checking for the lock now. Also fixed the channel restart stuff,
          we can now specify and restart particular channels too. ........

2007-06-11 04:21 +0000 [r68595]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * pbx/pbx_config.c: "dialplan save" produced garbage in the config
          file

2007-06-08 22:23 +0000 [r68527]  Russell Bryant <russell@digium.com>

        * /, apps/app_dictate.c: Merged revisions 68526 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r68526 | russell | 2007-06-08 17:22:36 -0500 (Fri, 08 Jun 2007) |
          4 lines Don't automatically hang up after running Dictate so that
          callers can exit cleanly using '#' (closes issue #9577, patch
          from Thomas Andrews) ........

2007-06-08 15:52 +0000 [r68450]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_iax2.c: actually remember the type/subclass of full
          frames that are in process

2007-06-08 00:17 +0000 [r68370-68401]  Joshua Colp <jcolp@digium.com>

        * /, main/say.c: Merged revisions 68397 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r68397 | file | 2007-06-07 20:15:33 -0400 (Thu, 07 Jun 2007) | 2
          lines Don't call ast_waitstream_full when the control file
          descriptor and audio file descriptor are not set, simply call
          ast_waitstream! (issue #8530 reported by rickead2000) ........

        * main/dnsmgr.c, /: Merged revisions 68368 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r68368 | file | 2007-06-07 19:59:04 -0400 (Thu, 07 Jun 2007) | 2
          lines Do a DNS lookup immediately upon calling the dnsmgr
          function, don't wait until a refresh happens. (issue #9097
          reported by plack) ........

2007-06-07 23:14 +0000 [r68354]  Russell Bryant <russell@digium.com>

        * /, main/say.c: Merged revisions 68351 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r68351 | russell | 2007-06-07 18:13:33 -0500 (Thu, 07 Jun 2007) |
          3 lines Fix a problem where saying a character wouldn't properly
          break out when the caller pressed '#' (issue #8113, reported by
          patbaker82, patch from jamesgolovich (hey, long time no see!) and
          patbaker82) ........

2007-06-07 23:00 +0000 [r68326]  Jason Parker <jparker@digium.com>

        * apps/app_voicemail.c: Fix incorrect French syntax of "old
          messages". Request for feedback was sent to asterisk-dev mailing
          list, with little response. Issue 9118, patch by junky.

2007-06-07 22:14 +0000 [r68313]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_iax2.c: some improvements to the IAX2 full frame
          dropping logic recently added: - use inaddrcmp(), since we have
          it - output the type of frame and subclass being dropped, and the
          type/subclass that is already being processed (which caused the
          drop)

2007-06-07 21:16 +0000 [r68280]  Russell Bryant <russell@digium.com>

        * channels/chan_agent.c, apps/app_queue.c: Fix loading persistent
          queue members when using realtime configuration for queues. Also,
          remove an unneeded leading slash for the astdb family. (issue
          #9911, patch by atis)

2007-06-07 20:25 +0000 [r68211-68249]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Fix an issue with newer phones which
          require packets be padded out to the correct length. Issue 9887,
          patch by DEA.

        * apps/app_voicemail.c, /: Merged revisions 68204 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r68204 | qwell | 2007-06-07 15:02:50 -0500 (Thu, 07 Jun 2007) | 4
          lines Don't try to save voicemail greetings unless the user
          presses '1' to accept/save. Issue 9904, patch by me. ........

2007-06-07 19:47 +0000 [r68198]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Submitting a fix for Issue 8016. Added a
          check to make sure that greetings get stored properly. (Issue
          8016, reported by edhorton, patched by alamantia with
          modification by me. Thanks to Jason Parker for the advice on
          this).

2007-06-07 19:46 +0000 [r68196]  Olle Johansson <oej@edvina.net>

        * channels/chan_features.c: Disable chan_features by default in
          menuselect

2007-06-07 19:30 +0000 [r68192]  Russell Bryant <russell@digium.com>

        * main/strcompat.c: Include stdarg.h for build issues on Solaris
          (issue #9381)

2007-06-07 18:39 +0000 [r68071-68157]  Joshua Colp <jcolp@digium.com>

        * main/channel.c: Fix logic when doing a name based channel search
          for a structure when you want to start from a specific point in
          the channel list. (issue #9324 reported by slavon)

        * apps/app_dial.c, /: Merged revisions 68070 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2
          lines Allow the 'g' option to work if used with the 'S' option.
          (issue #9888 reported by gasparz) ........

2007-06-07 10:00 +0000 [r67993-68030]  Olle Johansson <oej@edvina.net>

        * res/res_jabber.c: Adding a few Todo's to res_jabber so we don't
          forget.

        * res/res_jabber.c: Ok, we found out that this is not about if you
          have any *active* clients using TLS, but if you have initialized
          TLS at all during the lifetime of the module. So if you reload to
          disable TLS, it won't help.

        * res/res_jabber.c: If you have a jabber client that uses TLS,
          refuse unload. Bad fix, but will prevent crashes while we are
          trying to find a workaround. Iksemel development seems to have
          stalled and we might have to stop using the TCP/TLS connections
          in that library and use our own, which would scale better from a
          poll/select perspective I guess. It would also make it easier to
          migrate to OpenSSL and stop Asterisk from depending on both
          OpenSSL and GnuTLS.

        * include/asterisk/jabber.h, res/res_jabber.c: Issue #9738 - Make
          sure we can unload res_jabber. Patch by phsultan - thanks! Due to
          a bug in the iksemel library, this will not work if you are using
          GTLS in the connection. That's being investigated. If you figure
          out a way to handle that without us having to patch iksemel, let
          us know in the bug report. Thanks.

2007-06-07 00:10 +0000 [r67924-67941]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_sip.c: Merged revisions 67938 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2
          lines Only notify the devicestate system of a peer state change
          when the peer is built from the config file. (issue #9900
          reported by arkadia) ........

        * main/file.c: Properly handle cases where a stream can't be
          written to. (issue #9757 reported by junky)

2007-06-06 22:08 +0000 [r67862-67872]  Russell Bryant <russell@digium.com>

        * res/res_snmp.c: Disable reload functionality in res_snmp. It is
          not possible to initialize the snmp library more than once
          without completely unloading the module and loading it again.
          (issue #9571, reported by hristo, additional helpful debug
          information from festr, patch from me)

        * channels/chan_sip.c: Fix a crash when doing call pickups with SIP
          phones. The code unlocked the channel when it should not have.
          (issue #9652, reported by corruptor, fixed by me)

2007-06-06 19:26 +0000 [r67804]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Fix for Issue 9810. There was a segfault
          under a specific set of circumstances: 1. VoiceMailMain was
          configured in the dialplan with an extension as its argument 2. A
          message was left for this mailbox 3. Tried to call VoiceMailMain
          but hung up before entering password. This was fixed by checking
          that a pointer was non-null prior to trying to dereference it.
          (Issue 9810, reported by xmarksthespot, patched by Corydon76 with
          modifications by me).

2007-06-06 16:55 +0000 [r67716]  Russell Bryant <russell@digium.com>

        * main/channel.c, /, include/asterisk/linkedlists.h: Merged
          revisions 67715 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) |
          5 lines We have some bug reports showing crashes due to a double
          free of a channel. Add a sanity check to ast_channel_free() to
          make sure we don't go on trying to free a channel that wasn't
          found in the channel list. (issue #8850, and others...) ........

2007-06-06 13:30 +0000 [r67594-67650]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c, /: Merged revisions 67649 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2
          lines Reinvite the RTP back to the Asterisk machine when the
          timeout happens. (issue #9888 reported by gasparz) ........

        * main/translate.c: Fix plc_samples warning when registering a
          translator. (issue #9897 reported by xylome)

        * apps/app_directed_pickup.c: Include macroexten while searching
          for a channel to pick up in case they are in a macro. (issue
          #9491 reported by jamesb63)

        * res/res_agi.c: Make the new "agi debug off" CLI command work.
          (issue #9890 reported by eliel)

        * /, main/devicestate.c: Merged revisions 67593 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r67593 | file | 2007-06-06 08:18:36 -0400 (Wed, 06 Jun 2007) | 2
          lines Revert channel name splitting fix for Zap. The moral of the
          story is don't use - in your user/peer names. (issue #9668
          reported by stevedavies) ........

2007-06-05 23:01 +0000 [r67558]  Russell Bryant <russell@digium.com>

        * apps/app_meetme.c: Fix some crashes related to the use of the
          "meetme" CLI command. The code for this command was not locking
          the conference list at all. (issue #9351, reported by and patch
          submitted by Junk-Y, committed patch is different and by me)

2007-06-05 21:30 +0000 [r67526]  Steve Murphy <murf@digium.com>

        * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: this fixes bug
          9883, wherein macros were not allowing the includes construct.
          fixed and tested, looks OK. Now includes can serve as an adjunct
          to catch.

2007-06-05 20:53 +0000 [r67457-67492]  Russell Bryant <russell@digium.com>

        * include/asterisk/linkedlists.h: This bug has been hanging over my
          head ever since I wrote this SLA code. Every time I tried to go
          debug it by adding some debug output, the behavior would change.
          It turns out I wasn't crazy. I had the following piece of code:
          if (remove) AST_LIST_REMOVE_CURRENT(...); Well,
          AST_LIST_REMOVE_CURRENT was not wrapped in braces, so my
          conditional statement didn't do much good at all. It always ran
          at least all of the macro minus the first statement, so I was
          seeing list entries magically disappear when they weren't
          supposed to. After many hours of debugging, I have come to this
          extremely irritating fix. :) (issues #9581, #9497)

        * channels/chan_zap.c: Suppress a bunch of debug output unless
          option_debug is on

2007-06-05 18:32 +0000 [r67424]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Fix for bug number 9786, wherein voicemails
          saved to IMAP storage using extensions other than gsm were unable
          to be played over the phone. (Issue 9786, reporter:
          xmarksthespot, Patched by xmarksthe spot with revisions by me,
          reviewed by Russell Bryant).

2007-06-05 18:18 +0000 [r67421]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Correctly update date/time on devices
          throughout the life of the device, instead of just at
          registration. Issue 9152, yet another patch by DEA.

2007-06-05 18:17 +0000 [r67420]  Steve Murphy <murf@digium.com>

        * pbx/pbx_ael.c: Added code to automatically add a default case to
          switches that don't have one. In some cases, rather than fall
          thru, it results in a goto with -1 result, which terminates the
          extension; a sort of dialplan seqfault, sort of. This was
          required to fix bug reported in 9881

2007-06-05 17:07 +0000 [r67360-67372]  Russell Bryant <russell@digium.com>

        * main/channel.c: Handle a failure in malloc() in
          ast_safe_string_alloc()

        * main/channel.c: Fix a problem that showed itself by causing Zap
          channel names to be completely bogus on my machine.
          ast_safe_string_alloc() was broken. It called vsnprintf() on a
          va_args list twice without re-initializing it. After the first
          usage, va_end() and va_start() must be called again.

2007-06-05 16:14 +0000 [r67329-67334]  Christian Richter <christian.richter@beronet.com>

        * /, channels/misdn/chan_misdn_config.h: Merged revisions 67307 via
          svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05 Jun 2007) |
          1 line briding is a bool, fixed copy and paste issue. ........

        * channels/chan_misdn.c, /: Merged revisions 67306 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05
          Jun 2007) | 1 line simplified the EVENT_SETUP handling in the
          cb_events function a lot. Commented the different possibilities a
          bit and made functions of shared code. When the dialed extension
          does not exist in the extensions.conf we'll jump into the 'i'
          extension if this does exist, else we disconnect the call with
          the cause:1 = No Route to Destination. ........

2007-06-05 15:51 +0000 [r67308]  Russell Bryant <russell@digium.com>

        * main/asterisk.c, main/loader.c, include/asterisk/module.h: When
          shutting down "gracefully", go through and run the unload()
          callbacks for all of the modules. "stop now" is considered a
          non-graceful shutdown and will not go through this process.
          (issue #9804, reported by chrisost, patch by me)

2007-06-05 15:22 +0000 [r67304]  Joshua Colp <jcolp@digium.com>

        * channels/chan_iax2.c: Only muck with the thread structure if an
          idle one was found/created.

2007-06-05 14:35 +0000 [r67270]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_iax2.c: ensure that a burst of full frames
          (AST_FRAME_DTMF being the prime example) will not be processed
          out of order... this is a brute force fix, but seems to be the
          safest fix for now (thanks to the Digium PQ department for
          finding this bug)

2007-06-05 10:25 +0000 [r67210]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn_config.c, channels/chan_misdn.c, /,
          channels/misdn/chan_misdn_config.h: Merged revisions 67209 via
          svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 Jun 2007) |
          1 line added possibility to deactivate bridging per port ........

2007-06-04 23:43 +0000 [r67162]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * /, funcs/func_math.c: Merged revisions 67161 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r67161 | tilghman | 2007-06-04 18:41:49 -0500 (Mon, 04 Jun 2007)
          | 2 lines According to MATH, 0+1181000386 = 1181000448. Oops.
          ........

2007-06-04 23:31 +0000 [r67158]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix up a bunch of places where the iax2 pvt
          structure can disappear and the code did not account for it and
          crashes. (issues #9642, #9569, #9666, probably others ... based
          on the work by stevedavies and mihai, with additional changes
          from me)

2007-06-04 23:26 +0000 [r67121-67156]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Fix for skinny keepalives. If there is no
          traffic from the phone for (keep_alive * 1100) ms (arbitrarily
          adding 10% for network issues, etc), unregister the device. Issue
          8394, patch by DEA.

        * channels/chan_mgcp.c: Fixes for dtmf/dialing with mgcp (similar
          to the recent fix for chan_skinny) Issue 9855, patch by DEA.

2007-06-04 22:28 +0000 [r67119]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Add comments for two functions that get
          called with the appropriate call locked, but perform operations
          that could result in the pvt structure getting destroyed before
          returning again, causing numerous seg faults all over the module.
          (inspired by issues #9642, #9569, and #9666, and the work done by
          stevedavies and mihai)

2007-06-04 21:59 +0000 [r67073]  Steve Murphy <murf@digium.com>

        * main/cdr.c: This typo has been here since 1.4 forked. It has been
          the source of heartburn to many a dialplan/CDR programmer.

2007-06-04 21:47 +0000 [r67071]  Russell Bryant <russell@digium.com>

        * main/rtp.c: Add a missing \n. (pointed out by jcmoore on IRC)

2007-06-04 19:31 +0000 [r67064-67068]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Better handle SIP devices that say they have
          SDP content... but really don't. (issue #9398 reported by
          mthomasslo)

        * apps/app_dial.c: Initialize cidname variable to nothing since it
          may be used without having been touched. (issue #9661 reported by
          dimas)

        * res/res_features.c: Returning a value that indicates the parking
          of a call was a success when it really wasn't (because the
          parking slot selected was in use) is the wrong thing to do.
          (issue #9723 reported by mdu113)

2007-06-04 17:11 +0000 [r67061]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * contrib/init.d/rc.debian.asterisk,
          contrib/init.d/rc.mandrake.asterisk, /,
          contrib/init.d/rc.redhat.asterisk,
          contrib/init.d/rc.gentoo.asterisk,
          contrib/init.d/rc.mandrake.zaptel,
          contrib/init.d/rc.slackware.asterisk: Merged revisions 67060 via
          svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r67060 | tilghman | 2007-06-04 12:10:30 -0500 (Mon, 04 Jun 2007)
          | 2 lines Add revision Id tags (by request of tzafrir) ........

2007-06-04 16:02 +0000 [r67026]  Russell Bryant <russell@digium.com>

        * configure, configure.ac: Change the configure script to build a
          test program against libcurl to make sure the results from
          curl-config can be used to compile successfully. This is intended
          to help prevent a situation where you are cross compiling, and
          the configure script finds the curl library installed on the
          host. (issue #9865, reported and patched by zandbelt)

2007-06-04 15:50 +0000 [r67021]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * res/res_jabber.c: Issue 9739 - Malformed jid causes a crash

2007-06-04 15:47 +0000 [r67018-67020]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Resolve a deadlock in chan_iax2. When
          handling an implicit ACK to a frame that was marked as the final
          transmission for a call, don't call iax2_destroy() for that call
          while the global frame queue is still locked. There is a very
          nice explanation of the deadlock in the report. (issue #9663,
          thorough report and patch from stevedavies, additional positive
          test reports from mihai and joff_oconnell)

        * include/asterisk/stringfields.h: Fix some compiler warnings in
          C++ modules. (issue #9866, reported by osk, patch by Corydon76)

2007-06-01 21:45 +0000 [r66919]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * funcs/func_odbc.c: On some drivers, deallocating the statement
          handle isn't enough. We also have to clear the cursor (nice,
          Oracle)

2007-06-01 21:31 +0000 [r66897-66917]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Removing extraneous debugging lines from
          revision 66897. Sorry :)

        * apps/app_voicemail.c: Submitting a fix for voicemail with IMAP
          storage. Attachments with format specified as gsm were duplicated
          (i.e. two attachments) were left. Thank you very much to
          xmarksthespot for submitting the patch that fixed this. (Issues
          9787 and 8873, Reported by xmarksthespot and jerjer, patched by
          xmarksthespot)

2007-06-01 19:41 +0000 [r66879-66881]  Russell Bryant <russell@digium.com>

        * channels/chan_skinny.c: Changes to the way DTMF is handled in the
          core broke dialing in chan_skinny. This patch makes chan_skinny
          usable again. I did not end up testing this, but there are
          multiple positive test reports listed in the bug report. (issue
          #9596, reported by pj, testing by pj and mvanbaak, and the fix
          was written by DEA)

        * apps/app_page.c: List app_meetme as a module that app_page
          depends on.

2007-05-31 23:03 +0000 [r66821]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * doc/asterisk.8: Issue 9850 - update preferred command line syntax

2007-05-31 18:41 +0000 [r66775]  Russell Bryant <russell@digium.com>

        * res/res_speech.c, include/asterisk/app.h,
          include/asterisk/speech.h: Change a couple of header files to not
          use "new", which is a reserved keyword in C++. (issue #9830,
          reported by osk)

2007-05-31 17:15 +0000 [r66770]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * /, apps/app_macro.c: Merged revisions 66744 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r66744 | tilghman | 2007-05-31 10:58:45 -0500 (Thu, 31 May 2007)
          | 2 lines Issue 9818 - Fix for issue 8329 breaks pbx_realtime.
          Issue 8329 will remain unfixed for pbx_realtime, but only because
          we lack core API to do it. ........

2007-05-31 16:14 +0000 [r66768]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_sip.c: Merged revisions 66764 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2
          lines It is now possible for this path of execution to have the
          frame pointer be NULL, therefore we need to check for it before
          trying to access it. (issue #9836 reported by barthpbx) ........

2007-05-30 23:26 +0000 [r66671]  Mark Michelson <mmichelson@digium.com>

        * apps/app_voicemail.c: Fixed seg-faults when recording greetings
          in voicemail with IMAP enabled. (Issue No. 9735, reported by
          xmarksthespot, patched by me)

2007-05-30 17:28 +0000 [r66602-66639]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Silly me for having out of date source! Oh
          well... I'm still leaving my comment.

        * channels/chan_sip.c: When calling some peer/host that may not
          exist/reply back... don't keep the dialog in memory for all of
          eternity.

        * channels/chan_zap.c, channels/chan_features.c: Change how channel
          names are generated a bit. (issue #9825 reported by eldadran)

2007-05-29 21:56 +0000 [r66538]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * /, funcs/func_strings.c: Merged revisions 66537 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r66537 | tilghman | 2007-05-29 16:49:35 -0500 (Tue, 29 May 2007)
          | 2 lines If the value of a variable passed to FIELDQTY is blank,
          then FIELDQTY should return 0, not 1. ........

2007-05-29 19:32 +0000 [r66474-66503]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Properly handle 408 request timeout -
          according to the RFC, the dialog dies if a request in a dialog
          gets this response.

        * channels/chan_sip.c: Don't issue hangup on hangup on hangup on
          hangup (for jcmoore)

2007-05-29 16:44 +0000 [r66437]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: Handle cases where a frame may have no data. (issue
          #9519 reported by dmb)

2007-05-29 16:07 +0000 [r66404-66414]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Don't reset hangupcause if we already have
          one

        * channels/chan_sip.c: Tracking down hanging channels, killing them
          one by one. Issue #9235 and related

2007-05-29 15:43 +0000 [r66398]  Joshua Colp <jcolp@digium.com>

        * doc/datastores.txt: Update datastores documentation. (issue #9801
          reported by mnicholson)

2007-05-29 09:41 +0000 [r66363]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Merged revisions 66349 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2
          lines Issue #9802 - Change inuse counter on CANCEL ........

2007-05-28 23:16 +0000 [r66312]  Joshua Colp <jcolp@digium.com>

        * channels/chan_zap.c: Make the usedistinctiveringdetection option
          work again. (issue #9823 reported by premeau)

2007-05-27 04:12 +0000 [r66244]  Jason Parker <jparker@digium.com>

        * channels/chan_zap.c: I don't know what this was trying to do, but
          it's clearly incorrect. Issues 9808 and 9809.

2007-05-25 14:43 +0000 [r66160]  Kevin P. Fleming <kpfleming@digium.com>

        * configure, configure.ac: have to check for OSP toolkit _after_
          checking for OpenSSL

2007-05-25 14:41 +0000 [r66159]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * /, main/say.c: Merged revisions 66127 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r66127 | tilghman | 2007-05-25 08:46:35 -0500 (Fri, 25 May 2007)
          | 2 lines Issue 9791 - Fix pronunciation of seconds in Dutch
          ........

2007-05-25 14:28 +0000 [r66157]  Kevin P. Fleming <kpfleming@digium.com>

        * configure, configure.ac, channels/chan_gtalk.c, makeopts.in,
          res/res_jabber.c: handle the GNUTLS library properly in the
          configure script and build system don't build in OSP support
          unless we have found and are allowed to use SSL support

2007-05-24 22:23 +0000 [r66076]  Russell Bryant <russell@digium.com>

        * main/channel.c: if the string field init fails, clean up the
          stuff that was allocated already

2007-05-24 22:16 +0000 [r66074]  Joshua Colp <jcolp@digium.com>

        * main/slinfactory.c: Fix slinfactory logic when dealing with
          frames coming in that may already be in the signed linear format.

2007-05-24 22:07 +0000 [r66068-66070]  Russell Bryant <russell@digium.com>

        * main/channel.c: Check the result of ast_string_field_init() in
          ast_channel_alloc()

        * main/rtp.c: Make 1.4 build on my machine, too..

2007-05-24 20:54 +0000 [r66029-66030]  Jason Parker <jparker@digium.com>

        * configure: Rebuild configure script for previous ar fix.

        * configure.ac: Following moving strip to AC_PATH_TOOL, we need to
          do something similar for ar.

2007-05-24 20:42 +0000 [r65978-66026]  Russell Bryant <russell@digium.com>

        * configure, include/asterisk/autoconfig.h.in, configure.ac:
          Checking for the strip application needs to be done with
          AC_PATH_TOOL instead of AC_PATH_PROG to properly handle cross
          compilation environments.

        * Makefile: Clear CFLAGS before running make for menuselect. (issue
          #9784, reported by ovi, patch by me)

2007-05-24 18:28 +0000 [r65965-65967]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_gtalk.c: oops, use #ifdef instead of #if

        * channels/chan_gtalk.c: don't reference GnuTLS headers and
          functions unless the configure script found it

        * main/rtp.c: don't use uninitialized variables

2007-05-24 15:27 +0000 [r65902]  Joshua Colp <jcolp@digium.com>

        * main/manager.c: Add the ability to blacklist certain commands
          from being executed using the Command AMI action. (issue #9240
          reported by junky)

2007-05-24 15:26 +0000 [r65892-65901]  Olle Johansson <oej@edvina.net>

        * channels/chan_gtalk.c: Issue 7672 - fix by zandbelt - Asterisk
          core dump since the GnuTLS interface did not support
          multithreading correctly.

        * channels/chan_gtalk.c: Issue 8193 - NAT issues with gtalk/STUN.
          Patch by phsultan. Thanks!

2007-05-24 15:16 +0000 [r65877-65883]  Jason Parker <jparker@digium.com>

        * .cleancount: Update cleancount for that last commit - just for
          good measure.

        * include/asterisk/translate.h, codecs/codec_speex.c,
          main/translate.c, codecs/codec_ilbc.c: Fix handling of
          zero-length frames when a codec is capable of native PLC. Issue
          9183, patch by Mihai.

2007-05-24 15:08 +0000 [r65866]  Dwayne M. Hubbard <dhubbard@digium.com>

        * funcs/func_math.c: merged qwell's func_math patch for issue 9507

2007-05-24 15:08 +0000 [r65863]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: I like it when the RTP stack compiles myself...

2007-05-24 15:05 +0000 [r65857]  Olle Johansson <oej@edvina.net>

        * channels/chan_gtalk.c: Issue 7686, fix by phsultan, NAT issues
          when calling from gtalk to SIP over nat.

2007-05-24 15:04 +0000 [r65842-65853]  Russell Bryant <russell@digium.com>

        * apps/app_festival.c: Ensure that frames are fully initialized.
          This will probably fix getting weird timestamp log messages in
          logs when using the Festival app. (issue #9781, patch by me)

        * main/rtp.c: Fix the calculation of the RTT for RTCP. The previous
          code would result in oscillating and incorrect data.
          Additionally, the RTT would sometimes report negative values due
          to incorrect calculations. (issue #9601, patch from davetroy)

2007-05-24 14:48 +0000 [r65841]  Olle Johansson <oej@edvina.net>

        * channels/chan_gtalk.c: Issue #8536 - Caller ID not set in CDR for
          jingle

2007-05-24 14:42 +0000 [r65839]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_sip.c: Merged revisions 65837 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2
          lines Allow RFC2833 to be negotiated when an INVITE comes in
          without SDP and is not matched to a user or peer. (issue #9546
          reported by mcrawford) ........

2007-05-24 14:38 +0000 [r65836]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c, res/res_jabber.c: Issue 8409 - phsultan -
          Fix "login" as component to jabber server. ...and, by accident,
          fix a bug in chan_sip for stopping a loop on retransmits of BYE
          requests.

2007-05-24 09:37 +0000 [r65768]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c, /: Merged revisions 65767 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24
          Mai 2007) | 1 line we should only activate the generator in
          chan_misdn, when asterisk hask not yet taken the call
          (WAITING4DIGS state). Alerting audio will be generated fomr
          asterisk for example. ........

2007-05-23 20:59 +0000 [r65677-65685]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_iax2.c: start the delayed PBX when receive voice or
          video full frames as well, and comment this delayed-PBX activity

        * /, channels/chan_sip.c: Merged revisions 65682 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007)
          | 2 lines ensure that variables are set on a newly created
          channel before we start a PBX on it ........

        * channels/chan_iax2.c: clear the 'delay PBX' flag when we are
          ready to start the PBX

        * channels/chan_iax2.c: don't start a PBX on a new incoming IAX2
          channel until we have some sort of response to our ACCEPT (ACK or
          anything else)

        * /, channels/chan_iax2.c: Merged revisions 65676 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007)
          | 2 lines if we are going to set variables on a newly created
          channel, it should be done *before* we start the PBX on it
          ........

2007-05-23 13:07 +0000 [r65589]  Russell Bryant <russell@digium.com>

        * channels/chan_zap.c, /: Merged revisions 65588 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) |
          3 lines Revert revision 62417 as someone reported problems with
          it to Mark. This was related to issue #9588. ........

2007-05-22 20:25 +0000 [r65541]  Kevin P. Fleming <kpfleming@digium.com>

        * build_tools/make_version: when building a version string for a
          developer branch, include the base branch in the version string

2007-05-22 18:40 +0000 [r65501]  Russell Bryant <russell@digium.com>

        * apps/app_voicemail.c, channels/chan_zap.c: List res_smdi as a
          dependency for app_voicemail and chan_zap (Thanks to mnicholson
          for pointing it out)

2007-05-22 15:04 +0000 [r65452]  Joshua Colp <jcolp@digium.com>

        * apps/app_meetme.c: Remove a double const.

2007-05-22 14:02 +0000 [r65408]  BJ Weschke <bweschke@btwtech.com>

        * apps/app_followme.c: Fix a problem with flag recognition.

2007-05-22 13:09 +0000 [r65394]  Russell Bryant <russell@digium.com>

        * /, apps/app_queue.c: Merged revisions 65389 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r65389 | russell | 2007-05-22 08:07:03 -0500 (Tue, 22 May 2007) |
          4 lines Fix a memory leak that I just noticed in the device state
          handling in app_queue. On most device state changes, it would
          leak roughly 8 to 64 bytes (the length of the name of the
          device). ........

2007-05-22 08:12 +0000 [r65342]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c, /: Merged revisions 65328 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22
          Mai 2007) | 1 line we stop the tones only when we're in the
          pre-call phase, otherwise e.g. when in CONNECTED state we should
          not stop tones when we receive an Information Message ........

2007-05-20 17:59 +0000 [r65250]  Joshua Colp <jcolp@digium.com>

        * res/res_agi.c: res_agi needs to export two symbols
          (ast_agi_register and ast_agi_unregister) for usage by others.
          (issue #9755 reported by mnicholson)

2007-05-18 22:26 +0000 [r65200-65201]  Steve Murphy <murf@digium.com>

        * main/cdr.c: Ugh. The svnmerge didn't catch the shift from cdr.c
          to main/cdr.c, and neither did I. This is the remainder of the
          9717 patch, the fix for the run-away FAIL status for a call

        * apps/app_dial.c, /, include/asterisk/cdr.h: Merged revisions
          65172 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1
          line This update will fix the situation that occurs as described
          by 9717, where when several targets are specified for a dial, if
          any one them reports FAIL, the whole call gets FAIL, even though
          others were ringing OK. I rearranged the priorities, so that a
          new disposition, NULL, is at the lowest level, and the
          disposition get init'd to NULL. Then, next up is FAIL, and next
          up is BUSY, then NOANSWER, then ANSWERED. All the related set
          routines will only do so if the disposition value to be set to is
          greater than what's already there. This gives the intended
          effect. So, if all the targets are busy, you'd get BUSY for the
          call disposition. If all get BUSY, but one, and that one rings is
          not answered, you get NOANSWER. If by some freak of nature, the
          NULL value doesn't get overridden, then the disp2str routine will
          report NOANSWER as before. ........

2007-05-18 18:16 +0000 [r65041-65123]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Merged revisions 65122 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2
          lines Not getting an ACK to a 200 OK in the initial invite is
          critical to the call. ........

        * /, channels/chan_sip.c: Merged revisions 65075 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5
          lines Issue 9235 - part of the problem, maybe not all. Please
          retry with this patch (and no other patch) if you have problems
          with hanging SIP channels. Thank you. A special Thank You to
          WeBRainstorm that gave me access to his system. ........

        * channels/chan_sip.c: - Adding support for putting calls OFF hold
          with a re-invite with blank SDP. This was a bug found while doing
          tests at SIPit in Antwerp. - In order to not duplicate code, I
          restructured some of the code for putting calls on/off hold.
          Thanks DEA for reminding me. This fix has been asleep in the
          videocaps branch until now.

2007-05-18 12:40 +0000 [r65039]  Christian Richter <christian.richter@beronet.com>

        * /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged
          revisions 65007 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) |
          1 line fixed a warning regarding Keypad encoding. encode the IE
          sending_complete at the right position. ........

2007-05-18 10:37 +0000 [r64974]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Issue 9487 - stop media flows at hangup of
          call

2007-05-18 08:58 +0000 [r64904]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c, /: Merged revisions 64902 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18
          Mai 2007) | 1 line we *need* to send a PROCEEDING when
          sending_complete is set, even if need_more_infos is requested.
          ........

2007-05-18 02:48 +0000 [r64868]  Russell Bryant <russell@digium.com>

        * apps/app_queue.c: Fix a small bug I noticed while working on
          something else. app_queue did not unregister its device state
          monitoring callback in unload_module(). So, this would make
          Asterisk crash on the first device state change after you unload
          the module.

2007-05-17 21:19 +0000 [r64820]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * /, include/asterisk/linkedlists.h: Merged revisions 64819 via
          svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r64819 | tilghman | 2007-05-17 16:14:36 -0500 (Thu, 17 May 2007)
          | 2 lines How is it that we never caught that this is returning
          the opposite of our documentation, until now? ........

2007-05-17 16:53 +0000 [r64761]  Jason Parker <jparker@digium.com>

        * apps/app_voicemail.c, /: Merged revisions 64758 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r64758 | qwell | 2007-05-17 11:52:38 -0500 (Thu, 17 May 2007) | 4
          lines If we have a negative current message, we shouldn't go back
          even further... Issue 9727. ........

2007-05-17 16:52 +0000 [r64756-64759]  Russell Bryant <russell@digium.com>

        * contrib/scripts/astxs (removed): Remove script that is no longer
          functional since the build system was redone. (issue #9340,
          reported by junky)

        * apps/app_dial.c: Increase the size of a buffer to support longer
          dial strings for channels. (issue #9291, reported and fix
          suggested by meni)

2007-05-17 16:10 +0000 [r64720-64754]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Even more direct RTP setup fixes! Don't
          allow a codec that isn't supported to creep into the SDP of
          either side. (issue #9446 reported by marcelbarbulescu)

        * apps/app_voicemail.c: Fix authuser support. (issue #9740 reported
          by xmarksthespot)

2007-05-17 06:13 +0000 [r64686]  Russell Bryant <russell@digium.com>

        * README: Update the main README to reflect the new build process
          for 1.4 and above. (issue #9725, patch by eliel)

2007-05-16 11:01 +0000 [r64516-64609]  Olle Johansson <oej@edvina.net>

        * /: Blocking patch already in this code

        * channels/chan_sip.c: Fix auth on BYE. (Different patch than for
          1.2)

        * channels/chan_sip.c: Issue #9681 - Handle www-auth on BYE

        * channels/chan_sip.c: Final part of issue #9483 - fixing
          transfer() of sip calls in the dial plan (twilson)

        * channels/chan_sip.c: Issue #9439 - properly handle username
          parameters in SIP uri.

        * /, channels/chan_sip.c: Merged revisions 64535 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2
          lines Support SIP uri's starting with SIP: and sip: (reported by
          Tony Mountfield on the mailing list. Thanks!) ........

        * /, channels/chan_sip.c: Merged following patch with a lot of
          changes for 1.4 ------ Merged revisions 64514 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6
          lines Issue #9726 - rlister - Better logging for ACL denials
          While at it, also added better logging and handling of peers that
          are not supposed to register. My patch, stole the issue report
          from Russell. My apologies, Russell :-) ........

2007-05-16 08:44 +0000 [r64515]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c, /: Merged revisions 64513 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16
          Mai 2007) | 1 line in the case immediate=yes, we directly jump
          into the dialplan, where people can use PlayTones to indicate a
          Dialtone, so we don't need to to that by ourself. also we should
          not do a dialtone_indicate for incoming calls on a TE port in
          overlapdialmode. ........

2007-05-15 19:52 +0000 [r64353-64426]  Russell Bryant <russell@digium.com>

        * res/res_features.c: Properly fix a problem that occurs when you
          set PARKINGEXTEN to an exten where a call is already parked.
          (issue #9723, patch by me)

        * res/res_features.c: When someone requests a specific parking
          space using the PARKINGEXTEN variable, ensure that no other
          caller is already there. (issue #9723, reported by mdu113, patch
          by me)

2007-05-14 19:26 +0000 [r64324]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Change -2 to XMIT_ERROR to clarify a bit
          more

2007-05-14 19:13 +0000 [r64306]  Russell Bryant <russell@digium.com>

        * channels/chan_alsa.c: Properly handle AST_CONTROL_PROGRESS by
          just ignoring it. An unknown indication will trigger an error and
          cause sounds to stop, which in this case, is ringing.

2007-05-14 18:52 +0000 [r64280]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Handle network errors, like host or network
          unreachable, in a better way. This means that calls to hosts or
          qualify (OPTION) messages will fail quicker if the TCP/IP stack
          tells us that there is an issue. Since this is an unconnected UDP
          socket, we will not get error messages directly in most cases,
          but maybe on the second and third try. This is already
          implemented in trunk.

2007-05-14 18:48 +0000 [r64240-64278]  Joshua Colp <jcolp@digium.com>

        * codecs/codec_speex.c: Properly set datalen field when doing PLC
          in codec_speex. (issue #9722 reported by mihai)

        * /, main/devicestate.c: Merged revisions 64275 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r64275 | file | 2007-05-14 14:34:06 -0400 (Mon, 14 May 2007) | 2
          lines Only perform stripping of - strings from the channel name
          for Zap channels. Anywhere else we might remove a legitimate part
          of a device name. (issue #9668 reported by stevedavies) ........

        * main/channel.c: Fix scenario where if a phone that simply called
          Echo() put itself on hold it could never get off hold.

2007-05-14 13:58 +0000 [r64193]  Steve Murphy <murf@digium.com>

        * main/cdr.c, main/pbx.c, channels/chan_local.c: As per 9570,
          worrisome CDR warnings have been removed, that are either not
          helpful, or not relevant.

2007-05-14 10:39 +0000 [r64157]  Olle Johansson <oej@edvina.net>

        * main/channel.c: Add hangupcause when we lack codecs for
          transcoding

2007-05-12 22:27 +0000 [r64044-64114]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: This concludes my final adventure with
          bitmasks and the onhold flag. Would anyone care for some peanuts?

        * channels/chan_sip.c: Tweak hold flags some more. They can be of
          three states when active: active, inactive, one direction.

        * channels/chan_sip.c: Ensure the onhold flag is set no matter what
          when being put on hold.

2007-05-11 20:16 +0000 [r63982]  Jason Parker <jparker@digium.com>

        * main/manager.c: Hide manager password from "manager show user
          foo". I realize that there are other ways to get this, but we
          really don't need to just show it in plain text so easily. Issue
          9273, patch by junky

2007-05-11 16:35 +0000 [r63905]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * contrib/scripts/safe_asterisk, Makefile, /: Merged revisions
          63903 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r63903 | tilghman | 2007-05-11 11:31:03 -0500 (Fri, 11 May 2007)
          | 2 lines Issue 9121 - fixups for safe_asterisk script ........

2007-05-11 16:05 +0000 [r63886]  Russell Bryant <russell@digium.com>

        * main/manager.c: When MD5 authentication is not possible because
          there is no challenge present, either because the Challenge
          action was never issued, or some other reason, give a proper
          error message and return an error instead of claiming that the
          user wasn't found. (reported by jsmith on IRC)

2007-05-11 15:43 +0000 [r63872]  Joshua Colp <jcolp@digium.com>

        * res/res_features.c: Make the PARKINGEXTEN feature of parking
          actually work. (issue #9708 reported by mdu113)

2007-05-10 23:15 +0000 [r63830]  Jason Parker <jparker@digium.com>

        * /, channels/chan_iax2.c: Merged revisions 63828 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4
          lines Fix an issue with trying to kill a thread before it gets
          created. Issue 9709, patch by nic_bellamy. ........

2007-05-10 22:23 +0000 [r63804]  Russell Bryant <russell@digium.com>

        * main/manager.c: Strip terminal escape sequences from CLI command
          output that is going to be sent out over the manager interface.
          (issue #9659, reported by pari, fixed by me)

2007-05-10 20:48 +0000 [r63750]  Doug Bailey <dbailey@digium.com>

        * main/callerid.c: Add test for negative offsets in cid data to
          prevent infinite loops.

2007-05-10 20:46 +0000 [r63749]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Merged revisions 63748 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4
          lines Do not allocate SIP pvt's for PEERs we can not reach. This
          was seen as a lot of dialogs being created then immediately
          destroyed at reload/restart of the SIP channel. ........

2007-05-09 19:22 +0000 [r63656-63698]  Joshua Colp <jcolp@digium.com>

        * main/channel.c: Use the DTMF frame on the channel when returning
          a DTMF frame from AST_FRAME_NULL or AST_FRAME_VOICE.

        * channels/chan_sip.c: Do not prematurely go on hold if sendonly
          was not actually set.

2007-05-09 17:25 +0000 [r63654]  Matthew Fredrickson <creslin@digium.com>

        * channels/chan_zap.c, /: Merged revisions 63653 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2
          lines Make sure we only create a DSP if it's requested on
          SUB_REAL ........

2007-05-09 16:55 +0000 [r63612]  Russell Bryant <russell@digium.com>

        * main/channel.c: Modify ast_senddigit_begin() to use the same
          assumptions used elsewhere in the code in that if a channel does
          not have a send_digit_begin() callback, it only cares about DTMF
          END events. (pointed out by Michael Neuhauser on the asterisk-dev
          list)

2007-05-09 16:54 +0000 [r63611]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_sip.c: Merged revisions 63610 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2
          lines Properly handle hints that point to multiple devices in
          chan_sip. Why chan_sip is even doing this I have no idea but I
          would rather not go into a rant. (issue #9536 reported by
          rlister) ........

2007-05-09 16:43 +0000 [r63608]  Russell Bryant <russell@digium.com>

        * main/channel.c: Only call ast_senddigit_begin() in
          ast_senddigit() if the channel has a send_digit_begin() callback.
          Checking the END_DTMF_ONLY flag was the wrong thing to do,
          because that flag indicates that a *bridged* channel only wants
          DTMF END events coming from this channel.

2007-05-09 14:50 +0000 [r63566]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * /, apps/app_directory.c: Merged revisions 63565 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r63565 | tilghman | 2007-05-09 09:48:06 -0500 (Wed, 09 May 2007)
          | 2 lines Replicate fix from 51158 (app_voicemail) to
          app_directory (Issue 9224) ........

2007-05-09 13:24 +0000 [r63535]  Russell Bryant <russell@digium.com>

        * Makefile: I have seen multiple people post questions trying to
          figure out what the message "The configure script must be
          executed before running 'make'" means. So, add another like that
          says to specifically run ./configure. If this isn't obvious
          enough, then they should be using something like AsteriskNOW and
          not installing from source.

2007-05-09 13:17 +0000 [r63534]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /,
          channels/misdn/isdn_msg_parser.c: Merged revisions
          62945,63402,63519 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) |
          1 line when we're in state WAITING4DIGS, we use the asterisk
          tone-generator which prods us, so we can't just return -1 in
          misdn_write in this case. Added a MISDN_KEYPAD channel variable,
          and fixed the sending of keypad. this enables us to modify the
          call forward parameters in the switch. ........ r63402 | crichter
          | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line added
          application misdn_check_l2l1 which tries to pull up the L1/L2 on
          all ports that have the layers down in a group. It waits then for
          a timeout. This helps for scenarios where multiple PMP BRIs are
          grouped together, or where a provider has a faulty PTP
          Implementation, that looses the L2 after a while. ........ r63519
          | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line
          release_chan frees ch, so we should never touch ch after
          release_chan, this may cause segfaults. ........

2007-05-09 13:04 +0000 [r63532]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Don't retransmit 200 OK's on ignore status.
          (Reported on asterisk-users)

2007-05-08 22:38 +0000 [r63478]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * /, apps/app_macro.c: Merged revisions 63477 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r63477 | tilghman | 2007-05-08 17:19:15 -0500 (Tue, 08 May 2007)
          | 2 lines Issue 9602 - segfault in app_macro ........

2007-05-08 16:53 +0000 [r63403-63448]  Russell Bryant <russell@digium.com>

        * res/res_features.c: I mixed up the use of the find_feature()
          function, so I renamed it find_dynamic_feature, and changed the
          code to use the correct lock when using it.

        * res/res_features.c: Use a read/write lock when accessing the
          built-in features.

        * contrib/scripts/realtime_pgsql.sql (added),
          contrib/realtime_pgsql.sql (removed): Move realtime_pgsql.sql to
          contrib/scripts to be with the rest of the sql examples. (issue
          #9676, suretec)

2007-05-08 06:22 +0000 [r63360]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_voicemail.c, /: Merged revisions 63359 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r63359 | tilghman | 2007-05-08 01:20:16 -0500 (Tue, 08 May 2007)
          | 2 lines Issue 9527 - upon entering a folder, no message is
          selected (curmsg == -1), so deleting causes memory corruption
          (beyond bounds) ........

2007-05-07 22:28 +0000 [r63329]  Russell Bryant <russell@digium.com>

        * configs/res_pgsql.conf.sample (added),
          configs/extconfig.conf.sample, contrib/realtime_pgsql.sql
          (added): Add a sample configuration file and example tables for
          use with res_config_pgsql. (issue #9676, suretec)

2007-05-07 21:45 +0000 [r63283-63286]  Joshua Colp <jcolp@digium.com>

        * main/channel.c, include/asterisk/app.h, /, main/app.c: Merged
          revisions 63285 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2
          lines Properly handle what happens during a masquerade in
          relation to group counting. (issue #9657 reported by ramonpeek)
          ........

        * channels/chan_sip.c: Minor backport of revision 59083 in trunk.
          Don't queue an unhold frame up if the call was never on hold to
          begin with.

2007-05-07 20:05 +0000 [r63196-63254]  Olle Johansson <oej@edvina.net>

        * main/config.c: Don't remove configuration from memory just
          because one section failed.

        * /: Guess svnmerge doesn't handle files that move around. Blocking
          patch to ./config.c

2007-05-06 12:28 +0000 [r63152]  Olle Johansson <oej@edvina.net>

        * main/file.c: Stop the video stream when you stop playback of all
          streams for a call

2007-05-04 20:03 +0000 [r63099]  Jason Parker <jparker@digium.com>

        * res/res_jabber.c: Fix a crash when checking version attribute in
          an incoming XML caps element. Issue 9667, patch by phsultan.

2007-05-04 16:45 +0000 [r63047]  Pari Nannapaneni <paripurnachand@digium.com>

        * configs/manager.conf.sample: explanation for httptimeout in
          manager.conf

2007-05-03 16:44 +0000 [r62989]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_sip.c: Merged revisions 62987 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2
          lines When a peer is seeded or built tell the devicestate core to
          update it's status. This is easier then having chan_sip load
          before pbx_config. (issue #9658 reported by dlynes) ........

2007-05-03 16:38 +0000 [r62986]  Kevin P. Fleming <kpfleming@digium.com>

        * main/loader.c: improve loader a bit, by avoiding trying to
          initialize embedded modules twice and avoiding trying to load
          modules from disk when they have been loaded already during the
          'preload' pass (reported by blitzrage on IRC, patch by me)

2007-05-03 15:23 +0000 [r62942]  Russell Bryant <russell@digium.com>

        * main/channel.c: Fix YADB (Yet Another DTMF Bug) ((C) Russell
          Bryant, 2007, TM, Patent Pending). This set of changes came from
          a debugging session I had with Dwayne Hubbard. When he called
          into his home FXO, ran the Echo application, and pressed a digit,
          the digit would be echoed back and would never end. This is
          fixed, along with a couple other little improvements. * When
          chan_zap is in the middle of playing a digit to a channel, it
          feeds back null frames, not voice frames. So, I have modified
          ast_read to check the timing on emulated DTMF when it receives
          null frames, in addition to where it was doing this on voice
          frames. * Make a tweak to setting the duration on emulated DTMF
          digits. If there was no duration specified, it set it to be the
          minimum, instead of the default. * Instead of timing the emulated
          digits off of the number of samples in audio frames that pass
          through, just use time values. Now there is no code in this
          section that assumes 8kHz audio.

2007-05-03 14:41 +0000 [r62913]  Steve Murphy <murf@digium.com>

        * pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-test19
          (added), pbx/ael/ael-test/ael-test18/extensions.ael,
          pbx/ael/ael-test/ael-test19/extensions.ael (added),
          pbx/ael/ael-test/ael-test19 (added),
          pbx/ael/ael-test/ref.ael-test20 (added),
          pbx/ael/ael-test/ael-test20/extensions.ael (added),
          pbx/ael/ael-test/ael-test20 (added): updated the ael regressions
          to match what's in trunk

2007-05-03 14:36 +0000 [r62912]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h,
          channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
          channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged
          revisions 61357,61770,62885 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) |
          1 line some fixes for PMP Hold/Retrieve, it should work now, when
          briding=no ........ r61770 | crichter | 2007-04-24 15:50:05 +0200
          (Di, 24 Apr 2007) | 1 line added lock for sending messages to
          avoid double sending. shuffled some empty_chans after the
          cb_event calls, this avoids that a release_complete from a quite
          different call releases a fresh created setup by accident.
          ........ r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03
          Mai 2007) | 1 line fixed the problem that misdn_write did not
          return -1 when called with 0 samples in a frame this resultet in
          a deadlock in some circumstances, when the call ended because of
          a busy extension. added encoding of keypad. ........

2007-05-03 13:54 +0000 [r62883]  Steve Murphy <murf@digium.com>

        * pbx/ael/ael-test/ref.ael-test18 (added),
          pbx/ael/ael-test/ref.ael-vtest13,
          pbx/ael/ael-test/ael-test18/extensions.ael (added),
          pbx/ael/ael-test/ael-test18 (added),
          pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael.tab.c,
          pbx/ael/ael.y, pbx/ael/ael.tab.h, pbx/ael/ael-test/ref.ael-test7:
          These mods fix bug 9623, where an '@' in the eswitch contents
          causes a syntax error. I also updated the regressions.

2007-05-03 00:23 +0000 [r62797-62842]  Kevin P. Fleming <kpfleming@digium.com>

        * res/res_config_odbc.c, /: Merged revisions 62841 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r62841 | kpfleming | 2007-05-02 20:23:00 -0400 (Wed, 02
          May 2007) | 2 lines doh... initializing the pointer variable will
          work just a bit better ........

        * res/res_config_odbc.c, /: Merged revisions 62796 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r62796 | kpfleming | 2007-05-02 19:53:46 -0400 (Wed, 02
          May 2007) | 7 lines increase reliability and efficiency of static
          Realtime config loading via ODBC: don't request fields we aren't
          going to use don't request sorting on fields that are pointless
          to sort on explicitly request the fields we want, because we
          can't expect the database to always return them in the order they
          were created (reported by blitzrage in person (!), patch by me)
          ........

        * res/res_config_pgsql.c: improve static Realtime config loading
          from PostgreSQL: don't request sorting on fields that are
          pointless to sort on use ast_build_string() instead of snprintf()
          don't request the list of fieldnames that resulted from the query
          when we both knew what they were before we ran the query _AND_ we
          aren't going to do anything with them anyway (patch by me,
          inspired by blitzrage's bug report about res_config_odbc)

2007-05-02 22:59 +0000 [r62739-62789]  Russell Bryant <russell@digium.com>

        * main/channel.c: Merge changes from team/russell/inband_dtmf ...
          Fix some issues related to generating inband DTMF. There are two
          changes here: 1) The list of DTMF tones in the senddigit_begin()
          function explicitly specified 100ms of the tone followed by 100ms
          of silence. This really broke things with the way that Asterisk
          now wants complete control over when the digit begins and ends.
          So, regardless of what Asterisk really wanted to do, this was
          going to play out the tone at the length it wanted to. This
          caused various problems like DTMF translation to inband to be
          extremely unreliable. The list of tones has been changed so that
          the correct DTMF tone is played indefinitely until Asterisk tells
          it to stop. 2) ast_write() had to be modified to let a DTMF_END
          frame get processed even when a generator is present. This is how
          the tone will finally get stopped. (issues #8944, #9250, #9348,
          maybe others. Thanks to mdu113 from #8944 for the testing and
          feedback!)

        * main/manager.c: Backport the change that only went in to trunk
          that fixes the command manager action over http. (reported
          internally by pari and bkruse)

2007-05-02 20:46 +0000 [r62738]  Steve Murphy <murf@digium.com>

        * main/cdr.c, main/pbx.c, /: Merged revisions 62737 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r62737 | murf | 2007-05-02 14:10:32 -0600 (Wed, 02 May
          2007) | 1 line Some tweaks to satisfy CDR bug 8796, where being
          in 'h' extension louses up the dst field ........

2007-05-02 17:43 +0000 [r62692]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * /, channels/chan_iax2.c: Merged revisions 62691 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007)
          | 4 lines Issue 9638 - if a text frame is sent with no
          terminating NULL through a bridged IAX connection, the remote end
          will receive garbage characters tacked onto the end. ........

2007-05-02 17:10 +0000 [r62689]  Steve Murphy <murf@digium.com>

        * configs/extensions.conf.sample, main/channel.c, main/pbx.c,
          channels/chan_zap.c, cdr/cdr_radius.c: a)In chan_zap, set the
          clid, src fields in channel_alloc call. b)in the channel_alloc
          func, set the cid_num and name fields from the arglist[blush]. c)
          don't update the channel app & app data fields if you are in the
          'h' extension. d)the load_module func in cdr_radius needs to
          return DECLINE, SUCCESS.

2007-05-02 06:15 +0000 [r62624]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Don't unlock a channel that we already know
          does not exist (propably isue 8228)

2007-05-01 21:57 +0000 [r62548]  Russell Bryant <russell@digium.com>

        * /, res/res_features.c: Merged revisions 62547 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 May 2007) |
          4 lines Remove an unnecessary check that makes it so if you hang
          up after doing an attended transfer before the target extension
          answers the channel, the transfer is not successful. (issue
          #9338, patch by svanlund) ........

2007-05-01 21:34 +0000 [r62545]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_voicemail.c: Bug 9590 - Memory leaks around find_user()
          (found by rayjay, different fixes by me)

2007-05-01 16:26 +0000 [r62497]  Russell Bryant <russell@digium.com>

        * /, configs/indications.conf.sample: Merged revisions 62496 via
          svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01 May 2007) |
          3 lines Add indications.conf information for the Philippines.
          (issue #9525, reported and patched by loloski) ........

2007-04-30 15:58 +0000 [r62414-62419]  Russell Bryant <russell@digium.com>

        * channels/chan_zap.c, /: Merged revisions 62417 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) |
          4 lines This patch fixes an issue where depending on the cause
          code, when the network sends a PRI disconnect, the call may not
          be properly hung up. (issue #9588, reported and patched by
          softins) ........

        * include/asterisk/http.h, main/http.c: When serving dynamic
          content, include a Cache-Control header to instruct the browsers
          to not store the resulting content. (issue #9621, reported by
          Pari, patch by me)

2007-04-30 14:52 +0000 [r62371]  Jason Parker <jparker@digium.com>

        * configs/iax.conf.sample: Remove unused (and potentially
          confusing) jitterbuffer options from sample config.

2007-04-30 14:36 +0000 [r62369]  Joshua Colp <jcolp@digium.com>

        * main/asterisk.c, /: Merged revisions 62368 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r62368 | file | 2007-04-30 11:34:07 -0300 (Mon, 30 Apr 2007) | 2
          lines Update copyright notice. It's now the year 2007! ........

2007-04-29 05:50 +0000 [r62299-62331]  Russell Bryant <russell@digium.com>

        * channels/chan_zap.c: Fix a bug that made the "language" setting
          in zapata.conf not functional. (issue #9626, reported and fixed
          by sergee)

        * apps/app_meetme.c: Note that the "talker optimization" option
          will be enabled by default in 1.6

2007-04-27  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.4 released.

2007-04-27 21:10 +0000 [r62218]  Russell Bryant <russell@digium.com>

        * channels/chan_agent.c: Fix a weird problem where when a caller
          talking to someone sitting behind an agent channel sent a digit,
          the digit would be played to the agent for forever. This is
          because chan_agent always returned -1 from its send_digit_begin
          and _end callbacks. This non-zero return value indicates to the
          Asterisk core that it would like an inband DTMF generator put on
          the channel. However, this is the wrong thing to do. It should
          *always* return 0, instead. When the digit begin and end
          functions are called on the proxied channel, the underlying
          channel will indicate whether inband DTMF is needed or not, and
          the generator will be put on that one, and not the Agent channel.
          (issue #9615, #9616, reported by jiddings and BigJimmy, and fixed
          by me)

2007-04-27 16:17 +0000 [r62174]  Jason Parker <jparker@digium.com>

        * /, codecs/codec_zap.c: Merged revisions 62173 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3
          lines This transcoder message needn't be a NOTICE. I've seen it
          cause confusion more than a few times. ........

2007-04-27 16:14 +0000 [r62171]  Russell Bryant <russell@digium.com>

        * main/pbx.c: If no variables were passed into
          pbx_substitute_variables_helper_full(), then don't even bother
          creating a temporary bogus channel, since that is only for
          allowing certain functions to operate on the variables as if they
          were on a channel. Most importantly, this fixes a crash. (issue
          #9613, reported by callguy, fixed by me)

2007-04-27 14:04 +0000 [r62095-62137]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Merged revisions 62126 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4
          lines Issue #7351 - SIP Cancel fails due to the wrong contact
          uri. Reported by PPYY, failed to fix by OEJ final fix by wojtekka
          - THANKS!!!! THis was a hard one to catch. ........

        * channels/chan_zap.c, main/manager.c: Issue #9608 - fix some
          annoying DEBUG messages not controlled by option_debug (DEA).
          Thanks!

2007-04-26 16:33 +0000 [r61959-62038]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_iax2.c: Merged revisions 62037 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2
          lines Revert previous fix for when the IAX2 channel goes funky
          (that's the technical term). This is causing legit calls to be
          prematurely hung up. (issue #9600 reported by justdave) ........

        * main/channel.c: Missed an ast_app_group_discard during merge.
          Thanks blitzrage!

        * res/res_monitor.c: Don't always say that the channel is being
          paused if it is actually being unpaused in the Manager ack
          message. (reported by jsmith in #asterisk-bugs)

        * main/config.c, /: Merged revisions 61958 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r61958 | file | 2007-04-25 21:25:03 -0400 (Wed, 25 Apr 2007) | 2
          lines Don't count failed include attempts against the
          configuration include level. (issue #9593 reported by mostyn)
          ........

2007-04-25 22:29 +0000 [r61914]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_zap.c, /: Merged revisions 61913 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007)
          | 2 lines handle a very bizarre race condition with channels
          being redirected before a simple switch can be started on them
          (issue #9286) ........

2007-04-25 21:59 +0000 [r61863-61870]  Russell Bryant <russell@digium.com>

        * /, channels/chan_iax2.c: Merged revisions 61866 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) |
          2 lines If the callerid= option is specified, but empty, clear
          any previous data. ........

        * /, channels/chan_iax2.c: Merged revisions 61862 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) |
          2 lines Ensure that callerid settings are reset on a reload.
          ........

2007-04-25 19:21 +0000 [r61805]  Joshua Colp <jcolp@digium.com>

        * main/cli.c, main/channel.c, include/asterisk/app.h,
          funcs/func_groupcount.c, /, main/app.c: Merged revisions 61804
          via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2
          lines Merge rewritten group counting support. No more storing
          data on the variable list of the channels. That was bad, mmmk?
          (issue #7497 reported by sabbathbh) ........

2007-04-25 16:22 +0000 [r61799]  Russell Bryant <russell@digium.com>

        * channels/chan_zap.c, /: Merged revisions 61798 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) |
          3 lines Fix a typo where cid_num got copied instead of cid_ani.
          (issue #9587, reported and patched by xrg) ........

2007-04-24  Russell Bryant <russell@digium.com>

        * Asterisk 1.4.3 released.

2007-04-24 21:34 +0000 [r61781-61787]  Russell Bryant <russell@digium.com>

        * main/manager.c, /: Merged revisions 61786 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24 Apr 2007) |
          4 lines Don't crash if a manager connection provides a username
          that exists in manager.conf but does not have a password, and
          also requests MD5 authentication. (ASA-2007-012) ........

        * main/channel.c, include/asterisk/channel.h: Improve DTMF handling
          in ast_read() even more in response to a discussion on the
          asterisk-dev mailing list. I changed the enforced minimum length
          of a digit from 100ms to 80ms. Furthermore, I made it now enforce
          a gap of 45ms in between digits. These values are not
          configurable in a configuration file right now, but they can be
          easily changed near the top of main/channel.c.

2007-04-24 18:43 +0000 [r61779]  Dwayne M. Hubbard <dhubbard@digium.com>

        * channels/chan_zap.c, /: Merged revisions 61777 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r61777 | dhubbard | 2007-04-24 13:20:31 -0500 (Tue, 24 Apr 2007)
          | 1 line removed #if 0 block from chan_phone, chan_zap, and
          chan_modem restart_monitor() ........

2007-04-24 16:16 +0000 [r61774]  Russell Bryant <russell@digium.com>

        * main/dial.c: Add a few more state changes in
          handle_frame_ownerless() so that the SLA code will get notified
          of these changes even when an owner channel is not provided. This
          isn't from a specific bug report, it's just something I noticed
          while poking around.

2007-04-24 16:07 +0000 [r61772]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_sip.c: Merged revisions 61771 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2
          lines Allow RFC2833 to be sent in the response SDP when an INVITE
          comes in without SDP. (issue #9546 reported by mcrawford)
          ........

2007-04-23 18:17 +0000 [r61763-61765]  Russell Bryant <russell@digium.com>

        * main/pbx.c: Some dialplan functions, such as CUT(), expect to
          operate on variables on a channel. So, this little hack lets them
          work in places where a channel doesn't exist, such as within
          DUNDi configuration. (issue #9465, reported and patched by
          Corydon76, testing by blitzrage)

        * main/channel.c: Ensure that digits passing through Asterisk have
          a reasonable minimum length. It is currently 100 ms. If someone
          thinks this should be different, feel free to speak up. (related
          to issues #8944, #9250, and #9348)

2007-04-20 21:35 +0000 [r61705-61707]  Jason Parker <jparker@digium.com>

        * main/rtp.c: Avoid invalid seqno cycling detection. Per comment
          from Dave Troy: This adds back in some simple typecasting I had
          in an earlier version which I realize now may be breaking things.
          Issue #9554.

        * main/loader.c, /: Merged revisions 61704 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r61704 | qwell | 2007-04-20 16:14:27 -0500 (Fri, 20 Apr 2007) | 4
          lines Fix an issue that I noticed while looking over issue 9571.
          The reload timestamp was getting set after reloading the built-in
          stuff, and before the modules. ........

2007-04-20 20:42 +0000 [r61697]  Russell Bryant <russell@digium.com>

        * main/rtp.c: Remove a stray debug message introduced by a recent
          commit.

2007-04-20 19:51 +0000 [r61694]  Jason Parker <jparker@digium.com>

        * /, apps/app_queue.c: Merged revisions 61692 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r61692 | qwell | 2007-04-20 14:49:54 -0500 (Fri, 20 Apr 2007) | 5
          lines If the '* to hangup' option is not enabled, we don't need
          to disable * as a valid exit key. If it was enabled, this
          statement would've never been checked in the first place. Issue
          #9552 ........

2007-04-20 18:19 +0000 [r61690]  Russell Bryant <russell@digium.com>

        * main/config.c, apps/app_voicemail.c, main/manager.c,
          include/asterisk/config.h: Fix the UpdateConfig manager action to
          properly treat "variables" and "objects" differently (a=b versus
          a=>b). (issue #9568, reported by pari, patch by me)

2007-04-19 08:37 +0000 [r61686]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Merged revisions 61685 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r61685 | oej | 2007-04-19 09:56:21 +0200 (Thu, 19 Apr 2007) | 3
          lines Send NOTIFY to Contact: in SUBSCRIBE - as reported by
          Intertex and Citel. Fixed during SIPit 20 in Antwerp. ........

2007-04-19 04:36 +0000 [r61681-61683]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * main/manager.c: Bug 9557 - simple reason why reading a function
          always returned NULL

        * funcs/func_callerid.c, funcs/func_language.c, funcs/func_moh.c,
          funcs/func_groupcount.c, /, funcs/func_timeout.c,
          funcs/func_cdr.c: Merged revisions 61680 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18 Apr 2007)
          | 5 lines Bug 9557 - Specifying the GetVar AMI action without a
          Channel parameter can cause Asterisk to crash. The reason this
          needs to be fixed in the functions instead of in AMI is because
          Channel can legitimately be NULL, such as when retrieving global
          variables. ........

2007-04-18 22:10 +0000 [r61678]  Kevin P. Fleming <kpfleming@digium.com>

        * sounds/Makefile: allow external build systems to extract the
          required sound file versions

2007-04-18 20:46 +0000 [r61674-61676]  Olle Johansson <oej@edvina.net>

        * main/rtp.c: Clean upp formatting, add some doxygen stuff while
          we're in cleaning mode... Thanks Kevin!

        * main/rtp.c: Issue #9554 - Improve RTCP (Dave Troy)

2007-04-16 14:47 +0000 [r61664-61666]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: #9483, half of patch by twilson to solve 302
          redirect issues

        * /: Blocking AstHoloPatch from 1.2

2007-04-13 21:17 +0000 [r61658]  Steve Murphy <murf@digium.com>

        * main/cdr.c: This is a fix to the way CDR merge handles the data
          that results from ForkCDR.

2007-04-13 19:17 +0000 [r61648-61656]  Joshua Colp <jcolp@digium.com>

        * apps/app_dial.c, /: Merged revisions 61655 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2
          lines Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves
          the same as OUTBOUND_GROUP except it will get unset after use so
          it won't get accidentally inherited. (issue #BE-140) ........

        * apps/app_speech_utils.c: Do not bother looking for a result if
          none are present.

        * channels/chan_sip.c: For those very verbose SIP implementations
          that attach tons of info to the Contact header... let's increase
          our variable sizes. (issue #9535 reported by jeffg)

2007-04-13 17:10 +0000 [r61645]  Russell Bryant <russell@digium.com>

        * apps/app_voicemail.c: Eliminate a compiler warning with
          ODBC_STORAGE enabled so that it will build under dev-mode.

2007-04-13 17:01 +0000 [r61644]  Steve Murphy <murf@digium.com>

        * channels/chan_oss.c: A fix for chan_oss that resulted from the
          CDR changes; it helps to use the right info.

2007-04-13 16:32 +0000 [r61641]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Don't assume the callid of a dialog will be
          set, as in some circumstances it may not. (issue #9534 reported
          by tecnoxarxa)

2007-04-11 16:05 +0000 [r61477]  Russell Bryant <russell@digium.com>

        * /, channels/chan_sip.c: Merged revisions 61476 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) |
          5 lines If someone sets the "useragent" option in sip.conf to be
          empty, then don't add the User-Agent header at all. It is an
          optional header, anyway. Also, the bug report says that some of
          Japan's SIP providers don't allow it for some weird reason.
          (issue #9488, reported by makoto, fixed by me) ........

2007-04-11 15:39 +0000 [r61443]  Nadi Sarrar <ns@beronet.com>

        * channels/chan_misdn.c: Don't export AOCD variables on
          misdn_hangup anymore, this was mainly a fix for trunk..

2007-04-11 15:09 +0000 [r61377-61427]  Russell Bryant <russell@digium.com>

        * /, channels/chan_sip.c: Merged revisions 61426 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) |
          6 lines Fix a bug with switching between host=dynamic and using
          specific hosts for peers. The code would only reset the peer's
          address when it is dynamic if it was a new peer structure. Now,
          it will also reset the address if it was already in the peer
          list, but before the reload, it was not dynamic. (issue #9515,
          reported by caio1982, fixed by me) ........

        * main/http.c: Add "svgz" to the mimetypes table. (issue #9510,
          bkruse) In passing, constify the elements of the mimetypes table.

        * /, channels/chan_sip.c: Merged revisions 61376 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) |
          5 lines Remove the attempt at reporting configuration errors in
          sip.conf. This can cause a bunch of improper messages when using
          realtime. I give up. As oej tried to convince me when I put this
          in, there is just no easy way to do it. (inspired by a message on
          the -dev list) ........

2007-04-11 13:40 +0000 [r61342-61373]  Nadi Sarrar <ns@beronet.com>

        * channels/chan_misdn.c: Export AOCD variables on misdn_hangup.

        * channels/chan_misdn.c: Ignore facility messages in case we don't
          have a corresponding channel object.

        * channels/chan_misdn.c: AOCD's are now exported to asterisk
          channel variables.

2007-04-10 16:05 +0000 [r61220]  Russell Bryant <russell@digium.com>

        * main/Makefile, main/http.c, main/minimime (removed): File upload
          support was added to solve some needs for the Asterisk GUI.
          However, after much discussion, it has been decided that adding
          this to 1.4 is not in the best interests of the project. It has
          been removed here, but will remain in trunk.

2007-04-10 12:43 +0000 [r61183]  Nadi Sarrar <ns@beronet.com>

        * channels/misdn_config.c, /: Merged revisions 61170 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr
          2007) | 2 lines msns config parameter defaults to '*' ........

2007-04-10 05:18 +0000 [r61136]  Steve Murphy <murf@digium.com>

        * apps/app_cdr.c, main/cdr.c, res/res_features.c: Finished up a
          previous fix to overcome a compiler warning; the app NoCDR() has
          been updated to mark the channel CDR as POST_DISABLED instead of
          destroying the CDR; this way its flags are propagated thru a
          bridge and the CDR is actually dropped. The cases where only one
          channel in a bridge has a CDR was cleaned up.

2007-04-09 19:58 +0000 [r61072]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Merged revisions 61038 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3
          lines - Don't send ActionID before Response: header. - Don't use
          a blank in an AMI header ........

2007-04-09 19:55 +0000 [r61062-61070]  Kevin P. Fleming <kpfleming@digium.com>

        * main/minimime/mm_envelope.c, res/res_features.c: fix up some
          warnings found using --enable-dev-mode

        * main/minimime/Doxyfile (removed),
          main/minimime/tests/messages/CVS (removed),
          main/minimime/tests/CVS (removed): remove some more stuff we
          don't need

2007-04-09 19:41 +0000 [r61042-61044]  Russell Bryant <russell@digium.com>

        * main/minimime/test (removed): Remove another directory that
          should no longer be there

        * main/minimime/Make.conf (removed), main/minimime/mytest_files
          (removed), main/minimime/.cvsignore (removed), main/minimime/sys
          (removed), main/minimime/mm-docs (removed): Remove various files
          that I thought I already removed.

2007-04-09 19:05 +0000 [r61022]  Jason Parker <jparker@digium.com>

        * apps/app_queue.c: Use the appropriate interface name with
          COMPLETECALLER. Issue 9395.

2007-04-09 18:32 +0000 [r60989]  Steve Murphy <murf@digium.com>

        * channels/chan_oss.c, main/channel.c, main/cdr.c,
          channels/chan_phone.c, channels/chan_misdn.c,
          channels/chan_skinny.c, channels/chan_features.c,
          channels/chan_h323.c, channels/chan_alsa.c, channels/chan_nbs.c,
          channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c,
          channels/chan_vpb.cc, channels/chan_local.c, channels/chan_zap.c,
          channels/chan_sip.c, res/res_features.c, channels/chan_agent.c,
          include/asterisk/channel.h, channels/chan_gtalk.c,
          channels/chan_iax2.c: This is a big improvement over the current
          CDR fixes. It may still need refinement, but this won't have as
          many folks bothered.

2007-04-09 18:02 +0000 [r60984]  Olle Johansson <oej@edvina.net>

        * res/res_jabber.c: Add final new line after JabberEvent

2007-04-09 17:22 +0000 [r60936]  Jason Parker <jparker@digium.com>

        * /, apps/app_directory.c: Merged revisions 60935 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r60935 | qwell | 2007-04-09 12:22:15 -0500 (Mon, 09 Apr 2007) | 5
          lines Allow matching on names shorter than 3 chars. This also
          fixes the case where somebody wants to match on less then 3
          chars. Issue 9071 ........

2007-04-09 03:01 +0000 [r60847-60850]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * main/asterisk.c, include/asterisk.h, /: Merged revisions 60849
          via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007)
          | 2 lines Don't check for error when lowering priority (according
          to the manpage, it should never happen anyway). It might could
          happen, though, if another thread messed with the priority, so
          safeguard against that (reported via -dev list). ........

        * channels/chan_local.c, /: Merged revisions 60846 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08
          Apr 2007) | 2 lines Bug 9505 - If the return value for
          local_queue_frame is set, then p->lock is no longer valid.
          ........

2007-04-09 01:03 +0000 [r60762-60798]  Joshua Colp <jcolp@digium.com>

        * apps/app_dial.c, /: Merged revisions 60797 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2
          lines When calling a device that then forwards us elsewhere... we
          have to make our channels compatible if it is the only channel
          being dialed. (issue #9445 reported by marcelbarbulescu) ........

        * apps/app_queue.c: Allow app_queue to use MONITOR_EXEC even if
          MONITOR_OPTIONS is not set. (issue #9495 reported by cduffy)

2007-04-08 14:14 +0000 [r60661-60713]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * /, apps/app_macro.c: Merged revisions 60711 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r60711 | tilghman | 2007-04-08 09:00:22 -0500 (Sun, 08 Apr 2007)
          | 2 lines Gosub called within a Macro resets the arguments
          improperly and causes general weirdness. (Issue 8329) ........

        * main/http.c: Fix --enable-dev-mode

        * channels/chan_oss.c: Off by one error, resulting in a crash
          (Issue 9500)

        * /, main/file.c: Merged revisions 60660 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r60660 | tilghman | 2007-04-07 20:39:25 -0500 (Sat, 07 Apr 2007)
          | 2 lines Bug 9486 - memory leak when opening a filestream
          ........

2007-04-06 20:58 +0000 [r60603]  Russell Bryant <russell@digium.com>

        * main/minimime/sys/mm_queue.h, main/minimime/Doxyfile,
          main/minimime/mimeparser.yy.c, main/minimime/minimime.c,
          main/manager.c, main/minimime/mm_mimepart.c,
          main/minimime/test.sh, configure, include/asterisk/compat.h,
          main/strcompat.c, main/minimime/mm_internal.h, main/http.c,
          main/minimime/tests/parse.c, main/minimime/mm_base64.c,
          main/minimime/mm_mimeutil.c, main/minimime/mm.h,
          main/minimime/tests, main/minimime/mm_header.c,
          main/minimime/mm_error.c, main/Makefile,
          main/minimime/mm_codecs.c, main/minimime/mm_param.c,
          configure.ac, main/minimime/Makefile, main/minimime/mm_init.c,
          include/asterisk/manager.h, main/minimime/strlcpy.c,
          configs/http.conf.sample, main/minimime/mm_parse.c,
          main/minimime/tests/create.c, main/minimime/mm_contenttype.c,
          main/minimime/mm_util.c, main/minimime/mm_envelope.c,
          main/minimime/tests/messages/test1.txt, main/minimime/mm_mem.c,
          main/minimime/tests/messages/test2.txt,
          main/minimime/tests/messages/test3.txt,
          main/minimime/mimeparser.h, main/minimime/mimeparser.tab.c,
          main/minimime/tests/messages/test4.txt,
          main/minimime/tests/messages/test5.txt, main/minimime/mm_util.h,
          main/minimime/tests/messages/test6.txt, main/minimime/strlcat.c,
          main/minimime/mm_mem.h, main/minimime/tests/messages/test7.txt,
          main/minimime/mimeparser.l, main/minimime/mm_context.c,
          main/minimime/mimeparser.tab.h, main/minimime (added),
          main/minimime/mm_warnings.c, main/minimime/mm_queue.h,
          main/minimime/tests/messages, include/asterisk/autoconfig.h.in,
          main/minimime/mimeparser.y, Makefile.moddir_rules,
          main/minimime/sys, main/minimime/tests/Makefile: To be able to
          achieve the things that we would like to achieve with the
          Asterisk GUI project, we need a fully functional HTTP interface
          with access to the Asterisk manager interface. One of the things
          that was intended to be a part of this system, but was never
          actually implemented, was the ability for the GUI to be able to
          upload files to Asterisk. So, this commit adds this in the most
          minimally invasive way that we could come up with. A lot of work
          on minimime was done by Steve Murphy. He fixed a lot of bugs in
          the parser, and updated it to be thread-safe. The ability to
          check permissions of active manager sessions was added by Dwayne
          Hubbard. Then, hacking this all together and do doing the
          modifications necessary to the HTTP interface was done by me.

2007-04-06 20:32 +0000 [r60568-60572]  Dwayne M. Hubbard <dhubbard@digium.com>

        * UPGRADE.txt: clarified a sentence in the format_wav section

        * UPGRADE.txt: updated UPGRADE.txt with format_wav GAIN change and
          plan to remove GAIN code from trunk

2007-04-06 19:50 +0000 [r60521-60565]  Russell Bryant <russell@digium.com>

        * apps/app_meetme.c: When a station picks up a trunk that was on
          hold, make the hints reflect that nobody has the trunk on hold
          anymore.

        * apps/app_meetme.c: Fix a few problems with SLA. (issue #9459,
          reported by francesco_r, fixed by me) * The original behavior was
          that if one station put a call on hold, another one picked it up,
          and then hung up, the code would still consider the call on hold
          by the first station, so the trunk would not be hung up. However,
          to better comply with what most people seem to expect it to
          behave, it will now hang up the trunk. * Fix a problem with
          "barge=no". This was only intended to prevent people from joining
          calls that are in progress. However, it also prevented other
          people from picking up a call that was on hold. This has been
          fixed. * When there are no active stations on a trunk and it is
          on hold, the code now indicates the HOLD and UNHOLD conditions to
          the trunk channel. This allows music on hold to be played to the
          trunk when it is on hold.

2007-04-06 18:21 +0000 [r60459-60485]  Matt Frederickson <creslin@digium.com>

        * channels/chan_zap.c: Make sure we check the faxdetect option
          before doing fax processing

        * channels/chan_zap.c, /: Merged revisions 60456 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2
          lines There should only be one code path for doing DTMF
          conditionals on channels. This fixes it. ........

2007-04-06 14:49 +0000 [r60399]  Kevin P. Fleming <kpfleming@digium.com>

        * /, codecs/codec_zap.c: Merged revisions 60398 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r60398 | kpfleming | 2007-04-06 09:41:37 -0500 (Fri, 06 Apr 2007)
          | 2 lines remove undocumented 'cardsmode' parameter and stop
          searching for transcoders during reload() ........

2007-04-06 01:14 +0000 [r60361]  Joshua Colp <jcolp@digium.com>

        * res/res_speech.c, apps/app_speech_utils.c,
          include/asterisk/speech.h: Add support for returning different
          types of results (ie: NBest).

2007-04-05 22:58 +0000 [r60325]  Dwayne M. Hubbard <dhubbard@digium.com>

        * formats/format_wav.c: modified default GAIN for issue 5823,
          thanks jrwalliker

2007-04-05 22:35 +0000 [r60323]  Steve Murphy <murf@digium.com>

        * configs/cdr_custom.conf.sample, configs/cdr.conf.sample: Added
          some clarification to the example configs for CDRs, on how to
          select a backend. Also, made cdr-csv the default if you 'make
          samples', and no other changes.

2007-04-05 16:10 +0000 [r60268]  Jason Parker <jparker@digium.com>

        * apps/app_voicemail.c, /: Merged revisions 60267 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r60267 | qwell | 2007-04-05 11:09:41 -0500 (Thu, 05 Apr 2007) | 5
          lines Just because we can't find the voicemail configuration
          file, doesn't mean that the module failed to load. The user could
          be using realtime. Issue #9473 ........

2007-04-05 15:47 +0000 [r60265]  Russell Bryant <russell@digium.com>

        * main/http.c: Add the MIME type for gif by request from Pari

2007-04-05 12:55 +0000 [r60214]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_sip.c: Merged revisions 60213 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2
          lines Only unlock our pvt and net locks if we are actually going
          to try to lock the owner again. (issue #9472 reported by zoa)
          ........

2007-04-04 17:40 +0000 [r60013-60137]  Russell Bryant <russell@digium.com>

        * main/manager.c, /: Merged revisions 60134 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04 Apr 2007) |
          6 lines It is valid to redirect channels via the manager
          interface that are not in the UP state. Instead of checking for
          that to prevent to ensure a dead channel doesn't get redirected,
          just use the ast_check_hangup() API call. (issue #9457, reported
          by Callmewind, patch by me) (related to issue #8977) ........

        * channels/chan_sip.c: Add a Content-Length of 0 to the response
          built by transmit_response_with_unsupported(). (issue #9454,
          reported by makoto, fixed by me)

        * /, channels/chan_sip.c: Merged revisions 60083 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) |
          4 lines Fix the return value of handle_common_options() so that
          it always properly indicates whether it handled the option or
          not. (issue #9455, reported by Netview, fixed by me) ........

        * apps/app_meetme.c: Fix a problem where if a trunk was hung up
          while it was on hold, all of the hints would reflect the line
          still on hold, even though it should reflect that it is back to
          not in use. (issue #9459, reported by francesco_r, fixed by me)

2007-04-03 19:40 +0000 [r59963]  Joshua Colp <jcolp@digium.com>

        * apps/app_speech_utils.c: Don't clash when a person both speaks
          and uses DTMF.

2007-04-03 19:16 +0000 [r59853-59939]  Russell Bryant <russell@digium.com>

        * /, channels/chan_sip.c: Merged revisions 59938 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) |
          4 lines Don't attempt to report configuration errors in
          build_user(). oej pointed out that for a "friend" entry, this
          won't work, because all user options are valid for peers, but not
          the other way around. ........

        * /, channels/chan_sip.c: Merged revisions 59916 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) |
          3 lines Make chan_sip report when it encounters an unknown
          option. (issue #9440, reported by nightcrawler) ........

        * /, main/app.c: Merged revisions 59886 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03 Apr 2007) |
          5 lines When doing a built-in blind or attended transfer, restore
          the ability to use '#' to terminate the number and immediately do
          the transfer instead of having to dial the number and just wait
          for the feature digit timeout. (issue #8366, xueliangliang)
          ........

        * Makefile: Ensure that menuselect gets executed in dependency
          check mode every time you run make.

2007-04-03 11:02 +0000 [r59804]  Nadi Sarrar <ns@beronet.com>

        * channels/misdn_config.c, /, channels/misdn/chan_misdn_config.h:
          Merged revisions 59788,59803 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2
          lines Use the new sysfs way of mISDN 1.2 to check if a port is NT
          or not. ........ r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di,
          03 Apr 2007) | 2 lines ptp is the 5th bit, not the 4th. ........

2007-04-03 07:20 +0000 [r59774]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/misdn_config.c,
          channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h:
          Merged revisions 59623-59624,59639 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) |
          1 line we can now make 30 channels on a PRI (before we forgot
          chan 31..) ........ r59624 | crichter | 2007-04-02 09:25:54 +0200
          (Mo, 02 Apr 2007) | 1 line don't be verbose if no need ........
          r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) |
          1 line added option which allows us to accept incoming SETUP
          Messages without automatically sending Proceeding or Setup
          Acknowledge, this is useful with some broken switches and if you
          want to Release incoming calls without previously having
          acknowledged them. The new option is
          noautorespond_on_setup=yes|no default is no, so we don't break
          the existing behaviour ........

2007-04-02 18:58 +0000 [r59724]  Joshua Colp <jcolp@digium.com>

        * apps/app_voicemail.c, /: Merged revisions 59723 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r59723 | file | 2007-04-02 14:55:25 -0400 (Mon, 02 Apr 2007) | 2
          lines Increase the maximum size for a string of mailboxes to
          1024. (issue #9270 reported by rtucker) ........

2007-04-02 17:31 +0000 [r59688]  Steve Murphy <murf@digium.com>

        * pbx/pbx_ael.c: continue in for-loop should go to the incrementer,
          not the test. As per 9435, thanks to marcelbarbulescu

2007-04-02 15:39 +0000 [r59654]  Russell Bryant <russell@digium.com>

        * main/netsock.c, /: Merged revisions 59608 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01 Apr 2007) |
          6 lines Add the SO_REUSEADDR flag to sockets handled by netsock.
          This is needed by the patch that went in for issue 7874.
          chan_iax2 needs to be able to create socket that is lisetning on
          INADDR_ANY, but also be able to bind sockets to specific
          addresses. (Thanks to Stevenson on the asterisk-dev mailing list
          for explaining why this flag was needed.) ........

2007-03-30 22:50 +0000 [r59573]  Jason Parker <jparker@digium.com>

        * configure, main/Makefile, acinclude.m4: Add linux-uclibc host
          arch..."thingy". Sorry, I don't know what it's called...

2007-03-30 17:51 +0000 [r59452-59522]  Steve Murphy <murf@digium.com>

        * main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c,
          include/asterisk/cdr.h: several changes via kpflemings review

        * main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c,
          include/asterisk/cdr.h: These mods fix CDR issues from 8221,
          8593, 8680, 8743, and perhaps others. Mainly with CDRs generated
          from transfer situations.

        * configs/extensions.conf.sample: A small clarification to keep
          bugs from being filed, and confusion from rising, if
          clearglobalvars is set, and globals are set in the AEL file.
          (9419)

2007-03-29 17:43 +0000 [r59363]  Russell Bryant <russell@digium.com>

        * res/res_jabber.c: When building a response to a subscription, the
          "from" must be the full Jabber ID. This fixes some problems where
          jabber users are not able to add their Asterisk account to their
          user list, since they are unable to get Asterisk to approve their
          subscription. (issue #8210, reported by caspy, and verified by
          bradtem)

2007-03-29 17:38 +0000 [r59361]  Joshua Colp <jcolp@digium.com>

        * /, apps/app_meetme.c: Merged revisions 59360 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2
          lines Keep a global array of variables indicating whether certain
          conference rooms are in use. This ensures that two people going
          into a new dynamic conference when the 'e' option is set don't go
          into the same conference room. (issue #8835 reported by eliel)
          ........

2007-03-29 17:17 +0000 [r59304-59358]  Russell Bryant <russell@digium.com>

        * main/rtp.c, /: Merged revisions 59357 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) |
          5 lines If an error occurs when reading from an RTP socket, and
          the error code does not indicate that we should try again, then
          return NULL instead of a "null frame". This will prevent Asterisk
          from trying over and over again, and eventually causing the
          system to crash. (issue #8285, john) ........

        * channels/chan_iax2.c: When the IAX2 read callback gets called,
          return NULL instead of a "null frame". This will cause Asterisk
          to hangup the call instead of keep trying whatever it was doing.
          Under normal conditions, this function would *never* be called.
          However, the author of this patch says an error will occur that
          will cause it to get called every 100 thousand calls or so. When
          this does happen, it puts the channel in a loop that eventually
          brings down the system. So, hangup up the call is certainly a
          better alternative. (issue #8286, john)

        * Makefile: Export the GTK2 library and include information to sub
          Makefiles.

2007-03-29 16:07 +0000 [r59300-59302]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * /, cdr/cdr_odbc.c: Merged revisions 59301 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r59301 | tilghman | 2007-03-29 11:04:46 -0500 (Thu, 29 Mar 2007)
          | 3 lines Issue 9415 - No point to getting a diagnostic field if
          we aren't doing anything with the information. (Plus, it tends to
          crash the Postgres ODBC driver.) ........

2007-03-28 03:38 +0000 [r59281-59289]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * res/res_odbc.c: Another crash that I thought we had fixed already
          - Issue 9396

        * apps/app_voicemail.c, /: Merged revisions 59283 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r59283 | tilghman | 2007-03-27 18:36:49 -0500 (Tue, 27 Mar 2007)
          | 2 lines Oops ........

        * apps/app_voicemail.c, /: Merged revisions 59280 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r59280 | tilghman | 2007-03-27 18:31:20 -0500 (Tue, 27 Mar 2007)
          | 2 lines Fix a few remaining bad mmap(2) return values ........

2007-03-27 23:20 +0000 [r59262-59278]  Russell Bryant <russell@digium.com>

        * /, apps/app_directory.c: Merged revisions 59277 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r59277 | russell | 2007-03-27 18:19:41 -0500 (Tue, 27 Mar 2007) |
          3 lines Fix the check of the return value from mmap(). Thanks to
          Corydon for catching this one. ........

        * apps/app_directory.c: Fix app_directory to actually compile with
          ODBC_STORAGE, and update the code to the latest res_odbc API.

        * apps/Makefile: Fix app_directory when ODBC_STORAGE is being used.
          The Makefile did not properly ensure that this information got
          copied from what was selected for app_voicemail. (issue #9224)

        * channels/chan_sip.c: Fix the check that ensures that the CHANNEL
          function's first argument is "rtpqos". Thanks, Corydon. :)

2007-03-27 18:16 +0000 [r59261]  Steve Murphy <murf@digium.com>

        * pbx/pbx_ael.c: via 9373 (duplicate context in AEL crashes
          asterisk), kpfleming pointed on asterisk-dev, that DECLINE in
          this case the proper thing to do. This change now has it doing
          the proper thing.

2007-03-27 18:05 +0000 [r59256-59259]  Russell Bryant <russell@digium.com>

        * /, channels/chan_iax2.c: Merged revisions 59258 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 Mar 2007) |
          4 lines Fix the use of the "sourceaddress" option when "bindaddr"
          is set to 0.0.0.0 instead of having each interface explicitly
          listed. (issue #7874, patch by stevens) ........

        * channels/chan_sip.c, funcs/func_channel.c: Convert the RTPQOS
          function to just be additional parameter of the CHANNEL function.
          This way, it will be possible for other RTP based channel drivers
          to expose this information in the future.

2007-03-27 15:00 +0000 [r59254]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c, /: Merged revisions 59252 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27
          Mär 2007) | 1 line fixed #9355 ........

2007-03-26 21:45 +0000 [r59230]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * channels/chan_sip.c: Oops, this should be case insensitive

2007-03-26 21:41 +0000 [r59228]  Steve Murphy <murf@digium.com>

        * pbx/pbx_ael.c: fix for 9373 (duplicate context in AEL crashes
          asterisk). I turned a duplicate context from a WARNING to an
          ERROR. Now you get a module load failure, and asterisk just
          exits. That's better than a crash, right\?

2007-03-26 21:37 +0000 [r59227]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * channels/chan_sip.c: Change this to a single dp function to make
          oej happy.

2007-03-26 20:06 +0000 [r59225]  Steve Murphy <murf@digium.com>

        * main/config.c: Fix for 9257; by eliminating the globals in
          main/config.c, we make it thread-safe, which is a minimum
          requirement.

2007-03-26 19:34 +0000 [r59223]  Joshua Colp <jcolp@digium.com>

        * apps/app_speech_utils.c: Add ability to specify no timeout. This
          means as soon as the prompt is done playing it moves on to the
          next priority.

2007-03-26 18:33 +0000 [r59215-59217]  Russell Bryant <russell@digium.com>

        * apps/app_voicemail.c: Somehow the code for building the email for
          voicemail got out of sync. This change makes a few tweaks to get
          1.4 in sync with trunk. (issue #9301)

        * apps/app_meetme.c: Fix some codec negotiation problems when
          CallerID support is not enabled in SLA. (issue #9308, reported by
          twilson)

2007-03-26 18:13 +0000 [r59213]  Joshua Colp <jcolp@digium.com>

        * apps/app_speech_utils.c: Make SpeechBackground obey the digit
          timeout value.

2007-03-26 17:53 +0000 [r59207-59209]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: Rename the new dialplan functions to match
          the variable name

        * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: The
          AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in
          some because they get set in sip_hangup. So, there are common
          situations where the variables will not be available in the
          dialplan at all. So, this patch provides an alternate method for
          getting to this information by introducing AUDIORTPQOS and
          VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76,
          with some testing by blitzrage)

2007-03-26 17:38 +0000 [r59206]  Steve Murphy <murf@digium.com>

        * main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c,
          pbx/ael/ael.flex: A fix for the flex input files, DONT_COMPILE,
          and STANDALONE_AEL

2007-03-26 15:25 +0000 [r59202]  Nadi Sarrar <ns@beronet.com>

        * channels/misdn/isdn_lib.c, channels/misdn_config.c,
          channels/misdn/isdn_lib.h, channels/chan_misdn.c, configure,
          include/asterisk/autoconfig.h.in, channels/misdn/Makefile,
          channels/misdn/chan_misdn_config.h, configure.ac: * mISDN >= 1.2
          provides a dsp pipeline for i.e. echo cancellation modules, make
          chan_misdn use it. * add a check for linux/mISDNdsp.h to
          configure.ac and update the autogenerated files: 'configure',
          'autoconfig.h.in' (the 'configure' script was not in sync with
          the latest configure.ac, so the diff is a bit bigger than
          expected).

2007-03-26 15:16 +0000 [r59200]  Joshua Colp <jcolp@digium.com>

        * pbx/ael/ael_lex.c: Have ast_copy_string magically appear in the
          aelparse binary! DONT_OPTIMIZE should now work once again.

2007-03-24 01:39 +0000 [r59195]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_sip.c: Merged revisions 59194 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2
          lines Only try to handle a response if it has a response code.
          (ASA-2007-011) ........

2007-03-23 16:11 +0000 [r59188-59189]  Steve Murphy <murf@digium.com>

        * /: blocking out the fix in 59187... already incorporated here

        * /, apps/app_macro.c: Merged revisions 59186 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r59186 | murf | 2007-03-23 09:57:26 -0600 (Fri, 23 Mar 2007) | 1
          line Added a few words in the Macro doc strings about the
          behavior of macros with hangups (et al.), as per 9337 ........

2007-03-22 23:40 +0000 [r59180-59182]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_sip.c: don't allow string input to overrun the
          buffer to hold it (ASA-2007-010)

        * channels/chan_misdn.c: remove variables that are no longer used
          (--enable-dev-mode is good, developers should be using it)

2007-03-22 14:40 +0000 [r59145]  Steve Murphy <murf@digium.com>

        * utils/Makefile: The stuff in utils was compiling with -O6 even if
          DONT_OPTIMIZE is set in menuconfig. Added the include to fix that

2007-03-21 18:08 +0000 [r59081-59089]  Joshua Colp <jcolp@digium.com>

        * main/http.c: Add svg mimetype for pari.

        * res/res_monitor.c, /: Merged revisions 59086 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r59086 | file | 2007-03-21 14:03:20 -0400 (Wed, 21 Mar 2007) | 2
          lines Indicate the filename changed when it is changed. (issue
          #9311 reported by jsmith) ........

        * channels/chan_sip.c: Until we can do media level parsing for
          sendrecv/etc just use the first value found. This crept up when a
          phone was offered audio+video and returned an inactive video
          stream. chan_sip thought the phone said to put the person on hold
          but that was totally wrong. (issue #9319 reported by benbrown)

2007-03-20 21:04 +0000 [r59078]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * main/logger.c: Fix defines for inline stack backtraces (only used
          by developers anyway)

2007-03-20 20:42 +0000 [r59076]  Joshua Colp <jcolp@digium.com>

        * channels/iax2-parser.c: Copy len variable as well, should fix
          remaining IAX2 DTMF issues.

2007-03-20 17:48 +0000 [r59069-59070]  Steve Murphy <murf@digium.com>

        * apps/app_stack.c: Ooops. Sorry, messed up app_stack. This should
          return it to its previous, untouched, state.

        * apps/app_stack.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h:
          The fix for the AEL <<security hole>> (bug 9316) is here...

2007-03-20 13:16 +0000 [r59064]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/misdn_config.c,
          channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
          channels/misdn/chan_misdn_config.h: Merged revisions
          58849-58850,59062-59063 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) |
          1 line added method standard_dec for dialing out on groups, to
          avoid conflicts, which caused issues with some ISDN providers
          ........ r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13
          Mär 2007) | 1 line fixed the crypt_keys stuff ........ r59062 |
          crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line
          avoid sending a disconnect when we already received one. ........
          r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) |
          1 line modified a loglevel ........

2007-03-19  Jason Parker  <jparker@digium.com>

        * Asterisk 1.4.2 released.

2007-03-19 22:29 +0000 [r59049]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * funcs/func_strings.c: Oops, this should have been a %d all along

2007-03-19 15:52 +0000 [r59042]  Joshua Colp <jcolp@digium.com>

        * funcs/func_cdr.c: Fix typo in help for CDR function. (issue #9295
          reported by ajohnson)

2007-03-19 15:42 +0000 [r59040]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * configs/sip_notify.conf.sample: Fix unescaped semicolon (reported
          via -dev list)

2007-03-18 20:37 +0000 [r59037]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Issue #9313, Asterisk crash on SIP return
          code 0 (reported by qwerty1979)

2007-03-18 16:36 +0000 [r59035]  BJ Weschke <bweschke@btwtech.com>

        * apps/app_followme.c: Don't return a non-zero return code if the
          profile doesn't exist, to match what the documentation says it
          already does. (#9307 Reported by kkiely)

2007-03-16 16:12 +0000 [r58992]  Joshua Colp <jcolp@digium.com>

        * apps/app_page.c: Wait for the async thread to exit when hanging
          up all of the paged phones under all circumstances. (issue #9181
          reported by PhilSmith)

2007-03-16 01:42 +0000 [r58947-58957]  Russell Bryant <russell@digium.com>

        * configs/sla.conf.sample: fix a couple SLA documentation
          references

        * doc/ajam.tex (removed), doc/manager.tex (removed), doc/misdn.tex
          (removed), doc/freetds.txt (added), doc/odbcstorage.txt (added),
          doc/sla.tex, doc/cygwin.txt (added), doc/model.txt (added),
          doc/channelvariables.txt (added), doc/ael.txt (added),
          doc/billing.tex (removed), build_tools/prep_tarball,
          doc/callingpres.txt (added), doc/enum.txt (added),
          doc/localchannel.tex (removed), doc/musiconhold-fpm.txt (added),
          doc/cdrdriver.tex (removed), build_tools/make_buildopts_h,
          doc/security.txt (added), doc/imapstorage.txt (added),
          doc/PEERING, main/pbx.c, doc/odbcstorage.tex (removed),
          doc/freetds.tex (removed), doc/privacy.txt (added), configure.ac,
          doc/iax.txt (added), doc/ael.tex (removed),
          doc/channelvariables.tex (removed), doc/enum.tex (removed),
          doc/security.tex (removed), doc/math.txt (added), Makefile,
          doc/imapstorage.tex (removed), doc/privacy.tex (removed),
          doc/realtime.txt (added), doc/dundi.txt (added), doc/mysql.txt
          (added), apps/app_voicemail.c, doc/cliprompt.txt (added),
          doc/chaniax.txt (added), doc/app-sms.txt (added),
          doc/ast_appdocs.tex (removed), doc/realtime.tex (removed),
          doc/ices.txt (added), doc/dundi.tex (removed),
          doc/linkedlists.txt (added), doc/queuelog.txt (added),
          doc/extconfig.txt (added), doc/radius.txt (added),
          doc/cliprompt.tex (removed), doc/chaniax.tex (removed),
          doc/hardware.txt (added), doc/mp3.txt (added), doc/app-sms.tex
          (removed), doc/ices.tex (removed), doc/asterisk.tex (removed),
          doc/queuelog.tex (removed), doc/configuration.txt (added),
          doc/asterisk-conf.txt (added), doc/sla.pdf (added),
          doc/ip-tos.txt (added), doc/hardware.tex (removed), doc/h323.txt
          (added), doc/mp3.tex (removed), doc/configuration.tex (removed),
          doc/asterisk-conf.tex (removed), doc/jitterbuffer.txt (added),
          doc/channels.txt (added), doc/ip-tos.tex (removed),
          doc/extensions.txt (added), doc/queues-with-callback-members.txt
          (added), doc/apps.txt (added), makeopts.in, doc/ajam.txt (added),
          doc/misdn.txt (added), doc/manager.txt (added),
          doc/jitterbuffer.tex (removed), doc/extensions.tex (removed),
          doc/billing.txt (added), doc/localchannel.txt (added),
          doc/queues-with-callback-members.tex (removed), doc/cdrdriver.txt
          (added), doc/00README.1st (added): Making these documentation
          changes in the 1.4 branch upset various people, so these chanes
          will only be done in the trunk.

        * build_tools/prep_tarball: Add the --pdf option to the usage of
          rubber in prep_tarball

        * Makefile, build_tools/menuselect-deps.in, configure,
          include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
          configure script checking for GTK2 and some additional Makefile
          targets to support gmenuselect

2007-03-15 23:52 +0000 [r58946]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * main/pbx.c, doc/ast_appdocs.tex: Refashion dump command to match
          common syntax and update the resulting appdocs TeX file

2007-03-15 23:24 +0000 [r58941]  Russell Bryant <russell@digium.com>

        * doc/asterisk.tex: add a link to the rubber homepage

2007-03-15 23:11 +0000 [r58939]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_setcdruserfield.c, main/pbx.c,
          apps/app_hasnewvoicemail.c, apps/app_settransfercapability.c:
          Expand deprecation warnings from simply warning on use to the
          builtin documentation.

2007-03-15 22:51 +0000 [r58935-58937]  Russell Bryant <russell@digium.com>

        * doc/asterisk.tex, Makefile: Add Asterisk version information to
          the generated PDF

        * build_tools/prep_tarball: have prep_tarball attempt to build
          asterisk.pdf

2007-03-15 22:32 +0000 [r58933]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * funcs/func_realtime.c: Function works fine, but the documentation
          is backwards.

2007-03-15 22:25 +0000 [r58931]  Russell Bryant <russell@digium.com>

        * doc/ajam.tex (added), doc/manager.tex (added), doc/misdn.tex
          (added), doc/freetds.txt (removed), doc/odbcstorage.txt
          (removed), configure, doc/sla.tex, doc/cygwin.txt (removed),
          doc/model.txt (removed), doc/channelvariables.txt (removed),
          doc/ael.txt (removed), doc/billing.tex (added),
          doc/callingpres.txt (removed), doc/enum.txt (removed),
          doc/localchannel.tex (added), doc/musiconhold-fpm.txt (removed),
          doc/cdrdriver.tex (added), build_tools/make_buildopts_h,
          doc/security.txt (removed), doc/imapstorage.txt (removed),
          doc/PEERING, main/pbx.c, doc/odbcstorage.tex (added),
          doc/freetds.tex (added), doc/privacy.txt (removed), configure.ac,
          doc/iax.txt (removed), doc/ael.tex (added),
          doc/channelvariables.tex (added), doc/enum.tex (added),
          doc/security.tex (added), doc/math.txt (removed), Makefile,
          doc/imapstorage.tex (added), doc/privacy.tex (added),
          doc/realtime.txt (removed), doc/dundi.txt (removed),
          doc/mysql.txt (removed), apps/app_voicemail.c, doc/cliprompt.txt
          (removed), doc/chaniax.txt (removed), doc/app-sms.txt (removed),
          doc/ast_appdocs.tex (added), doc/realtime.tex (added),
          doc/ices.txt (removed), doc/dundi.tex (added),
          doc/linkedlists.txt (removed), doc/queuelog.txt (removed),
          doc/extconfig.txt (removed), doc/radius.txt (removed),
          doc/cliprompt.tex (added), doc/chaniax.tex (added),
          doc/hardware.txt (removed), doc/mp3.txt (removed),
          doc/app-sms.tex (added), doc/ices.tex (added), doc/asterisk.tex
          (added), doc/queuelog.tex (added), doc/configuration.txt
          (removed), doc/asterisk-conf.txt (removed), doc/sla.pdf
          (removed), doc/ip-tos.txt (removed), doc/hardware.tex (added),
          doc/h323.txt (removed), doc/mp3.tex (added),
          doc/configuration.tex (added), doc/asterisk-conf.tex (added),
          doc/jitterbuffer.txt (removed), doc/channels.txt (removed),
          doc/ip-tos.tex (added), doc/extensions.txt (removed),
          doc/queues-with-callback-members.txt (removed), doc/apps.txt
          (removed), makeopts.in, doc/ajam.txt (removed), doc/misdn.txt
          (removed), doc/manager.txt (removed), doc/jitterbuffer.tex
          (added), doc/extensions.tex (added), doc/billing.txt (removed),
          doc/localchannel.txt (removed),
          doc/queues-with-callback-members.tex (added), doc/cdrdriver.txt
          (removed), doc/00README.1st (removed): Merge changes from
          svn/asterisk/team/russell/LaTeX_docs. * Convert most of the doc
          directory into a single LaTeX formatted document so that we can
          generate a PDF, HTML, or other formats from this information. *
          Add a CLI command to dump the application documentation into
          LaTeX format which will only be include if the configure script
          is run with --enable-dev-mode. * The PDF turned out to be close
          to 1 MB, so it is not included. However, you can simply run "make
          asterisk.pdf" to generate it yourself. We may include it in
          release tarballs or have automatically generated ones on the web
          site, but that has yet to be decided.

2007-03-15 18:13 +0000 [r58923]  Joshua Colp <jcolp@digium.com>

        * channels/chan_iax2.c: Don't assume that the pvt structure will
          still exist after calling schedule_delivery as it may not. (issue
          #9278 reported by fmachado)

2007-03-14 19:18 +0000 [r58894-58906]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: Some people like to put "limitonpeer"
          instead of "limitonpeers" in their configuration. While we're at
          it, support "limitonpeerz" and "limitonpeerssssss". (inspired by
          issue #9172)

        * doc/sla.pdf, doc/sla.tex: Add a more basic example setup to the
          examples section

        * doc/security.txt, /: Merged revisions 58896 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14 Mar 2007) |
          3 lines Add a note to the security file that the Asterisk CLI and
          log files may contain sensitive information, and that people
          should keep this in mind. ........

        * configs/sla.conf.sample, apps/app_meetme.c: By default, don't
          attempt to do any CallerID handling at all with SLA because it is
          known to not work properly in some situations. However, add an
          option to enable it for those that would like to use it anyway.
          The short story behind this is that to properly handle CallerID
          with SLA, we need the ability to change the CallerID on an
          existing call, and we are not ready to handle that.

2007-03-14 01:47 +0000 [r58880]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * funcs/func_strings.c: Issue 9162 -
          pbx_substitute_variables_helper assumes the buffer is initialized
          to all zeroes. This fixes a case where it wasn't.

2007-03-13 23:19 +0000 [r58870-58872]  Russell Bryant <russell@digium.com>

        * apps/app_meetme.c: Ensure that the blinky lights show that the
          trunk stopped ringing when the trunk hangs up before a station
          has answered it. (issue #9234, reported by francesco_r)

        * configs/sla.conf.sample: fix the reference to the SLA
          documentation

2007-03-13 11:49 +0000 [r58843-58848]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Merged revisions 58847 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2
          lines Issue #9229 - No port in request URI on register to non
          default SIP ports (neelakantan) ........

        * channels/chan_sip.c: Don't hangup the call on OK or errors on
          MESSAGE and INFO inside of a dialog (like video update requests).

        * channels/chan_sip.c: Issue #9251 - Clear From URI from user
          attributes (tgrman)

2007-03-12 13:08 +0000 [r58825-58826]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
          revisions 57034,57523,57753,58558 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) |
          1 line fixed bugs.digium.com bugs: #9157 and bugs.beronet.com
          bugs: #302, #303, #304 ........ r57523 | crichter | 2007-03-02
          19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line fixed typo ........
          r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) |
          1 line fixed another place where the out_cause was hardcoded to
          16 ........ r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09
          Mar 2007) | 1 line we can free channel 31 as well, since we can
          occupy it ........

        * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
          channels/chan_misdn.c, channels/misdn/ie.c,
          channels/misdn/isdn_msg_parser.c: added UU transceiving and
          corect handling for rdnis

2007-03-12 01:21 +0000 [r58779-58783]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: Allow RFC2833 compensation to compensate for even
          stupider implementations by queueing up the end frame at the
          start, not the actual end. (issue #8963 reported by AndrewZ)

        * channels/chan_sip.c, configs/sip.conf.sample: Add
          matchexterniplocally setting which only substitutes your
          externip/externhost setting if it matches the localnet setting. I
          know of at least two people who need opposite settings, so I made
          it an option! (issue #8821 reported by kokoskarokoska)

2007-03-10 18:11 +0000 [r58638-58705]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix a few more places in chan_iax2 where
          the ast_frame used for receiving a frame was not properly
          initialized. - Interpolating a frame when the jitterbuffer is in
          use - decrypting a frame when IAX2 encryption is on - frames in
          an IAX2 trunk

        * apps/app_meetme.c: Make the compiler happy and initialize a
          variable.

        * doc/sla.pdf (added), doc/sla.txt (removed), doc/sla.tex (added):
          Merge some updates to the SLA documentation. I plan to keep
          working on this to explain all of the expected behavior with call
          handling, configuration details for specific phones, and other
          things. However, I got tired of doing it in plain text, so I
          switched to using LaTeX. I have included the PDF version. I
          haven't been able to get a nice looking plain text version out of
          it yet, but I'm not terribly concerned since this is supposed to
          be more of the manual, while the plain text sample configuration
          file is the reference.

2007-03-09 21:08 +0000 [r58584-58604]  Joshua Colp <jcolp@digium.com>

        * apps/app_voicemail.c: Fix spelling of unavailable in voicemail
          documentation. (issue #9248 reported by tensai)

        * /, channels/chan_sip.c: Merged revisions 58579 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2
          lines If we are unable to lookup the host in a c line we have to
          abort, otherwise the previous data is gone and we will
          (potentially) have no data when all is said and done. ........

2007-03-08 22:15 +0000 [r58510-58512]  Russell Bryant <russell@digium.com>

        * apps/app_meetme.c: Hang up the channel that put the call on hold
          in the event processing thread to avoid a race condition. Also,
          if the station originated the call that it is putting on hold,
          don't hang up the trunk if it was the only station on the call
          and it is hanging up due to hold and not a normal hangup.

        * channels/chan_zap.c: Add a missing break statement so that
          handling the above event does not incorrectly destroy the
          channel. (issue #9242, andrew)

2007-03-08 21:33 +0000 [r58479]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * res/res_odbc.c: Fix segfault (Issue 9236)

2007-03-08 20:54 +0000 [r58474]  Russell Bryant <russell@digium.com>

        * apps/app_meetme.c: Refactor hold handling a bit so that it does
          not require keeping the call up when a call is put on hold.

2007-03-08 18:01 +0000 [r58389-58436]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: Make early SDP seeding even smarter! We have to check
          codecs in the make_compatible function too. (issue #9221 reported
          by marcelbarbulescu)

        * main/dsp.c, /: Merged revisions 58388 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2
          lines Only print out debug message if the definition that makes
          the variables shows up was actually defined. (issue #9233
          reported by serginuez) ........

2007-03-08 13:23 +0000 [r58351-58354]  Kevin P. Fleming <kpfleming@digium.com>

        * main/http.c: this change was not needed; fclose() handles closing
          the file descriptor already

        * apps/app_meetme.c: fix a compiler warning, and overwriting 'res'
          value

        * main/http.c: fix two cases where HTTP session file descriptors
          would not be closed

2007-03-08 01:01 +0000 [r58243-58320]  Russell Bryant <russell@digium.com>

        * channels/chan_zap.c, configure, configure.ac: If we receive
          ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256,
          tzafrir) Also, update the configure script to make sure that we
          don't try to build chan_zap if the installed version of zaptel
          does not include ZT_EVENT_REMOVED.

        * /, channels/chan_iax2.c: (This bug was reported to me by Kinsey
          Moore) Merged revisions 58242 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) |
          7 lines Fix a problem where the Asterisk channel name could be
          that of the wrong IAX2 user for a call. This is because the first
          step of choosing this name is to look for an IAX2 peer that
          happens to have the same IP/port number that this call is coming
          from and assuming that is it. However, this is not always
          correct. So, I have made it change this name after authentication
          happens since at that point, we have an exact match. ........

2007-03-07 17:52 +0000 [r58240]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c, channels/chan_sip.c: Ensure we have (or should have)
          at least one matching codec before attempting early bridge SDP
          seeding. (issue #9221 reported by marcelbarbulescu)

2007-03-07 00:27 +0000 [r58165-58168]  Russell Bryant <russell@digium.com>

        * main/manager.c, /: Merged revisions 58164 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) |
          4 lines If the channels acquired using the manager Redirect
          action are not up, then don't attempt to do anything with them.
          It could lead to weird behavior, including crashes. (issue #8977)
          ........

2007-03-06 23:10 +0000 [r58121]  Steve Murphy <murf@digium.com>

        * /, channels/chan_sip.c: Merged revisions 58115 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1
          line Fix for 9220: Eyebeam cannot renew subscriptions for
          presence info. Reason: re-SUBSCRIBE requests don't include Accept
          headers, which the rfc says are optional (to put it tersely), (it
          uses MAY), and luckily, the sip_pvt struct has the format info
          stored, so we simply leave it if the format is set, and the
          accept header null. ........

2007-03-06 23:00 +0000 [r58119]  Russell Bryant <russell@digium.com>

        * configs/voicemail.conf.sample: Clarify the documentation of the
          dialout and sendvoicemail options. (issue #9000, caio1982 and
          serge-v)

2007-03-06 20:37 +0000 [r58053]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Merged revisions 58052 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2
          lines Change error message to proper message ........

2007-03-06 18:01 +0000 [r58023]  Russell Bryant <russell@digium.com>

        * channels/chan_skinny.c: Return an error of transmit_response is
          called without a session. (issue #9002)

2007-03-05 19:19 +0000 [r57870-57914]  Joshua Colp <jcolp@digium.com>

        * channels/chan_iax2.c: Since chan_iax2 does not support reception
          of DTMF with duration ensure that it is set to 0 on the frame.
          (issue #8521 reported by gdhgdh)

        * apps/app_meetme.c: Don't create a listen channel and record the
          conference unless the option is turned on. (issue #9204 reported
          by francesco_r)

        * apps/app_voicemail.c, /: Merged revisions 57869 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2
          lines Make create_dirpath use our standard for return values. -1
          is failure, 0 is success. (issue #9205 reported by ballares)
          ........

2007-03-05 15:20 +0000 [r57826]  Steve Murphy <murf@digium.com>

        * main/pbx.c, /: Merged revisions 57825 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1
          line Fixed a typo introduced via 9156 (either the gotos or their
          doc strings are wrong) ........

2007-03-05 04:19 +0000 [r57768-57798]  Joshua Colp <jcolp@digium.com>

        * main/slinfactory.c: Don't allow a NULL pointer to reach
          ast_frdup. (issue #9155 reported by cmaj)

        * res/res_jabber.c: Don't reference a potentially NULL pointer.
          (issue #9199 reported by klolik)

        * main/rtp.c: Preserve marker bit when P2P bridging. (issue #9198
          reported by edgreenberg)

2007-03-03 15:31 +0000 [r57707]  Steve Murphy <murf@digium.com>

        * pbx/ael/ael-test/ref.ael-vtest13, pbx/ael/ael-test/ref.ael-test2,
          pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test7:
          Updated the regression tests

2007-03-03 06:45 +0000 [r57649]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_voicemail.c, /: Merged revisions 57648 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007)
          | 2 lines Memory leak of a list, if call recording was abandoned
          ........

2007-03-03 00:59 +0000 [r57620]  Dwayne M. Hubbard <dhubbard@digium.com>

        * main/say.c: submitted patch for Georgian language, issue 9010,
          submitted by Alexander Shaduri

2007-03-03 00:02 +0000 [r57591]  Russell Bryant <russell@digium.com>

        * configs/sla.conf.sample: add missing configuration template.
          Thanks to Lacy Moore on asterisk-users for pointing this out\!

2007-03-02  Russell Bryant  <russell@digium.com>

        * Asterisk 1.4.1 released.

2007-03-02 23:03 +0000 [r57556]  Russell Bryant <russell@digium.com>

        * configure, configure.ac: Update the check that is used to
          determine whether zaptel transcoder support is present. The
          interface has changed.

2007-03-02 17:06 +0000 [r57477]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_sip.c: Merged revisions 57475 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2
          lines If a SIP message comes in and goes to a method handler that
          requires additional values that may not be present then send back
          an error. ........

2007-03-02 16:55 +0000 [r57426-57473]  Steve Murphy <murf@digium.com>

        * main/pbx.c, /: Merged revisions 57458 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1
          line further refinement in wording of goto documentation, as per
          9156, goto not proceeding to next instruction ........

        * pbx/pbx_ael.c, utils/ael_main.c: I almost had comma escapes
          right, but 9184 points out the problem-- the escape is removed by
          pbx_config, and pbx_ael should also, before sending it down into
          the pbx engine. Also, you have to insert it back in, if you are
          generating extensions.conf code from the AEL.

2007-03-02 00:20 +0000 [r57364-57396]  Russell Bryant <russell@digium.com>

        * main/file.c: Return the correct digit that interrupted the
          stream. This fixes exiting the Background application when using
          the m option. (issue #9176, mjagdis)

        * configs/sla.conf.sample, apps/app_meetme.c, doc/sla.txt,
          include/asterisk/channel.h: Merge changes from
          svn/asterisk/team/russell/sla_updates * Originally, I put in the
          documentation that only Zap interfaces would be supported on the
          trunk side. However, after a discussion with Qwell, we came up
          with a way to make IP trunks work as well, using some things
          already in Asterisk. So, here it is, this now officially supports
          IP trunks. * Update the SLA documentation to reflect how to setup
          IP trunks. * Add a section in sla.txt that describes how to set
          up an SLA system with voicemail. * Simplify the way DTMF
          passthrough is handled in MeetMe. * Fix a bug that exposed itself
          when using a Local channel on the trunk side in SLA. The
          station's channel needs to be passed to the dial API when dialing
          the trunk. * Change a WARNING message to DEBUG in channel.h. This
          message is of no use to users.

2007-03-01 22:21 +0000 [r57318]  Joshua Colp <jcolp@digium.com>

        * channels/chan_local.c, /: Merged revisions 57317 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar
          2007) | 2 lines Don't even attempt to optimize things when a
          proxy channel is involved. It will just explode in weird and
          unexplaineable ways. (issue #9175 reported by
          clegall_proformatique) ........

2007-03-01 03:02 +0000 [r57263]  TransNexus OSP Development <support@transnexus.com>

        * doc/osp.txt: 1. Corrected a typo for www.etsi.org. Thank Patrick.

2007-02-28 23:01 +0000 [r57144-57207]  Russell Bryant <russell@digium.com>

        * configs/sla.conf.sample, doc/sla.txt: minor tweaks to the sla
          docs

        * configs/sla.conf.sample, apps/app_meetme.c: Merge more changes
          from svn/asterisk/team/russell/sla_updates * Add support for
          private hold. By setting "hold=private" for a trunk, only the
          station that put the call on hold will be able to retrieve it
          from hold. Also, by setting "hold=private" for a station, any
          call that station puts on hold can only be retrieved by that
          station.

        * apps/app_meetme.c: Minor formatting change

        * configs/sla.conf.sample, apps/app_meetme.c: Merge changes from
          svn/asterisk/team/russell/sla_updates * Add support for the
          "barge=no" option for trunks. If this option is set, then
          stations will not be able to join in on a call that is on
          progress on this trunk.

2007-02-28 19:23 +0000 [r57139]  Steve Murphy <murf@digium.com>

        * main/pbx.c, /: Merged revisions 57118 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1
          line a small documentation update, to reflect reality in the goto
          doc strings, as per 9156, Goto does not proceed to next prio if
          jump fails ........

2007-02-28 18:57 +0000 [r57093]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_agent.c: Merged revisions 57092 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb
          2007) | 2 lines Fix a few more issues with the agent logoff CLI
          command. (issue #9123 reported by arbrandes) ........

2007-02-28 18:20 +0000 [r57089]  Russell Bryant <russell@digium.com>

        * configs/sla.conf.sample, apps/app_meetme.c: Merge current set of
          changes from svn/asterisk/team/russell/sla_updates * Add support
          for station ring delays. Ring delays can be set globally for a
          station or for specific trunks on the station. * Fix a few bugs
          in existing code. * Restructure and Reorganize code to improve
          readability and maintainability. * Improve formatting of the "sla
          show (trunks|stations)" CLI commands.

2007-02-28 17:55 +0000 [r57053-57055]  Joshua Colp <jcolp@digium.com>

        * apps/app_meetme.c: Picky compiler...

        * apps/app_speech_utils.c: Better handle timeouts when the
          individual speaks after everything has been played but before the
          timeout ends.

2007-02-28 17:15 +0000 [r57049]  Steve Murphy <murf@digium.com>

        * pbx/pbx_ael.c: I was surprised that I had not yet downgraded
          missing goto targets and macro call defs to a warning, in case
          they are in extensions.conf; I rectified this problem. Also, A
          goto in a macro to a target in a catch block was not being found;
          I fixed this too; the cause was that I needed to treat catch
          statements like an extension in the find_match code.

2007-02-27 17:36 +0000 [r56975]  Russell Bryant <russell@digium.com>

        * apps/app_voicemail.c: Fix voicemail email attachments. I missed
          the conversion of one of the line endings and there was an extra
          one where it should not have been. (issue #9128)

2007-02-26 22:01 +0000 [r56922]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_lookupcidname.c, apps/app_lookupblacklist.c: Picky,
          picky... show deprecation warning in application help, too
          (reported via list)

2007-02-26 20:42 +0000 [r56888]  Russell Bryant <russell@digium.com>

        * channels/chan_alsa.c: Restore the behavior of Asterisk 1.2 where
          if a device was not specified in alsa.conf, then we just use the
          system default, instead of creating our own default of hw:0,0.
          (issue #9139)

2007-02-26 20:07 +0000 [r56856]  Joshua Colp <jcolp@digium.com>

        * /, pbx/pbx_config.c: Merged revisions 56850 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2
          lines Obey the clearglobalvars option in extensions reload (or
          dialplan reload depending on your version). (issue #9146 reported
          by ramonpeek) ........

2007-02-26 20:04 +0000 [r56847]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Fix a crash in my last change to
          iax2_indicate(). (issue #9150)

2007-02-26 19:33 +0000 [r56805-56839]  Joshua Colp <jcolp@digium.com>

        * apps/app_record.c: Update app_record documentation to use new CLI
          command, core show file formats. (issue #9151 reported by junky)

        * main/pbx.c: Use ast_strlen_zero to see if the language and/or
          context argument is not present for Background instead of just
          checking if it is NULL. (issue #9141 reported by mjagdis)

2007-02-26 16:51 +0000 [r56785]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Do more complete locking of the
          chan_iax2_pvt struct in the indicate callback. (Problem brought
          up by Ben Smithurst on the asterisk-dev list)

2007-02-26 16:36 +0000 [r56783]  Joshua Colp <jcolp@digium.com>

        * main/asterisk.c: Allow both of the show version files and core
          show file versions CLI commands to work. (issue #9135 reported by
          mvanbaak)

2007-02-26 01:04 +0000 [r56730-56740]  Russell Bryant <russell@digium.com>

        * apps/app_meetme.c: Move a comment to be in the correct struct.

2007-02-25 14:46 +0000 [r56685]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * main/channel.c, /: Merged revisions 56684 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007)
          | 3 lines Issue 9130 - If prev is the last item on the channel
          list, then evaluating additional conditions (e.g. name prefix)
          will cause a NULL dereference. ........

2007-02-24 02:02 +0000 [r56569]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Make sure to set a speeddials parent on
          creation. Don't crash if hold is pressed when no call is active.
          Don't return in places that we shouldn't..

2007-02-24 00:53 +0000 [r56548]  Kevin P. Fleming <kpfleming@digium.com>

        * codecs/codec_zap.c: update to match zaptel 1.4 API change that
          was committed a few minutes ago

2007-02-23 23:24 +0000 [r56505]  Russell Bryant <russell@digium.com>

        * main/asterisk.c, /: Merged revisions 56504 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) |
          8 lines Fix up a couple more signal handlers to not do bad things
          that could cause various undesirable results. The other day, I
          made Asterisk deadlock by hitting Control-C because of a bad
          signal handler. Now, signal handlers just set a flag and write to
          an alert pipe for the flag to be handled. Then, there is another
          thread that is monitoring for these flags. If being run in
          console mode, it is just the main thread. If Asterisk is in the
          background, a thread is created to do it. ........

2007-02-23 21:53 +0000 [r56457]  Joshua Colp <jcolp@digium.com>

        * main/sched.c: Change log notice to debug. It is possible for a
          scheduled item to execute and be deleted at close to the same
          time and unavoidable. If this happens this message creeps up.

2007-02-23 20:20 +0000 [r56407]  Russell Bryant <russell@digium.com>

        * /, channels/chan_iax2.c: Merged revisions 56406 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) |
          4 lines Don't destroy mutexes before unregistering all of the
          entry points from the core. Also, fix a potential memory leak
          from not destroying the locks for all of the possible call
          numbers (about 32k of them). ........

2007-02-23 18:59 +0000 [r56372]  Kevin P. Fleming <kpfleming@digium.com>

        * build_tools/make_version_h: build special version strings for
          AADK/S800i builds

2007-02-23 17:58 +0000 [r56341]  Russell Bryant <russell@digium.com>

        * apps/app_voicemail.c: The IMAP storage code uses the same code to
          build the email that is used when voicemail is sent via email
          using something like sendmail. In the patch from bug 8033 to fix
          various IMAP storage problems, the line endings in the email file
          were changed in the code from "\n" to "\r\n". However, this
          breaks sending regular voicemail to email. So, this change
          conditionally sets line endings to "\r\n" only if IMAP_STORAGE is
          enabled. (issue #9128, patch by jarjarbinks, modified by me to
          not break IMAP storage)

2007-02-22 23:08 +0000 [r56277]  Russell Bryant <russell@digium.com>

        * configs/sla.conf.sample, main/dial.c, apps/app_meetme.c,
          doc/sla.txt: Merge changes from team/russell/sla_updates. This
          batch of changes to the SLA code does a few different things. * I
          made the SLA code event driven instead of having to act in a lot
          of busy loops while dialing things to wait for state changes.
          This makes the code more efficient and readable at the same time.
          * I have implemented a couple of new features. The first is
          inbound trunk ringing timeouts. This is an option that defines
          how long to let an incoming call on a trunk to ring. * I have
          also implemented ring timeouts for stations. They may be
          specified for the entire station, meaning it is how long to let
          the station ring before giving up. You can also specify a ring
          timeout for a specific trunk on a station. So, you can say that
          you only want a specific station to ring 5 seconds if it is line1
          ringing, but otherwise, there is no timeout.

2007-02-22 18:49 +0000 [r56231]  Joshua Colp <jcolp@digium.com>

        * main/channel.c, /, channels/chan_sip.c: Merged revisions 56230
          via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2
          lines Only change the original or clone channel if it's the
          channel behind the proxy channel, not if it's just a regular
          bridged channel. ........

2007-02-22 14:06 +0000 [r56169]  TransNexus OSP Development <support@transnexus.com>

        * doc/osp.txt: Update OSP documentation for v1.4.

2007-02-22 10:33 +0000 [r56125]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Move message from verbose to debug

2007-02-22 02:39 +0000 [r56094]  Steve Murphy <murf@digium.com>

        * sounds/Makefile: updated the sound tarball versions in Makefile

2007-02-22 01:24 +0000 [r56011-56055]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: Restructure a little bit of code to reduce
          nesting. There is no functionality change here.

        * /, channels/chan_sip.c: Merged revisions 56010 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) |
          3 lines If we receive a frame that is not in any of the
          negotiated formats, then drop it. (potentially issue #8781 and
          SPD-12) ........

2007-02-22 00:35 +0000 [r56008]  Joshua Colp <jcolp@digium.com>

        * main/cli.c: Print out deprecation notice on usage output of CLI
          commands. (issue #8925 reported by blitzrage)

2007-02-22 00:08 +0000 [r56006]  Kevin P. Fleming <kpfleming@digium.com>

        * main/loader.c: disable unloading of embedded modules... there is
          a fundamental problem with doing so that will not be fixed in
          this version of Asterisk due to its invasiveness

2007-02-21 20:35 +0000 [r55957]  Joshua Colp <jcolp@digium.com>

        * /, apps/app_meetme.c: Merged revisions 55956 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2
          lines Change naughty warning message to provide useful
          information. If a write now fails on a channel in meetme it will
          tell you the channel name instead of spitting out the wrong error
          message. ........

2007-02-21 20:27 +0000 [r55954]  Jason Parker <jparker@digium.com>

        * channels/chan_gtalk.c: Fix locking issue, and accept
          "transport-accept" as a valid accept message. This should solve
          issues 8970 and 8503.

2007-02-21 20:22 +0000 [r55951]  Russell Bryant <russell@digium.com>

        * apps/app_meetme.c: Simplify the last change to app_meetme, and
          move the call to dispose_conf() up into the block where we know a
          conf exists.

2007-02-21 20:16 +0000 [r55914-55949]  Joshua Colp <jcolp@digium.com>

        * apps/app_meetme.c: Only dispose of the conference if one was
          created.

        * apps/app_speech_utils.c: Only start playing the next file if we
          have not been quieted.

        * channels/chan_sip.c: Add a flag that indicates whether a SIP
          dialog is an outgoing call or not. SIP_OUTGOING originally did it
          but it was repurposed to the direction of the last transaction,
          which can cause update_call_counter to falsely decrease the wrong
          counters. (please don't hurt me oej) (issue #8943 reported by
          mdu113)

2007-02-21 14:06 +0000 [r55869]  Kevin P. Fleming <kpfleming@digium.com>

        * /, build_tools/make_version: Merged revisions 55868 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r55868 | kpfleming | 2007-02-21 08:03:11 -0600 (Wed, 21
          Feb 2007) | 2 lines use new tag version script ........

2007-02-21 08:32 +0000 [r55834]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Issue #8848 - Turn off lamp more quickly
          after transfer (decrement inuse early on transferer's call leg)

2007-02-21 02:01 +0000 [r55799]  Jason Parker <jparker@digium.com>

        * channels/chan_gtalk.c: Fix segfault when buddy couldn't be found.
          Issue 7764, patch by sailer

2007-02-21 01:03 +0000 [r55751-55758]  Russell Bryant <russell@digium.com>

        * apps/app_meetme.c: Improve the reference counting to fix bugs
          where people report seeing conferences listed that have no
          members. (issue #9073)

2007-02-21 00:11 +0000 [r55670-55741]  Joshua Colp <jcolp@digium.com>

        * apps/app_voicemail.c: Better handle dropped IMAP connections.
          (issue #9054 reported by bsmithurst)

        * channels/chan_sip.c: Return behavior I removed. I did not
          remember that you could just add a localnet entry to make it
          work.

        * channels/chan_sip.c: Don't test our own address against the
          localnet settings. At least one person has had issues as a result
          of this from #7051 so I'm reversing it. (issue #8821 reported by
          kokoskarokoska)

        * /, channels/chan_agent.c: Merged revisions 55669 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb
          2007) | 2 lines Defer clearing callback information if channels
          are up until they are hung up. This ensures the hangup process
          goes smoothly and no channels get hung in limbo. (issue #8088
          reported by kebl0155) ........

2007-02-20 20:26 +0000 [r55589-55634]  Russell Bryant <russell@digium.com>

        * main/http.c: Add the Asterisk version information to the Server
          header in HTTP responses. (requested by Pari)

        * include/asterisk/manager.h: Increase the maximum number of
          manager headers to 128, at the request of Pari.

2007-02-20 16:53 +0000 [r55555]  Jason Parker <jparker@digium.com>

        * channels/chan_gtalk.c, res/res_jabber.c: No need to cast nor free
          with strdupa (thanks file) 55555!

2007-02-20 16:41 +0000 [r55553]  Russell Bryant <russell@digium.com>

        * configs/sla.conf.sample: Change the formatting of sla.conf.sample
          to make it more readable. (issue #9112, blitzrage)

2007-02-19 21:12 +0000 [r55483]  Olle Johansson <oej@edvina.net>

        * res/res_jabber.c: - Not sending arguments to an application is
          not "out of memory" - Making error messages a bit more clear

2007-02-19 18:11 +0000 [r55435]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_voicemail.c, /: Merged revisions 55434 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r55434 | tilghman | 2007-02-19 12:09:09 -0600 (Mon, 19 Feb 2007)
          | 2 lines forcename and forcegreetings options should check to
          see if the recording already exists ........

2007-02-19 14:52 +0000 [r55397]  Doug Bailey <dbailey@digium.com>

        * channels/chan_iax2.c: Changed iax2 process thread to detached to
          correct memory leak due to left over thread context on thread
          exit. Modified module unload process to avoid deadlocks on
          pthread cancels

2007-02-18 12:35 +0000 [r55250-55278]  Olle Johansson <oej@edvina.net>

        * /, apps/app_record.c: Merged revisions 55277 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2
          lines Documentation update (#9053, jsmith) ........

        * /: Block patch that was made only for 1.2 (already implemented in
          1.4 and trunk)

2007-02-17 17:39 +0000 [r55219]  Joshua Colp <jcolp@digium.com>

        * apps/app_queue.c: Add missing membername option to AddQueueMember
          documentation. (issue #9088 reported by seanbright)

2007-02-17 17:10 +0000 [r55217]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Fix an issue where callerid would not be
          displayed on some phones. Issue 8995, initial patch and research
          done by wedhorn

2007-02-17 03:55 +0000 [r55086-55154]  Joshua Colp <jcolp@digium.com>

        * apps/app_dial.c, /: Merged revisions 55153 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2
          lines Answer the channel before recording privacy information.
          (issue #8926 reported by lmamane) ........

        * apps/app_queue.c: Make the 'i' option of Queue actually work.
          (issue #8986 reported by utis)

        * /, channels/chan_sip.c: Merged revisions 55073 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2
          lines Allow chan_sip to handle attended transfers from a SIP
          phone that is sitting behind chan_agent. Yes folks, all it took
          was one line of code. (issue #8784 reported by pzieba) ........

2007-02-17 00:40 +0000 [r55006-55052]  Russell Bryant <russell@digium.com>

        * configure, include/asterisk/autoconfig.h.in, configure.ac: If the
          pg_config application is found, but there is probably executing
          it, then consider postgres unavailable. (issue #8637)

        * codecs/gsm/Makefile: Filter out yet another architecture that
          does not work with the optimizations in the built-in libgsm.
          (issue 8637, ovi)

        * /, apps/app_meetme.c, configs/meetme.conf.sample: Merged
          revisions 55005 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) |
          9 lines Revert the change I did in revisions 54955, 54969, and
          54970, in 1.2, 1.4, and trunk. I decided that once a conference
          is created from meetme.conf, it is acceptable behavior that the
          pin can not be changed until the conference goes away. I also
          added a note in meetme.conf to describe this behavior. We still
          have another issue in 1.4 and trunk where some conferences with
          no users don't go away. That is the real bug that needs to be
          addressed here. ........

2007-02-16 22:18 +0000 [r55002]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_agent.c: Merged revisions 54999 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb
          2007) | 2 lines Do not send indications through ast_indicate in
          chan_agent but instead go directly to the technology. This way
          when indications are emulated they happen on the Agent channel
          and do not screw up formats on the channels. (issue #8439
          reported by punkgode) ........

2007-02-16 21:12 +0000 [r54969]  Russell Bryant <russell@digium.com>

        * /, apps/app_meetme.c: Merged revisions 54955 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) |
          5 lines For conferences that are configured in meetme.conf, check
          the configuration file every time someone joins the conference
          instead of only when the conference is first created. This is to
          ensure that changes to the pin numbers in the config file are
          always honored. (issue #9073) ........

2007-02-16 18:51 +0000 [r54924]  Joshua Colp <jcolp@digium.com>

        * apps/app_dial.c: Need to check macro extension as well as macro
          context for directed pickup.

2007-02-16 18:03 +0000 [r54888-54898]  Russell Bryant <russell@digium.com>

        * pbx/pbx_config.c: Fix setting "autofallthrough" to yes by
          default. It was set to enabled in pbx.c. However, if the option
          was not present in extensions.conf, then pbx_config.c would set
          it back to disabled.

        * res/res_features.c: Clean up a few coding guidelines issues -
          spaces to tabs, use sizeof() to pass the size of a static buffer,
          add spaces ...

2007-02-16 17:25 +0000 [r54886]  Jason Parker <jparker@digium.com>

        * main/asterisk.c: Clarify a restart message. It's silly, but the
          reporter had a very valid point. Issue 9079

2007-02-16 17:02 +0000 [r54884]  Joshua Colp <jcolp@digium.com>

        * apps/app_dial.c: Allow directed pickup to pick up the real
          context instead of the macro context if a Macro is used. (issue
          #8984 reported by jamesb63)

2007-02-16 12:06 +0000 [r54772-54787]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Issue #7541 - Handle multipart attachments
          to SIP messages - even if boundary is quoted.

        * /, res/res_agi.c: Merged revisions 54771 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r54771 | oej | 2007-02-16 12:38:03 +0100 (Fri, 16 Feb 2007) | 2
          lines Issue #9069 - If we open with TH we should not close with
          /TD. (seanbright) ........

2007-02-16 00:48 +0000 [r54481-54714]  Joshua Colp <jcolp@digium.com>

        * apps/app_speech_utils.c: Don't let dtmf leak over into the engine
          and let it skew the results... also give DTMF results priority.
          (issue #9014 reported by surftek)

        * apps/app_dial.c, /: Merged revisions 54622 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2
          lines Use a separate variable to indicate execution should
          continue instead of the return value. (issue #8842 reported by
          pluto70) ........

        * apps/app_dial.c: Forward begin DTMF frames as well as end. (issue
          #9068 reported by mhardeman)

2007-02-14 18:44 +0000 [r54439]  Olle Johansson <oej@edvina.net>

        * /: Block patch only needed in 1.2

2007-02-14 16:56 +0000 [r54375]  Matt Frederickson <creslin@digium.com>

        * channels/chan_zap.c, /: Merged revisions 54373 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2
          lines When handling glare on a PRI, move the requested channel
          rather than hang up the old one. Fix for 8957 and 9011. ........

2007-02-14 01:09 +0000 [r54290]  Joshua Colp <jcolp@digium.com>

        * main/channel.c: Add G722 to ast_best_codec. If anyone disagrees
          with it's placement, feel free to change it. (issue #9045
          reported by gork)

2007-02-13 21:31 +0000 [r54204-54235]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: Remove a couple of leftover debug messages

        * include/asterisk/devicestate.h: Fix the documentation on the
          return values from device state provider registration and
          deletion.

        * channels/chan_sip.c: If we fail to create the SIP socket, then
          return -1 from reload_config() so that load_module() will return
          AST_MODULE_LOAD_DECLINE. Otherwise, the console will just get
          spammed with error messages every time chan_sip tries to send a
          message.

2007-02-13 18:41 +0000 [r54180]  Olle Johansson <oej@edvina.net>

        * /: Blocking patch for 1.2 only

2007-02-12 19:17 +0000 [r54066-54103]  Russell Bryant <russell@digium.com>

        * main/dial.c, include/asterisk/dial.h: Change
          ast_set_state_callback() to ast_dial_set_state_callback()

        * main/dial.c, apps/app_meetme.c, apps/app_page.c,
          include/asterisk/dial.h: - Add the ability to register a callback
          to monitor state changes in an asynchronous dial operation. -
          Rename the various references to "status" to "state" in the dial
          API

2007-02-12 16:34 +0000 [r54026]  Joshua Colp <jcolp@digium.com>

        * configure, configure.ac: Make the --without-oss argument work.
          (issue #9026 reported by puzzled)

2007-02-12 15:38 +0000 [r54002]  Russell Bryant <russell@digium.com>

        * configs/users.conf.sample: Fix a typo where "vmpassword" should
          be "vmsecret"

2007-02-10 09:09 +0000 [r53878-53881]  Paul Cadach <paul@odt.east.telecom.kz>

        * channels/chan_h323.c: Fix VLDTMF reception

        * apps/app_echo.c: Much simpler than previous one ;-)

        * main/channel.c: Provide correct DTMF duration

        * main/cli.c: Bring deprecated 'debug channel <x|all>' command back

2007-02-10 06:06 +0000 [r53850]  Kevin P. Fleming <kpfleming@digium.com>

        * configure, configure.ac, acinclude.m4: don't display the
          --with-imap message unless --with-imap was specified without a
          path use '-n' instead of '! -z' for tests

2007-02-10 01:02 +0000 [r53783-53821]  Russell Bryant <russell@digium.com>

        * apps/app_meetme.c: Add some output for "show application
          SLAStation/SLATrunk"

        * channels/chan_sip.c: Change some text to properly state "On
          Hold", which was already done in trunk.

        * configs/sla.conf.sample, include/asterisk/app.h,
          include/asterisk/utils.h, main/dial.c, apps/app_meetme.c,
          channels/chan_sip.c, doc/sla.txt (added),
          include/asterisk/linkedlists.h, include/asterisk/dial.h: Merge
          team/russell/sla_rewrite This is a completely new implementation
          of the SLA functionality introduced in Asterisk 1.4. It is now
          functional and ready for testing. However, I will be adding some
          additional features over the next week, as well. For information
          on how to set this up, see configs/sla.conf.sample and
          doc/sla.txt. In addition to the changes in app_meetme.c for the
          SLA implementation itself, this merge brings in various other
          changes: chan_sip: - Add the ability to indicate HOLD state in
          NOTIFY messages. - Queue HOLD and UNHOLD control frames even if
          the channel is not bridged to another channel. linkedlists.h: -
          Add support for rwlock based linked lists. dial.c: - Add the
          ability to run ast_dial_start() without a reference channel to
          inherit information from.

        * apps/app_echo.c: When the Echo() application receives the digit
          '#', echo that back as well. Since we already sent the BEGIN
          frame for that digit, it makes sense to send the END as well.

2007-02-09 23:52 +0000 [r53779-53781]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_gtalk.c: another dependency

        * apps/app_adsiprog.c, apps/app_voicemail.c, res/res_config_odbc.c,
          funcs/func_odbc.c, res/res_adsi.c: add some inter-module
          dependencies

        * build_tools/get_moduleinfo, build_tools/get_makeopts: fix awk
          scripts to work when both MODULEINFO and MAKEOPTS are present in
          a source file

2007-02-09 19:33 +0000 [r53749]  Joshua Colp <jcolp@digium.com>

        * apps/app_dial.c: Temporarily change musicclass on channel to one
          specified in Dial so that the 'm' option functions properly.
          (issue #8969 reported by christianbee)

2007-02-09 16:42 +0000 [r53715]  Kevin P. Fleming <kpfleming@digium.com>

        * doc/imapstorage.txt, configure, configure.ac: clarify the fact
          that voicemail IMAP storage cannot be built against a distro's
          binary c-client library package (at least not at this time)

2007-02-08 23:18 +0000 [r53672]  Olle Johansson <oej@edvina.net>

        * main/acl.c: Don't output debug unless we asked for it

2007-02-08 17:54 +0000 [r53601]  Joshua Colp <jcolp@digium.com>

        * apps/app_speech_utils.c: Fix timeout issue when utterance is
          longer then timeout itself.

2007-02-08 13:47 +0000 [r53530-53532]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * main/loader.c: Issue 9007 - Mutex not released on early return

        * apps/app_voicemail.c, /: Merged revisions 53529 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r53529 | tilghman | 2007-02-08 07:36:10 -0600 (Thu, 08 Feb 2007)
          | 2 lines Issue 9003 - If fullname is empty, quote() passes back
          "\"" ........

2007-02-07 23:52 +0000 [r53464-53497]  Russell Bryant <russell@digium.com>

        * main/db1-ast/Makefile: When building libdb1.a, put the additional
          flags needed at the beginning of ASTCFLAGS, instead of at the
          end. This way, we ensure that we find the local headers first
          before accidentally trying to use headers that exist in locations
          specified in the ASTCFLAGS passed from the main Makefile. (issue
          #8637, ovi)

        * main/Makefile: The clean target actually needs to run "distclean"
          on editline. This is because we need to make sure that its
          configure script gets executed again, because the CFLAGS we want
          to pass to editline may have changed.

2007-02-07 17:53 +0000 [r53434]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: We can not reliably do P2P bridging with DTMF passing
          back with compensation if we need to listen for DTMF frames.
          (issue #8962 reported by caio1982)

2007-02-07 17:39 +0000 [r53429]  Russell Bryant <russell@digium.com>

        * main/rtp.c: When parsing the NTP timestamp in a sender report
          message, you are supposed to take the low 16 bits of the integer
          part, and the high 16 bits of the fractional part. However, the
          code here was erroneously taking the low 16 bits of the
          fractional part. It then shifted the result 16 bits down, so the
          result was always zero. This fix makes it grab the appropriate
          high 16 bits, instead. (issue #8991, pointed out by
          andre_abrantes)

2007-02-07 17:04 +0000 [r53358-53399]  Joshua Colp <jcolp@digium.com>

        * apps/app_playback.c: Directly load say.conf in load_module
          instead of calling the reload function. (issue #8946 reported by
          junky)

        * /, channels/chan_iax2.c: Merged revisions 53357 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2
          lines Fix a few potential memory leaks with realtime users and
          peers. (issue #8999 reported by bsmithurst) ........

2007-02-07 15:33 +0000 [r53355]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * /, apps/app_macro.c: Merged revisions 53354 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r53354 | tilghman | 2007-02-07 09:30:02 -0600 (Wed, 07 Feb 2007)
          | 2 lines Issue 7440 - Macro called from Macro from the h
          extension exits prematurely ........

2007-02-07 09:22 +0000 [r53324]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
          revisions 52843 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 Jan 2007) |
          1 line fixed some possible segfaults. also fixed an very
          important bug which occurs on high load (when calls are very fast
          generated) ........

2007-02-07 05:24 +0000 [r53246-53294]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * res/res_jabber.c: Text fix for jabber reload command (reported by
          bkruse via IRC)

        * main/manager.c, /: Merged revisions 53245 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r53245 | tilghman | 2007-02-06 00:58:28 -0600 (Tue, 06 Feb 2007)
          | 2 lines Issue 8987 - Status could return two responses
          (mnicholson) ........

2007-02-05 23:43 +0000 [r53222]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Formatting

2007-02-05 17:06 +0000 [r53150-53152]  Joshua Colp <jcolp@digium.com>

        * apps/app_playback.c: Ensure say_cfg is NULL when the module is
          loaded. (issue #8946 reported by junky)

        * apps/app_playback.c: Unregister Playback CLI commands as well as
          dialplan application. (issue #8946 reported by junky)

2007-02-05 00:18 +0000 [r53143]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Add some comments on queue system behaviour
          and how it affects the SIP channel

2007-02-03 21:05 +0000 [r53138]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Make SIPDtmfMode application work with
          recent capability changes, and also fix an RTP stack issue when
          the auto option was used. (issue #8972 reported by mdu113)

2007-02-03 20:44 +0000 [r53135-53136]  Russell Bryant <russell@digium.com>

        * apps/app_dial.c, /: Merged revisions 53133 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03 Feb 2007) |
          4 lines set the DIALSTATUS variable to contain "INVALIDARGS" when
          the dial application exits early because of invalid arguments
          instead of just leaving it empty. (issue #8975) ........

2007-02-03 10:02 +0000 [r53131]  Paul Cadach <paul@odt.east.telecom.kz>

        * channels/h323/ast_h323.cxx: Remove quote from H.323 vendor string
          because due to compatibilities with CS1000 reported at
          www.voip-info.org

2007-02-02 21:26 +0000 [r53129]  BJ Weschke <bweschke@btwtech.com>

        * UPGRADE.txt, apps/app_queue.c: I'm baaaaaaaaaack. :) Post a
          warning to the console that things might possibly be
          misconfigured when queue member's states are still 'Not in Use'
          when we're about to bridge them with a caller from queue. Also,
          put some documentation quoted from oej's queues.txt efforts
          started in /trunk today. This commit puts #7433 into feedback
          state for 1.4, and pending no further negative feedback, it will
          finally be closed.

2007-02-02 17:15 +0000 [r53114-53120]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: Correct a copy/pasted error message line for RTCP.

        * main/config.c, /: Merged revisions 53117 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r53117 | file | 2007-02-02 10:58:09 -0600 (Fri, 02 Feb 2007) | 2
          lines Pass the glob expanded filename to process_text_line so
          that error messages contain the actual filename, not the original
          include one. (issue #8959 reported by tzafrir) ........

        * Makefile: Add systemname to asterisk.conf generation per recent
          discussions about it. (issue #8968 reported by blitzrage)

2007-02-02 00:24 +0000 [r53109]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c, configs/sip.conf.sample: Disable the direct
          p2p RTP call setup in SIP. You can enable it in sip.conf, but it
          is now considered experimental until we solve the
          AST_CONTROL_ANSWER with payload and videocaps stuff.

2007-02-01 22:24 +0000 [r53097-53104]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_sip.c: Merged revisions 53103 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2
          lines Copy noncodeccapability over to the joint variable so that
          telephone-event will get transmitted in the sent INVITE. ........

        * main/db1-ast/hash/hash.c: Huh... fix the berkeley DB to compile
          here as well, but it apparently required both dev mode and no
          optimizations to creep up.

        * /, channels/chan_sip.c: Merged revisions 53095 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2
          lines Don't negotiate RFC2833 when not configured to do so.
          (issue #8799 reported by mdu113) ........

2007-02-01 21:24 +0000 [r53093]  Russell Bryant <russell@digium.com>

        * funcs/func_strings.c: Fix the FIELDQTY function to not crash.
          (reported by blitzrage and Corydon on IRC)

2007-02-01 21:15 +0000 [r53091]  Olle Johansson <oej@edvina.net>

        * /: Going backwards, blame file.

2007-02-01 21:11 +0000 [r53086-53088]  Joshua Colp <jcolp@digium.com>

        * /, res/res_musiconhold.c: Merged revisions 53084 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r53084 | file | 2007-02-01 15:03:10 -0600 (Thu, 01 Feb
          2007) | 2 lines Return previous behavior of having MOH pick up
          where it was left off. (issue #8672 reported by sinistermidget)
          ........

        * funcs/func_strings.c: Make func_strings build under dev mode.
          Didn't I do this today already in the berkeley DB?

2007-02-01 21:05 +0000 [r53079-53085]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: - Clean INC_COUNT flag when we decrement
          call counter - If it's still set at time of dialog destruction,
          make sure we decrement the device call counter properly before we
          destroy the dialog

        * apps/app_queue.c: Change debug level for state change message
          that is not really informative when debugging app_queue

        * channels/chan_sip.c: Cleaning up the devicestate callback
          function

2007-02-01 20:13 +0000 [r53075-53077]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * funcs/func_strings.c: Oops.

        * /, funcs/func_strings.c: Merged revisions 53074 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r53074 | tilghman | 2007-02-01 14:07:35 -0600 (Thu, 01 Feb 2007)
          | 2 lines Bug 8965 ........

2007-02-01 19:33 +0000 [r53072]  Joshua Colp <jcolp@digium.com>

        * main/asterisk.c: Add missing 'F' letter to getopt so it magically
          becomes a valid option. (issue #8960 reported by tzafrir)

2007-02-01 19:21 +0000 [r53070]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * main/pbx.c, /, funcs/func_strings.c: Merged revisions 53069 via
          svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r53069 | tilghman | 2007-02-01 13:13:53 -0600 (Thu, 01 Feb 2007)
          | 2 lines No wonder FIELDQTY doesn't work with functions... the
          documentation in pbx.c was wrong ........

2007-02-01 17:37 +0000 [r53064]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Fix silly logic. We really want to write
          UDPTL frames out when the call is up.

2007-02-01 16:35 +0000 [r53062]  Olle Johansson <oej@edvina.net>

        * configs/sip.conf.sample: Add explanation of port= in combination
          with defaultip= (thanks jsmith)

2007-02-01 13:17 +0000 [r53060]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c: we update the name on any first reply of
          our setup

2007-02-01 11:07 +0000 [r53057]  Paul Cadach <paul@odt.east.telecom.kz>

        * channels/chan_h323.c: chan_h323 is very stable, so let it built
          by default

2007-02-01 00:24 +0000 [r53050-53052]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: When going on hold have the side that was put on hold
          reinvite back to Asterisk. When going off hold have the side that
          was taken off hold reinvited back to the other party.

        * main/rtp.c: Add more frame types to forward in the RTP bridge
          loops.

2007-01-31 21:32 +0000 [r52859-53046]  Russell Bryant <russell@digium.com>

        * main/cdr.c, main/manager.c, pbx/pbx_spool.c,
          channels/chan_skinny.c, channels/chan_h323.c, main/http.c,
          pbx/pbx_dundi.c, apps/app_rpt.c, channels/chan_mgcp.c,
          main/pbx.c, channels/chan_zap.c, /, apps/app_meetme.c,
          channels/chan_sip.c, apps/app_queue.c, channels/chan_iax2.c:
          Merged revisions 53045 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) |
          3 lines Fix a bunch of places where pthread_attr_init() was
          called, but pthread_attr_destroy() was not. ........

        * apps/app_userevent.c: Remove an extra \r\n from manager user
          events. (issue #8955, mnicholson)

        * main/rtp.c, /: Merged revisions 53039 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 Jan 2007) |
          3 lines Use the proper format string to print unsigned values in
          the rtp debug output. (issue #8954, wmis) ........

        * apps/app_queue.c: Only changed the paused status in an existing
          queue member if the paused column exists.

        * apps/app_queue.c: Instead of always creating a realtime queue
          member as unpaused, read the "paused" column and use that value
          for the paused status of the member. (issue #8949, jmls)

        * contrib/init.d/rc.suse.asterisk: Update init script for SuSE 10.
          (issue #8363, johnlange)

        * doc/cdrdriver.txt: Add documentation for using cdr_pgsql. (issue
          #8942, lters)

        * configure, include/asterisk/autoconfig.h.in, configure.ac,
          codecs/codec_gsm.c: When we are checking for a system installed
          version of libgsm, we need to check for gsm.h as well.
          Furthermore, when checking for this header, it may be located in
          a gsm/ sub directory, so check for that, as well. (issue #8773)

        * channels/chan_sip.c: Only set the DTMF flag on the rtp structure
          if the DTMF mode is actually RFC2833, not just that it is not
          INFO. This makes it get set for inband DTMF as well, which is not
          valid. (issue #8936)

        * main/asterisk.c, /: Merged revisions 52903 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30 Jan 2007) |
          9 lines The SIGHUP handler was implemented to allow admins to
          send SIGHUP to a running Asterisk process to reload the
          configuration. However, doing the actual reload in the signal
          handler itself is a very bad thing to do, because the reload
          process includes calling non-reentrant functions such as
          malloc/calloc/etc. If Asterisk is running in the background, then
          the reload will happen immediately. However, if running in
          console mode, the reload doesn't work until something is typed at
          the console. That sort of defeats the purpose, but I don't see an
          easy way to get around it at this point. ........

2007-01-30 15:29 +0000 [r52856]  Joshua Colp <jcolp@digium.com>

        * channels/chan_iax2.c: Drop the deprecated show commands since the
          original ones were changed back. (issue #8937 reported by
          PCadach)

2007-01-30 08:46 +0000 [r52807-52809]  Paul Cadach <paul@odt.east.telecom.kz>

        * channels/chan_h323.c: Revert reprecation of h.323 gk cycle
          command from pre-1.4 version instead of duplicated h323 cycle gk

        * res/res_odbc.c: Don't play with free()'d pointers

        * configure, acinclude.m4: Handle non-standard OpenH323/PWLib
          library names

2007-01-30 00:15 +0000 [r52763]  Russell Bryant <russell@digium.com>

        * /, channels/chan_iax2.c: Merged revisions 52762 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) |
          5 lines Fix the extraction of the timestamp from video frames. It
          was using the mapping for a mini-frame instead of a video-frame,
          which caused it to get invalid data. (issue #8795, mihai)
          ........

2007-01-29 23:43 +0000 [r52717]  Joshua Colp <jcolp@digium.com>

        * apps/app_mixmonitor.c, /: Merged revisions 52716 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan
          2007) | 2 lines Now that filename is part of the structure and
          since it comes before postprocess... we have to add it to our
          postprocess line. (reported on asterisk-dev by Boris Bakchiev)
          ........

2007-01-29 22:58 +0000 [r52688-52695]  Russell Bryant <russell@digium.com>

        * main/Makefile: Add a missing quotation mark. This was pointed out
          by jcmoore on #asterisk-dev.

        * main/manager.c: Remove a recursive lock of the manager session.
          This was pointed out by zandbelt in issue #8711.

2007-01-29 22:12 +0000 [r52679]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * pbx/pbx_config.c: Argument number correction

2007-01-29 21:36 +0000 [r52611-52647]  Russell Bryant <russell@digium.com>

        * main/Makefile: ASTLDFLAGS needs to be passed to the editline
          configure script as LDFLAGS. (issue #8928, zandbelt)

        * main/rtp.c: Fix a problem with packet-to-packet bridging and DTMF
          mode translation. P2P bridging can only be used when the DTMF
          modes don't match if the core is monitoring DTMF in both
          directions. Then, the core will handle the translation.
          Otherwise, this bridging method can not be used. (issue #8936)

        * main/manager.c: The session lock can not be held while calling
          action callbacks. If so, then when the WaitEvent callback gets
          called, then no event can happen because the session can't be
          locked by another thread. Also, the session needs to be locked in
          the HTTP callback when it reads out the output string. This fixes
          the deadlock reported in both 8711 and 8934. Regarding issue
          8711, there still may be an issue. If there is a second action
          requested before the processing of the first action is finished,
          there could still be some corruption of the output string buffer
          used to build the result. (issue #8711, #8934)

2007-01-29 18:59 +0000 [r52572]  Joshua Colp <jcolp@digium.com>

        * apps/app_voicemail.c: Use ast_calloc instead of malloc.

2007-01-29 17:57 +0000 [r52535]  Steve Murphy <murf@digium.com>

        * apps/app_voicemail.c, main/say.c: this is for 8778 (pt_BR
          backport to 1.4). It was committed to trunk via 7663. But it
          wasn't so much an enhancement as a fix for the bad language
          output for portuguese in Brazil, so, after a lot of prodding from
          patient Brazilians, here is the same fix for 1.4

2007-01-29 17:33 +0000 [r52523]  Joshua Colp <jcolp@digium.com>

        * apps/app_voicemail.c: Set quota information to 0 when creating a
          vm_state. (issue #8924 reported by neutrino88)

2007-01-29 16:54 +0000 [r52506]  Russell Bryant <russell@digium.com>

        * main/jitterbuf.c, include/jitterbuf.h: Clean up a few things in
          the last commit to the adaptive jitterbuffer code. - Specifically
          indicate to the compiler that the "dropem" variable only needs
          one but. - Change formatting to conform to coding guidelines.

2007-01-29 04:18 +0000 [r52494]  Jim Dixon <telesistant@hotmail.com>

        * main/jitterbuf.c, include/jitterbuf.h: Fixed problem with
          jitterbuf, whereas it would not complain about, and would allow
          itself to be overfilled (per the max_jitterbuf parameter). Now it
          rejects any data over and above that size, and complains about
          it.

2007-01-28 05:15 +0000 [r52462]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * configure, configure.ac: Suggested change to fix normal usage of
          --with-tds=/usr/local (Sean Bright, via asterisk-dev mailing
          list)

2007-01-27 02:13 +0000 [r52335-52416]  Joshua Colp <jcolp@digium.com>

        * /, apps/app_queue.c: Merged revisions 52415 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r52415 | file | 2007-01-26 21:09:10 -0500 (Fri, 26 Jan 2007) | 2
          lines Make COMPLETECALLER and COMPLETEAGENT output to queue_log
          follow documentation. (issue #7677 reported by amilcar) ........

        * main/manager.c: Have the manager interface send back an "Already
          logged in" message instead of "Invalid/Unknown Command" when the
          client authenticates for a second time. (issue #8509 reported by
          pari)

        * /, channels/chan_iax2.c: Merged revisions 52360 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2
          lines Make the last context entry read in the dominant one.
          (issue #8918 reported by pj) ........

        * main/file.c: Fix core show file formats CLI command.

2007-01-25 19:18 +0000 [r52163-52265]  Joshua Colp <jcolp@digium.com>

        * /, main/jitterbuf.c: Merged revisions 52264 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r52264 | file | 2007-01-25 14:15:29 -0500 (Thu, 25 Jan 2007) | 2
          lines Allow dequeueing of frames with negative timestamp by
          moving jitterbuffer frames check to jb_next. (issue #8546
          reported by harmen) ........

        * channels/chan_sip.c: Drop out variables I accidentally put in.

        * channels/chan_sip.c: Decrement onHold count if we are hung up on
          and still on hold. (issue #8909 reported by alexh42)

        * apps/app_mixmonitor.c, /: Merged revisions 52162 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan
          2007) | 2 lines Add another note about audio files being played
          back to each bridged party. (issue #8718 reported by ppyy)
          ........

2007-01-25 01:37 +0000 [r52107-52160]  Russell Bryant <russell@digium.com>

        * apps/app_voicemail.c, configs/users.conf.sample: By suggestion
          from kpfleming last week, change "vmpassword" to "vmsecret".

        * configure, configure.ac: Remove libnsl as a required lib for
          libiksemel to work. This change was already made in the trunk.
          (issue #8762)

        * include/asterisk/dial.h: Fix the formatting of doxygen comments
          to properly indicate that the comment documents the previous
          entity, as opposed to the next one.

2007-01-24 18:26 +0000 [r52052]  Steve Murphy <murf@digium.com>

        * utils/check_expr.c, utils/Makefile, /: Merged revisions 52002 via
          svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r52002 | murf | 2007-01-24 10:43:50 -0700 (Wed, 24 Jan 2007) | 1
          line updated check_expr via 8322 (refactoring of expression
          checking impl); elfring contributed a nice code reorg, I
          contributed some time to get it working again, better messages
          ........

2007-01-24 18:20 +0000 [r52016-52049]  Joshua Colp <jcolp@digium.com>

        * main/dial.c (added), apps/app_page.c, main/Makefile,
          include/asterisk/dial.h (added): Merge in dialing API and the
          app_page that uses it. (issue #BE-118)

        * channels/chan_sip.c: Fix changing channel formats when joint
          capability changes and there are no audio formats... I didn't
          break it originally! (issue #8535 reported by ivoc)

2007-01-24 17:14 +0000 [r52000]  Russell Bryant <russell@digium.com>

        * configure: rebuild configure script to reflect last chan_h323
          related changes.

2007-01-24 12:57 +0000 [r51979-51989]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c: added fix from #8899

        * channels/chan_misdn.c, /: Merged revisions 51966 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r51966 | crichter | 2007-01-24 11:48:09 +0100 (Mi, 24
          Jan 2007) | 1 line fixed the busy problem (dialstatus was not
          busy when we called a busy extension) ........

2007-01-24 09:30 +0000 [r51931]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Show capabilities *and* preference in
          general settings in "sip show settings" (reported by Clona/Telio
          - Thanks!)

2007-01-24 08:04 +0000 [r51895]  Paul Cadach <paul@odt.east.telecom.kz>

        * acinclude.m4: Allow x64 builds of H.323 (please, rebuild
          configure)

2007-01-24 00:59 +0000 [r51829-51848]  Russell Bryant <russell@digium.com>

        * main/channel.c, /: Merged revisions 51843 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 Jan 2007) |
          6 lines Fix an issue related to synchronization of recordings
          when using Monitor(). The bug is a miscalculation of the amount
          to seek the stream for writing to disk when the number of samples
          coming in and out of a channel do not match up. (issue #8298,
          #8887, report and patch by guillecabeza, patch files created and
          testing done by whoiswes) ........

        * apps/app_while.c, /: Merged revisions 51828 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r51828 | russell | 2007-01-23 18:17:50 -0600 (Tue, 23 Jan 2007) |
          4 lines Don't set a new value for the END_ variable on the
          channel before using the old value. If you do, it will lead to
          accessing a memory address that has been free()'d. (issue #8895,
          arkadia) ........

2007-01-23 22:46 +0000 [r51788]  Joshua Colp <jcolp@digium.com>

        * channels/chan_oss.c, channels/chan_phone.c, channels/chan_zap.c,
          channels/chan_sip.c, channels/chan_skinny.c,
          channels/chan_features.c, channels/chan_alsa.c,
          channels/chan_gtalk.c, channels/chan_iax2.c: Update channel
          drivers to use module referencing so that unloading them while in
          use will not result in crashes. (issue #8897 reported by junky)

2007-01-23 22:04 +0000 [r51750-51781]  Russell Bryant <russell@digium.com>

        * main/manager.c: Fix some bugs in process_message(). The manager
          session lock needs to be held when sending some sort of response,
          or calling one of the manager action callbacks. This resolves an
          issue where people using the GUI would get random crashes when
          they start clicking around a lot. (issue #8711, reported and
          debugged by zandbelt)

        * main/http.c: Fix setting the default port of 8088 on 64-bit or
          big-endian machines.

        * main/manager.c: When traversing the list of manager actions, the
          iterator needs to be initialized to the list head *after* locking
          the list. Also, lock the actions list in one place it is being
          accessed where it was not being done.

2007-01-23 20:32 +0000 [r51683-51716]  Steve Murphy <murf@digium.com>

        * res/res_features.c: this mod from 8593 (dstchannel in cdr is
          empty when transfer call).

        * main/callerid.c: via 8748 (callerid.c loses name when returning
          PRIVATE_NUMBER flag), the user suggested this mod, saying it
          would allow 'WITHHELD' to appear in the name field, which would
          be useful

2007-01-23 10:28 +0000 [r51648-51649]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /,
          channels/misdn/isdn_msg_parser.c: Merged revisions 50495,50506
          via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r50495 | crichter | 2007-01-11 14:27:52 +0100 (Do, 11 Jan 2007) |
          6 lines * more additions to make the RESTART message work * added
          fix for misdn_call to allow SETUPs with empty extensions,
          replaced the strtok_r functions with strsep for that (inspired by
          Sandro Cappellazzo, thanks) ........ r50506 | crichter |
          2007-01-11 15:45:38 +0100 (Do, 11 Jan 2007) | 1 line when we get
          L2 UP, the L1 is UP definitely too, so we set the L1 state up as
          well. ........

        * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
          channels/chan_misdn.c: manually merged r49922 and r50335, because
          of conflicts. this commint includes addition of the ISDN RESTART
          Message

2007-01-23 06:51 +0000 [r51615]  Paul Cadach <paul@odt.east.telecom.kz>

        * channels/chan_h323.c, channels/Makefile: Do not abort Asterisk
          startup if h323 configuration file not found (reported by
          mithraen)

2007-01-23 03:00 +0000 [r51513-51558]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Only change audio formats on the channel if
          we have an audio format to change to. (issue #8535 reported by
          ivoc)

        * /, res/res_musiconhold.c: Merged revisions 51512 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r51512 | file | 2007-01-22 20:41:35 -0500 (Mon, 22 Jan
          2007) | 2 lines Yield before reading from zaptel timing source
          under Solaris so that other threads get a chance to do things.
          (issue #7875 reported by bob) ........

2007-01-22 19:28 +0000 [r51409]  Steve Murphy <murf@digium.com>

        * pbx/pbx_ael.c: This fixes 8836, according to dnatural

2007-01-22 19:13 +0000 [r51360-51407]  Joshua Colp <jcolp@digium.com>

        * apps/app_mixmonitor.c, /: Merged revisions 51406 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan
          2007) | 2 lines Move filestream creation to Mixmonitor loop. This
          will prevent a blank file from being created if no frames ever
          pass through to be recorded. (issue #7589 reported by
          steve_mcneil) ........

2007-01-20 06:53 +0000 [r51348-51350]  Jason Parker <jparker@digium.com>

        * configs/say.conf.sample: Fix Italian numeral support in say.conf
          for "_[2-9]00" case. "2131" would've translated to something
          along the lines of (pardon my..Italian {or lack thereof})
          "duecentocentotrentuno", which makes no sense at all.

        * configs/say.conf.sample: Fix German language support in say.conf
          Properly support 21, 31, 41, 51, 61, 71, 81, and 91.
          einundzwanzig has the same format as zweiundzwanzig (as do all
          other "_ZX" spoken numerals) Fix support for numbers in the
          10,000,000 to 99,999,999 range. Add support for numbers in the
          100,000,000 to 999,999,999 range.

2007-01-20 00:13 +0000 [r51302-51343]  Russell Bryant <russell@digium.com>

        * apps/app_meetme.c: Remove an unused instance of an unnamed enum.

        * apps/app_meetme.c: Remove another duplicated definition

        * apps/app_meetme.c: Remove a variable that was declared twice.

        * codecs/gsm/Makefile: Add a couple more processors that need
          optimizations excluded. (issue #8637)

        * channels/chan_gtalk.c: Fix VLDTMF support in chan_gtalk.
          AST_FRAME_DTMF and AST_FRAME_DTMF_END are actually the same
          thing. So, a digit would have been interpreted incorrectly here.
          Since the channel driver will always have the begin and end
          callbacks called for a digit, only support the button-down and
          button-up messages.

        * .cleancount: Bump the cleancount since my last commit changed the
          channel structure.

        * channels/chan_oss.c, main/rtp.c, main/channel.c,
          channels/chan_phone.c, channels/chan_misdn.c,
          channels/chan_skinny.c, channels/chan_features.c,
          channels/chan_h323.c, channels/chan_alsa.c, channels/chan_mgcp.c,
          channels/chan_zap.c, channels/chan_local.c, main/frame.c,
          channels/chan_sip.c, channels/chan_agent.c,
          include/asterisk/channel.h, channels/chan_gtalk.c,
          channels/chan_iax2.c: Merge the changes from the
          /team/group/vldtmf_fixup branch. The main bug being addressed
          here is a problem introduced when two SIP channels using SIP INFO
          dtmf have their media directly bridged. So, when a DTMF END frame
          comes into Asterisk from an incoming INFO message, Asterisk would
          try to emulate a digit of some length by first sending a DTMF
          BEGIN frame and sending a DTMF END later timed off of incoming
          audio. However, since there was no audio coming in, the DTMF_END
          was never generated. This caused DTMF based features to no longer
          work. To fix this, the core now knows when a channel doesn't care
          about DTMF BEGIN frames (such as a SIP channel sending INFO
          dtmf). If this is the case, then Asterisk will not emulate a
          digit of some length, and will instead just pass through the
          single DTMF END event. Channel drivers also now get passed the
          length of the digit to their digit_end callback. This improves
          SIP INFO support even further by enabling us to put the real
          digit duration in the INFO message instead of a hard coded 250ms.
          Also, for an incoming INFO message, the duration is read from the
          frame and passed into the core instead of just getting ignored.
          (issue #8597, maybe others...)

        * main/asterisk.c: Merged revisions 51300 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r51300 | russell | 2007-01-19 10:44:09 -0600 (Fri, 19 Jan 2007) |
          4 lines Fix a memory leak on command line tab completion. The
          container for the matches was freed, but the individual matches
          themselves were not. (issue #8851, arkadia) ........

2007-01-19 00:17 +0000 [r51272-51274]  Dwayne M. Hubbard <dhubbard@digium.com>

        * channels/chan_zap.c: chan_zap compiles without libpri after
          committing 7877 patch

        * channels/chan_zap.c, /: Merged revisions 51271 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 Jan 2007)
          | 3 lines issue 7877: chan_zap module reload does not use
          default/initialized values on subsequent loads. Reset
          configuration variables to default values prior to parsing
          configuration file. ........

2007-01-18 23:36 +0000 [r51270]  Kevin P. Fleming <kpfleming@digium.com>

        * /: block this patch since it is already here

2007-01-18 22:50 +0000 [r51265]  Jason Parker <jparker@digium.com>

        * apps/app_voicemail.c, main/channel.c, main/pbx.c,
          funcs/func_strings.c, main/app.c: Add some more checks for
          option_debug before ast_log(LOG_DEBUG, ...) calls. Issue 8832,
          patch(es) by tgrman

2007-01-18 21:54 +0000 [r51262]  Russell Bryant <russell@digium.com>

        * Makefile, configure, main/Makefile, acinclude.m4, makeopts.in:
          Ensure that the locations given to the Asterisk configure script
          for ncurses, curses, termcap, or tinfo are further passed along
          to the editline configure script. This fixes some
          cross-compilation environments. (issue #8637, reported by ovi,
          patch by me)

2007-01-18 21:14 +0000 [r51256]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * /, main/stdtime/localtime.c: Merged revisions 51255 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r51255 | tilghman | 2007-01-18 15:11:34 -0600 (Thu, 18
          Jan 2007) | 2 lines If a timezone is not specified, assume
          localtime (instead of gmtime) (Issue #7748) ........

2007-01-18 19:17 +0000 [r51251]  Joshua Colp <jcolp@digium.com>

        * apps/app_speech_utils.c: Only start timeout once we reach the end
          of the files to play back.

2007-01-18 18:42 +0000 [r51245]  Jason Parker <jparker@digium.com>

        * main/cli.c: Fix an issue with file name completion in "module
          load" and "load". Issue 8846

2007-01-18 18:36 +0000 [r51243]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Copy MOH settings when calling a peer so
          that if they put someone on hold or get put on hold themselves
          they get the right music class. (issue #8840 reported by mdu113)

2007-01-18 18:28 +0000 [r51241]  Jason Parker <jparker@digium.com>

        * main/channel.c: Fix an issue with deprecated commands

2007-01-18 17:49 +0000 [r51236]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * contrib/scripts/vmdb.sql, /: Merged revisions 51235 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r51235 | tilghman | 2007-01-18 11:42:17 -0600 (Thu, 18
          Jan 2007) | 2 lines Document all the fields, including the
          indication that "uniqueid" should not be renamed. ........

2007-01-18 17:18 +0000 [r51233]  Russell Bryant <russell@digium.com>

        * main/manager.c: Make the "hasmanager" option in users.conf
          actually have an effect. (issue #8740, LnxPrgr3)

2007-01-18 00:48 +0000 [r51211-51213]  Joshua Colp <jcolp@digium.com>

        * apps/app_voicemail.c: Build the IMAP remote directory string
          better and properly. Fix an issue with encoding the GSM voicemail
          when attaching to the voicemail. (issue #8808 reported by
          akohlsmith)

        * main/rtp.c: Pass data as well for hold/unhold/vidupdate frames.
          (issue #8840 reported by mdu113)

2007-01-17 23:31 +0000 [r51198-51205]  Russell Bryant <russell@digium.com>

        * funcs/func_odbc.c: Fix some instances where when loading
          func_odbc, a double-free could occur. Also, remove an unneeded
          error message. If the failure condition is actually a memory
          allocation failure, a log message will already be generated
          automatically.

        * channels/chan_zap.c: Instead of dividing the offset by 2
          directly, make it more clear that the offset is being scaled by
          the size of the elements in the buffer. (Inspired by a discussing
          on the asterisk-dev list about this code)

        * /, channels/chan_sip.c: Merged revisions 51197 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17 Jan 2007) |
          3 lines Move the check for a failure of ast_channel_alloc() to
          before locking the pvt structure again. Otherwise, on a failure,
          this will cause a deadlock. ........

2007-01-17 20:56 +0000 [r51195]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * /, main/utils.c: Merged revisions 51194 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r51194 | tilghman | 2007-01-17 14:52:21 -0600 (Wed, 17 Jan 2007)
          | 4 lines When ast_strip_quoted was called with a zero-length
          string, it would treat a NULL as if it were the quoting character
          (and would thus return the string in memory immediately following
          the passed-in string). ........

2007-01-17 17:36 +0000 [r51186]  Jason Parker <jparker@digium.com>

        * apps/app_voicemail.c: re-add "password" for realtime voicemail

2007-01-17 06:36 +0000 [r51182]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: Return the correct result when directly writing out a
          packet so that the core doesn't then decide to handle it the
          regular way again. (issue #8833 reported by rcourtna)

2007-01-17 01:29 +0000 [r51176]  Kevin P. Fleming <kpfleming@digium.com>

        * apps/app_voicemail.c: a few more coding style cleanups and one
          bug fix (from AnthonyL)

2007-01-17 00:46 +0000 [r51172]  Joshua Colp <jcolp@digium.com>

        * channels/chan_iax2.c: Move rescheduling of lagrq/pings into the
          scheduler callback.

2007-01-17 00:20 +0000 [r51165-51170]  Jason Parker <jparker@digium.com>

        * main/rtp.c: Fix issue with dtmf continuation packets when the
          dtmf digit is 0... Issue 8831

        * apps/app_voicemail.c, contrib/scripts/vmdb.sql: Fix an issue with
          IMAP storage and realtime voicemail. Also update the vmdb sql
          script for IMAP specific options. Issue 8819, initial patches by
          bsmithurst (slightly modified by me)

        * doc/voicemail_odbc_postgresql.txt: change documentation to
          reflect new procedure in 1.4/trunk

2007-01-16 21:51 +0000 [r51159-51162]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * /, doc/voicemail_odbc_postgresql.txt (added): Merged revisions
          51161 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r51161 | tilghman | 2007-01-16 15:50:04 -0600 (Tue, 16 Jan 2007)
          | 2 lines Add documentation walkthrough on getting Postgres to
          work with voicemail (from Issue 8513) ........

        * apps/app_voicemail.c, /: Merged revisions 51158 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r51158 | tilghman | 2007-01-16 15:26:06 -0600 (Tue, 16 Jan 2007)
          | 2 lines Postgres driver doesn't like a NULL pointer when
          retrieving the length (Bug 8513) ........

2007-01-16 17:46 +0000 [r51150]  Matt O'Gorman <mogorman@digium.com>

        * apps/app_voicemail.c: minor things i missed before i get jumped
          on

2007-01-16 17:39 +0000 [r51148]  Joshua Colp <jcolp@digium.com>

        * /, res/res_features.c: Merged revisions 51145 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r51145 | file | 2007-01-16 12:36:50 -0500 (Tue, 16 Jan 2007) | 2
          lines Return previous behavior. ParkedCalls will be able to do
          DTMF based transfers again. trunk however will get an option to
          allow this to be set on/off. (issue #8804 reported by nortex)
          ........

2007-01-16 17:36 +0000 [r51146]  Jason Parker <jparker@digium.com>

        * main/file.c: Display more useful output when streaming files.
          Include the channel name to which the file is being played. Issue
          8828, patch by junky.

2007-01-16 05:55 +0000 [r51087]  Joshua Colp <jcolp@digium.com>

        * channels/chan_zap.c, /: Merged revisions 51085 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2
          lines Add none as a valid callgroup/pickupgroup option. I
          consider it a bug that it would inherit it all the way down and
          not have any way to reset it to nothing - so that's why it is in
          1.2. (issue #8296 reported by gkloepfer) ........

2007-01-16 01:15 +0000 [r51057]  Russell Bryant <russell@digium.com>

        * main/config.c: It is possible for the config pointer to be NULL
          here, so it needs to be checked before dereferencing it.

2007-01-16 00:22 +0000 [r51030]  Matt O'Gorman <mogorman@digium.com>

        * apps/app_voicemail.c, configs/users.conf.sample: Patch allows for
          changing voicemail password in users.conf from voicemail main,
          written by AnthonyL bug #8436

2007-01-15 23:49 +0000 [r50994]  Russell Bryant <russell@digium.com>

        * Makefile.rules: Filter out a few CFLAGS that are not valid
          CXXFLAGS.

2007-01-15 21:08 +0000 [r50957]  Matt O'Gorman <mogorman@digium.com>

        * apps/app_voicemail.c, /: Merged revisions 50946 via svnmerge from
          https://svn.digium.com/svn/asterisk/branches/1.2 ........ r50946
          | mogorman | 2007-01-15 14:44:53 -0600 (Mon, 15 Jan 2007) | 4
          lines Solves issue with forwarding voicemails from folders other
          than inbox. patch by anthonyl. ........

2007-01-15 18:23 +0000 [r50921]  Jason Parker <jparker@digium.com>

        * main/asterisk.c: re-add deprecated "show version" CLI command.

2007-01-15 16:36 +0000 [r50895]  Joshua Colp <jcolp@digium.com>

        * main/manager.c: Move event processing into do_message so that it
          gets executed again when events are tripped.

2007-01-15 15:03 +0000 [r50867]  Kevin P. Fleming <kpfleming@digium.com>

        * configure, include/asterisk/autoconfig.h.in, main/Makefile,
          configure.ac, Makefile.rules, acinclude.m4, makeopts.in: use the
          ACX_PTHREAD macro from the Autoconf macro archive for setting up
          compiler pthreads support... should improve portability to
          platforms with unusual pthreads requirements

2007-01-14 21:59 +0000 [r50820]  Joshua Colp <jcolp@digium.com>

        * main/astmm.c: Add missing newlines for two memory CLI commands.

2007-01-14 05:13 +0000 [r50782]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * main/db1-ast/db/db.c, main/db1-ast/recno/rec_get.c,
          main/db1-ast/btree/bt_seq.c, main/db1-ast/hash/hash_func.c,
          main/db1-ast/btree/bt_utils.c, main/db1-ast/recno/rec_seq.c,
          main/db1-ast/btree/bt_overflow.c, main/db1-ast/btree/bt_search.c,
          main/db1-ast/btree/bt_conv.c, main/db1-ast/btree/bt_close.c,
          main/db1-ast/btree/bt_put.c, main/db1-ast/recno/rec_utils.c,
          main/db1-ast/recno/rec_open.c, main/db1-ast/hash/hash_bigkey.c,
          main/db1-ast/recno/rec_delete.c, main/db1-ast/hash/hash_buf.c,
          main/db1-ast/hash/hash_page.c, main/db1-ast/recno/rec_close.c,
          main/db1-ast/recno/rec_put.c, main/db1-ast/include/ndbm.h,
          main/db1-ast/btree/bt_debug.c, main/db1-ast/mpool/mpool.c,
          main/db1-ast/btree/bt_split.c, main/db1-ast/btree/bt_open.c,
          main/db1-ast/btree/bt_delete.c, main/db1-ast/hash/hash_log2.c,
          main/db1-ast/hash/hsearch.c, /, main/db1-ast/btree/bt_page.c,
          main/db1-ast/recno/rec_search.c, main/db1-ast/btree/bt_get.c,
          main/db1-ast/hash/hash.c: Merged revisions 50781 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r50781 | tilghman | 2007-01-13 23:01:16 -0600 (Sat, 13
          Jan 2007) | 2 lines Bug 8814 - db should look for its header
          using a relative path, instead of the system path (Fixes FreeWRT)
          ........

2007-01-13 16:45 +0000 [r50754]  Kevin P. Fleming <kpfleming@digium.com>

        * Makefile, build_tools/make_sample_voicemail (added): when
          building the sample greetings for maibox 1234@default during
          'make samples', build a greeting for each language and file
          format the user selected to install with menuselect (reported by
          Brian Capouch on asterisk-dev)

2007-01-13 06:00 +0000 [r50674-50727]  Joshua Colp <jcolp@digium.com>

        * main/channel.c: Only write a frame out to the channel if one
          exists. There are cases where one may not and would therefore
          cause the channel driver to segfault. (issue #8434 reported by
          slimey)

        * res/res_snmp.c: Only join the snmp thread on an unload if the
          thread is actually running. (issue #8810 reported by junky)

2007-01-12 19:24 +0000 [r50647]  Jason Parker <jparker@digium.com>

        * configs/voicemail.conf.sample: Update documentation to state that
          you shouldn't use realtime static with voicemail.conf

2007-01-12 16:42 +0000 [r50602]  Joshua Colp <jcolp@digium.com>

        * main/manager.c: We need to check for res being 0 in do_message
          itself, otherwise our headers will get lost.

2007-01-12 14:42 +0000 [r50562]  Kevin P. Fleming <kpfleming@digium.com>

        * main/pbx.c, /: Merged revisions 50561 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r50561 | kpfleming | 2007-01-12 08:34:15 -0600 (Fri, 12 Jan 2007)
          | 2 lines minor documentation clarification ........

2007-01-11 05:53 +0000 [r50377-50468]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Remove check for channel state as it can
          definitely be something other then ring, and also clean up the
          code a bit. This should solve the parking issues and maybe some
          attended transfer issues people have been seeing.

        * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add
          support to see whether NAT was detected (yay symmetric RTP) and
          also add a check in chan_sip so that if NAT has been detected and
          the reinvite behind nat option has been turned off, then just do
          partial bridge. (issue #8655 reported by mnicholson)

        * apps/app_speech_utils.c: Merge speech-multi branch which adds
          support for joining multiple sound files together to be played
          one after another in SpeechBackground.

        * main/config.c: Fix parsing when using something like ldap
          settings. (done by anthonyl)

        * channels/chan_sip.c: Fix chan_sip not working issue. Let's not
          prematurely return 0. (issue #8783 reported by st41ker)

2007-01-10 16:45 +0000 [r50346]  Jason Parker <jparker@digium.com>

        * cdr/cdr_manager.c: Reverse some logic in cdr_manager, which made
          it fail to load if the config file existed. Issue 8777

2007-01-10 04:55 +0000 [r50266-50298]  Joshua Colp <jcolp@digium.com>

        * apps/app_dial.c, /: Merged revisions 50295 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r50295 | file | 2007-01-09 23:51:06 -0500 (Tue, 09 Jan 2007) | 2
          lines Add another return value to dial_exec_full that indicates
          execution is going to continuing at a new
          extension/context/priority and to just let it slide. (issue #8598
          reported by jon) ........

        * main/pbx.c: Ensure data's existence before trying to access it.
          (issue #8774 reported by rcourtna)

2007-01-10 02:17 +0000 [r50228]  Russell Bryant <russell@digium.com>

        * Makefile, /: Merged revisions 50227 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r50227 | russell | 2007-01-09 21:16:45 -0500 (Tue, 09 Jan 2007) |
          6 lines Make the number that represents the major version number
          a single digit instead of 2. Using two digits makes it an octal
          number when put into version.h, which breaks the compilation of
          any out of tree module that checks the version for any version
          after 1.2.7 (reported by Matteo Brancaleoni on the asterisk-dev
          mailing list, who gave credit to vihai for pointing it out)
          ........

2007-01-09 17:11 +0000 [r50186]  Jason Parker <jparker@digium.com>

        * main/cli.c: Re-add CLI command that should have only been
          deprecated in 1.4. Thanks kshumard! (reported in person, so no
          associated issue #)

2007-01-09 13:40 +0000 [r50151]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_voicemail.c, /: Merged revisions 50150 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r50150 | tilghman | 2007-01-09 07:30:04 -0600 (Tue, 09 Jan 2007)
          | 4 lines The advent of realtime has enabled people to use commas
          in the fullname field. This could cause an issue with sending
          voicemails, when the field is unquoted. (Issue 8595) ........

2007-01-09 11:25 +0000 [r50124]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: - handle re-invites properly in sip_hangup()
          - Add some invitestate status changes just to be sure

2007-01-08 23:39 +0000 [r50098]  Jason Parker <jparker@digium.com>

        * apps/app_voicemail.c: Fix an issue with voicemail and users.conf,
          where it wouldn't ever parse a password, since it was using
          "secret" instead of "password" Issue 8761, reported by and patch
          suggestion from ssokol.

2007-01-08 21:11 +0000 [r50073]  Matt O'Gorman <mogorman@digium.com>

        * apps/app_senddtmf.c: we can't unlock a channel if we cant find
          it. - AnthonyL bug #8741

2007-01-08 18:21 +0000 [r50032]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: Disable the more intense packet2packet bridging until
          the bugs can be worked out.

2007-01-08 14:26 +0000 [r49925-50006]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Issue #8677 - Handle failure of T.38
          re-invite This is not a fix, but adding an error message to tell
          the admin that we have a bad configuration. We should not send
          T.38 re-invites to devices that can't handle it (with the current
          architecture where you have to hard-code t.38 support per
          device). To really fix this, we need to figure out a way to tell
          the incoming call that the re-invite failed, so we can signal
          failure on that end and go back to the original call.

        * channels/chan_sip.c: Issue #8524, support multiple via header
          values (tardieu) Thanks!

        * channels/chan_sip.c: We only need one forward declaration

        * channels/chan_sip.c: Issue 8735: Terminate state when extension
          is unavailable for subscription

2007-01-08 05:11 +0000 [r49890]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_iax2.c: Merged revisions 49889 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2
          lines Ensure we use the default refresh value of 60 if the remote
          server does not send one. (issue #8746 reported by maethor)
          ........

2007-01-08 03:53 +0000 [r49866]  Kevin P. Fleming <kpfleming@digium.com>

        * configure, configure.ac: since we use AC_PATH_TOOL to find tools,
          we should use the results it provides for us (reported by Brian
          Capouch on the asterisk-dev list)

2007-01-07 21:44 +0000 [r49831-49834]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * /, apps/app_dictate.c: Merged revisions 49833 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r49833 | tilghman | 2007-01-07 15:43:10 -0600 (Sun, 07 Jan 2007)
          | 2 lines If openstream fails, then we crash (Issue 8564)
          ........

        * channels/chan_sip.c: Second condition was a subset of the first,
          so hold was never decremented, thus hint stayed stuck (Issue
          8747)

2007-01-06 00:24 +0000 [r49742]  Jason Parker <jparker@digium.com>

        * main/pbx.c, res/res_features.c, pbx/pbx_config.c: Save 1 whopping
          byte of allocated memory! This looks like it may have been a
          chicken/egg scenario.. You had to call a cleanup func, because
          everything was allocated. Then since you had to call a cleanup
          func, you were forced to allocate - ie; strdup("").

2007-01-05 23:51 +0000 [r49710-49715]  Kevin P. Fleming <kpfleming@digium.com>

        * configure, acinclude.m4: one more time...

        * configure, acinclude.m4: proper fix for r49712

        * configure, acinclude.m4: if --with-foo=<path> is specific for a
          configure option, ensure that it is used for header file checking
          as well

        * main/manager.c: ast_func_read() needs a writable copy of the
          function name to be passed

2007-01-05 23:16 +0000 [r49705]  Jason Parker <jparker@digium.com>

        * channels/chan_zap.c, codecs/codec_zap.c: Make codec_zap and
          chan_zap also depend on zaptel. This fixes an issue (8727) with
          zaptel being in a different directory, using --with-zaptel.

2007-01-05 22:52 +0000 [r49676-49680]  Kevin P. Fleming <kpfleming@digium.com>

        * main/manager.c: don't 'consume' the params list before we try to
          use it again

        * res/res_monitor.c, main/config.c, apps/app_setcdruserfield.c,
          main/manager.c, include/asterisk/jabber.h, apps/app_senddtmf.c,
          main/db.c, channels/chan_zap.c, channels/chan_sip.c,
          apps/app_meetme.c, res/res_features.c, channels/chan_agent.c,
          utils/astman.c, include/asterisk/manager.h, channels/chan_iax2.c,
          apps/app_queue.c, res/res_jabber.c: reduce stack consumption for
          AMI and AMI/HTTP requests by nearly 20K in most cases

2007-01-05 22:14 +0000 [r49675]  Joshua Colp <jcolp@digium.com>

        * main/channel.c: Don't keep repeating the warning over and over
          when the end of the call is reached. (issue #8724 reported by
          xrg)

2007-01-05 17:09 +0000 [r49581-49636]  Kevin P. Fleming <kpfleming@digium.com>

        * /, channels/chan_sip.c, channels/chan_skinny.c,
          channels/chan_iax2.c: Merged revisions 49635 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007)
          | 2 lines ensure that threads which are supposed to be detached
          (because we aren't going to wait on them) are created properly
          ........

        * channels/chan_iax2.c: revert the dynamic_list insertion change...
          that was not the right thing to do

        * channels/chan_iax2.c: create the IAX2 processing threads as
          background threads so they will use smaller stacks when we create
          a dynamic thread, put it on the dynamic_list right away so we
          don't lose track of it

2007-01-04 23:00 +0000 [r49568]  Joshua Colp <jcolp@digium.com>

        * channels/chan_iax2.c: It's possible for the iax2 pvt to
          disappear, so if it has... don't bother looking for dpentries.

2007-01-04 22:51 +0000 [r49553]  Kevin P. Fleming <kpfleming@digium.com>

        * include/asterisk/threadstorage.h, main/asterisk.c,
          build_tools/cflags.xml, include/asterisk.h, main/Makefile,
          main/threadstorage.c (added), main/utils.c: add support for
          tracking thread-local-storage objects that exist via
          'threadstorage' CLI commands

2007-01-04 22:28 +0000 [r49551]  Joshua Colp <jcolp@digium.com>

        * main/config.c: Only free comments and line buffer once we reach
          the first level. (issue #8678 reported by ssokol, fixed by
          anthonyl)

2007-01-04 21:58 +0000 [r49460-49536]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/iax2-parser.c, main/frame.c: don't mark these
          allocations as 'cache' allocations when caching has been disabled

        * channels/iax2-parser.c: if we're going to decrement the frame
          count when we free a frame, we should inrement it when we create
          one :-)

        * channels/iax2-parser.c, channels/iax2-parser.h,
          channels/chan_iax2.c: only do IAX2 frame caching for voice and
          video frames

        * main/frame.c: don't do frame header caching in the core if
          LOW_MEMORY is defined

        * channels/iax2-parser.c: don't define this type either if
          LOW_MEMORY is enabled

2007-01-04 18:11 +0000 [r49459]  Matt O'Gorman <mogorman@digium.com>

        * apps/app_voicemail.c, /: Merged revisions 49447 via svnmerge from
          https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49447
          | mogorman | 2007-01-04 11:45:16 -0600 (Thu, 04 Jan 2007) | 2
          lines converted a lot of 256 to PATH_MAX and some white space
          fixes. ........

2007-01-04 18:06 +0000 [r49457-49458]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/iax2-parser.c: don't do frame caching in LOW_MEMORY mode

        * codecs/Makefile: make building of codec_gsm against the system
          GSM library actually work

2007-01-04 16:50 +0000 [r49413]  Matt O'Gorman <mogorman@digium.com>

        * apps/app_voicemail.c, /: Merged revisions 49412 via svnmerge from
          https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49412
          | mogorman | 2007-01-04 10:48:43 -0600 (Thu, 04 Jan 2007) | 3
          lines good catch russell sorry i missed that. fix magic number
          with proper sizeof ........

2007-01-04 04:33 +0000 [r49388]  Russell Bryant <russell@digium.com>

        * funcs/func_realtime.c: Fix the REALTIME() dialplan function.
          ast_build_string() advances the string pointer to the position to
          begin the next write into the buffer. So, this pointer can not be
          used to copy the contents of the string later. The beginning of
          the buffer must be saved. Interestingly enough, this code could
          not have ever worked. (Pointed out by Sebb on IRC, thanks!)

2007-01-03 23:32 +0000 [r49355]  Matt O'Gorman <mogorman@digium.com>

        * apps/app_voicemail.c, /: Merged revisions 49354 via svnmerge from
          https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49354
          | mogorman | 2007-01-03 17:22:47 -0600 (Wed, 03 Jan 2007) | 6
          lines When using ODBC_STORAGE VoicemailMain doesn't create the
          subdirectories for a mailbox such as the INBOX directory. this
          patch solves that problem, was written by anthony be-125 ........

2007-01-03 09:06 +0000 [r49313]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/misdn_config.c,
          doc/misdn.txt, channels/misdn/isdn_lib.h, channels/chan_misdn.c,
          /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c,
          configs/misdn.conf.sample: Merged revisions
          48319,48321,48467,48552,48576,49135,49303 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) |
          1 line changed a few debugs to higher debug levels ........
          r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) |
          1 line added the export and import of the MISDN_ADDRESS_COMPLETE
          Variable to inidcate wether the extension is already completely
          dialed or if there might come additional digits by information
          elements. also added some docs for that. ........ r48467 |
          crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line
          removed FIXUP state. added check for channel allocation conflict
          when we create a setup while the other site creates a setup on
          the same channel, besides the check we resolve this conflict.
          ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18
          Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a
          preselected channel we just accept it, even when we're NT. added
          some checks for segfaults. ........ r48576 | crichter |
          2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we
          reject a channel, because it's in use already, we shouldn't
          process the setup anymore. made the channel allocation a bit
          easier and more understandable, removed a few unused lines
          ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02
          Jan 2007) | 1 line added check for channel ranges in the
          set/empty channel functions. set pmp_l1_check default to no.
          added misdn restart pid cli command. added cleaning of channel
          when we send a RELEASE_COMPLETE. ........ r49303 | crichter |
          2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added
          check for bridging in misdn_call to avoid setting
          echocancellation when 2 mISDN channels are involved and when
          bridging is set. That lead to a kernel panic before under
          different situations, because we switched about 2 times between
          hardware bridging and echocancelation * readded MISDN_URATE
          variable which got lost before, this should make app_v110 work
          again * fixed typo ........

2007-01-03 03:21 +0000 [r49282]  Kevin P. Fleming <kpfleming@digium.com>

        * Makefile, Makefile.rules: various Makefile improvements to get
          chan_vpb (and any other C++ modules) to build properly

2007-01-03 01:19 +0000 [r49259]  Joshua Colp <jcolp@digium.com>

        * channels/chan_iax2.c: Check pvt structure presence before passing
          to send_command. This gets rid of the irritating message about a
          packet without pvt structure. This happens because the scheduled
          item is getting cancelled at almost the exact moment it is
          getting executed.

2007-01-02 22:30 +0000 [r49237]  Steve Murphy <murf@digium.com>

        * main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c,
          pbx/ael/ael.flex: This is a slight modification to Josh's edits
          for #8579; both files edited were the produced by flex; so the
          source files need to be changed instead, and the generated files
          regenerated.

2007-01-02 19:58 +0000 [r49212]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Small cleanup of add_t38sdp - it's always
          enabled at that point in the code

2007-01-02 17:33 +0000 [r49189]  Jason Parker <jparker@digium.com>

        * main/pbx.c: Allow fractions of a second in the Wait()
          application, like it says it allows.

2007-01-02 13:59 +0000 [r49165]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_zap.c: remove comment that is unrelated to this
          function

2007-01-02 12:08 +0000 [r49145]  Olle Johansson <oej@edvina.net>

        * configs/features.conf.sample: Adding note on effect of
          applicationmap features on re-invites

2007-01-01 23:34 +0000 [r49098-49102]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_zap.c, build_tools/menuselect-deps.in, configure,
          configure.ac, codecs/codec_zap.c: check specifically for VLDTMF
          and transcoding support in the system's Zaptel installation, and
          make only the modules that need those features dependent on them
          (this will allow building the other Zaptel-using parts of
          Asterisk against older versions of Zaptel or those on other
          platforms that haven't caught up yet to the Linux version)

        * Makefile: use a simpler (and portable) method to ensure that
          menuselect is built as a host binary

        * Makefile: revert this change until a better solution can be
          found... 'env -i' was not being used properly, but even when
          changed to do so, this process fails during cross-compilation
          because the menuselect build still sees 'CC' as set to the
          cross-compiler

2007-01-01 20:14 +0000 [r49096]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: remove incomplete implementation of dnsmgr.
          Let's fix this in trunk.

2006-12-30 18:31 +0000 [r49063-49073]  Joshua Colp <jcolp@digium.com>

        * pbx/pbx_config.c: IAX has been deprecated for quite some time so
          we had better use IAX2 when creating the dial string for users.
          (issue #8697 reported by ssokol)

        * channels/chan_zap.c: Use asprintf to build the channel names
          instead of custom function. I believe the custom function is
          doing some things that are not portable across all
          implementations. (issue #8570 reported by hterag & issue #8692
          reported by nicolasg)

        * main/rtp.c: If the Packet2Packet bridge is being broken because
          of a masquerade then attempt to read a frame in so the masquerade
          actually happens. Otherwise weirdness will occur. (issue #8696
          reported by kjotte)

        * channels/chan_iax2.c: Initialize the packet queue in load_module
          instead of just declaring the list with the default value. (issue
          #8695 reported by ssokol)

2006-12-30 00:40 +0000 [r49061]  Steve Murphy <murf@digium.com>

        * pbx/pbx_ael.c: A fix for 8661, where the CUT func needed to have
          comma args converted to vertical bars. I hope this change does
          little harm.

2006-12-29 00:50 +0000 [r49042-49048]  Kevin P. Fleming <kpfleming@digium.com>

        * /: put this value into the correct property

        * /, BUGS: Merged revisions 49045 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r49045 | kpfleming | 2006-12-28 18:32:32 -0600 (Thu, 28 Dec 2006)
          | 2 lines location of the bug posting guidelines has changed
          ........

        * sample.call: simple commit to test CIA integration

2006-12-28 21:26 +0000 [r49032-49035]  Jason Parker <jparker@digium.com>

        * main/cli.c: Fix some deprecated commands. Issue 8682, patch by me

        * main/http.c: saw this in passing... fix a small typo

2006-12-28 20:08 +0000 [r49028]  Kevin P. Fleming <kpfleming@digium.com>

        * sounds/Makefile: new versions of sounds

2006-12-28 19:52 +0000 [r49024]  Jason Parker <jparker@digium.com>

        * main/http.c: make the uris_lock a rwlock instead of a mutex lock
          - needs to be forward ported to trunk

2006-12-28 19:43 +0000 [r49022]  Joshua Colp <jcolp@digium.com>

        * configure, include/asterisk/autoconfig.h.in, configure.ac,
          include/asterisk/lock.h: Backport support for read/write locks.

2006-12-28 19:21 +0000 [r49020]  Steve Murphy <murf@digium.com>

        * main/ast_expr2.fl, main/ast_expr2.c, main/frame.c,
          pbx/ael/ael.tab.c, main/ast_expr2.y, main/ast_expr2f.c,
          pbx/ael/ael_lex.c, include/asterisk/ael_structs.h,
          pbx/ael/ael.tab.h, utils/ael_main.c: removed <err.h> as in trunk
          from the ael stuff. Also, threw in a minor fix to frame.c to
          avoid build-killing compiler warnings.

2006-12-27 22:28 +0000 [r49009]  Joshua Colp <jcolp@digium.com>

        * main/ast_expr2f.c, pbx/ael/ael_lex.c: ast_copy_string is not
          available when LOW_MEMORY is used and things are being built in
          the utils directory, so we need to resort to the old method of
          strncpy. (issue #8579 reported by mottano)

2006-12-27 22:06 +0000 [r48998-49006]  Kevin P. Fleming <kpfleming@digium.com>

        * main/enum.c, main/asterisk.c, main/rtp.c, main/term.c,
          main/cdr.c, main/channel.c, main/udptl.c, main/pbx.c,
          main/dnsmgr.c, main/frame.c, main/manager.c, main/file.c,
          main/http.c, main/logger.c: since these variables all have static
          duration, none of them need initializers (they default to zero
          anyway)

        * include/asterisk/options.h, main/asterisk.c, main/file.c: move
          extern declaration for this option to a header file where it
          belongs provide an initial value for 'languageprefix' option,
          instead of relying on randomness to provide a useful value

2006-12-27 21:06 +0000 [r48993-48997]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Only include acl.h and lock.h once

        * channels/chan_sip.c: Only set rfc2833compensate flag once
          (handle_common_options)

        * channels/chan_sip.c: - Remove checking for T38 options twice.
          Keeping them in handle_common_options

2006-12-27 18:33 +0000 [r48987-48988]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_sip.c: make the option actually match the
          documentation

        * channels/iax2-parser.c, include/asterisk/utils.h,
          include/asterisk/astmm.h, main/frame.c, main/astmm.c: allow 'show
          memory' and 'show memory summary' to distinguish memory
          allocations that were done for caching purposes, so they don't
          look like memory leaks

2006-12-27 17:59 +0000 [r48975-48985]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c, configs/sip.conf.sample: Be a bit more
          politically correct

        * channels/chan_sip.c, configs/sip.conf.sample: Issue #8575 - Buggy
          cisco MWI support. Normally we try not to change our software for
          bugs in other devices. But in this case, the Cisco phones are so
          widespread so we try to implement a fix while waiting for a
          bugfix from Cisco.

        * channels/chan_sip.c: - Make sure handle_common_options return 1
          when we found a common option - Move uncommon (only global)
          option away from handle_common_options Reported by rizzo. Thanks!

        * channels/chan_sip.c: Issue 8599 (rizzo) Change invitestate before
          re-sending invite with auth.

        * /, channels/chan_sip.c: Fix bogus content-length in t38 sdp.
          (rizzo, #8600)

2006-12-26 05:20 +0000 [r48960-48966]  Joshua Colp <jcolp@digium.com>

        * apps/app_meetme.c: Get rid of a needless memory allocation and
          only create a conference structure in find_conf_realtime if data
          was read from realtime. (issue #8669 reported by robl)

        * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add an
          API call that initializes an RTP structure. We need this because
          chan_sip is cheeky and uses a temporary RTP structure for codec
          purposes, and the API calls that are used rely on the lock.
          (Pointed out on asterisk-dev by Andy Wang)

        * configure, configure.ac: Clean up autoconf file (gets rid of
          warnings seen when rebuilding configure) and rebuild configure.

2006-12-25 05:21 +0000 [r48931-48956]  Russell Bryant <russell@digium.com>

        * /, funcs/func_math.c: Merged revisions 48955 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r48955 | russell | 2006-12-25 00:19:48 -0500 (Mon, 25 Dec 2006) |
          6 lines Fix an error introduced by copying and pasting the
          handling of the >= operator for the MATH function. If a single
          equal sign was used as an operator, the function would treat it
          is as if it were the >= operator. Now, it properly handles it as
          an invalid operator. (issue #8665, patch by tempest1) ........

        * channels/chan_oss.c: Fix a typo in an error message that
          indicated that the MGCP channel type could not be registered,
          instead of the correct type, OSS.

        * /, channels/chan_iax2.c: Merged revisions 48943 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24 Dec 2006) |
          3 lines Check for the proper return value on an error in a call
          to mmap(). This was reported by Andy Wang on the asterisk-dev
          list. Thanks! ........

        * /, channels/chan_sip.c: Merged revisions 48939 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 Dec 2006) |
          3 lines Remove a couple of misplaced dots in log messages. This
          was reported by Andrea Spadaccini on the asterisk-dev mailing
          list. ........

        * main/http.c: Implement locking for the list of URI handlers to
          make it thread-safe.

2006-12-23  Kevin P. Fleming  <kpfleming@digium.com>

        * Asterisk 1.4.0 released.

2006-12-22 22:33 +0000 [r48870-48906]  Jason Parker <jparker@digium.com>

        * Makefile, main/stdtime/localtime.c: Minor fixes for Solaris.

        * channels/chan_skinny.c: Fix for issue 7774 - patch by alamantia

2006-12-21 20:26 +0000 [r48783]  Joshua Colp <jcolp@digium.com>

        * /, redhat/asterisk.spec: Merged revisions 48782 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r48782 | file | 2006-12-21 15:25:01 -0500 (Thu, 21 Dec 2006) | 2
          lines Add new silence sound files to the spec for Redhat. (issue
          #8652 reported by alvaro_palma_aste) ........

2006-12-20 02:56 +0000 [r48592-48637]  Joshua Colp <jcolp@digium.com>

        * apps/app_voicemail.c: vms doesn't exist on non-IMAP storage
          builds.

        * apps/app_voicemail.c: Pass 'vms' pointer to record_and_review so
          it is then passed to the IMAP store file function. (issue #8614
          reported by punknow)

        * doc/snmp.txt: find is not the same as bind when it comes to
          documentation. (issue #8626 reported by johann8384)

2006-12-19 21:28 +0000 [r48586]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/Makefile: suppress compiler warnings in this module
          until it can be improved

2006-12-19 21:12 +0000 [r48585]  Joshua Colp <jcolp@digium.com>

        * apps/app_dial.c, /: Merged revisions 48584 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r48584 | file | 2006-12-19 16:10:26 -0500 (Tue, 19 Dec 2006) | 2
          lines Free localuser structure when we fail to dial (issue #8612
          reported by rizzo) ........

2006-12-19 21:03 +0000 [r48583]  Luigi Rizzo <rizzo@icir.org>

        * apps/app_sms.c: fix a bogus datalen in the frames generated by
          app_sms (causing noisy output if you listen to the output!) This
          affects trunk as well, whereas 1.2 is ok.

2006-12-19 14:57 +0000 [r48577]  Kevin P. Fleming <kpfleming@digium.com>

        * res/res_config_odbc.c, funcs/func_odbc.c: use the proper variable
          type for these unixODBC API calls, eliminating warnings on 64-bit
          platforms that use the 'new' 64-bit types for ODBC API calls

2006-12-19 03:46 +0000 [r48571]  Joshua Colp <jcolp@digium.com>

        * Makefile: Use env -i to start a fresh environment when going to
          build menuselect. This is more portable then using unset. (issue
          #8543 reported by jtodd)

2006-12-18 17:23 +0000 [r48566]  Luigi Rizzo <rizzo@icir.org>

        * include/asterisk/channel.h: unbreak the macro used for
          incrementing the frame counters. I don't know when the bug was
          introduced, but with the typical usage c->fin =
          FRAMECOUNT_INC(c->fin) the frame counters stay to 0. affects
          trunk as well (fix coming).

2006-12-18 17:15 +0000 [r48564]  Joshua Colp <jcolp@digium.com>

        * channels/chan_iax2.c: Put thread into proper list if we abort
          handling due to an error, and also hold the lock while putting it
          back into the proper idle list so we don't prematurely get a
          signal. (issue #8604 reported by arkadia)

2006-12-18 11:59 +0000 [r48513-48554]  Kevin P. Fleming <kpfleming@digium.com>

        * codecs/lpc10/Makefile, main/Makefile, codecs/gsm/Makefile,
          utils/astman.c, utils/smsq.c, codecs/ilbc/Makefile,
          utils/ael_main.c: remove some now-unnecessary explicit includes
          of autoconfig.h clean up per-file dependencies during 'make
          clean'

        * build_tools/prep_tarball: need an additional argument here to
          make the downloads actually occur

        * configure, include/asterisk/autoconfig.h.in, configure.ac,
          acinclude.m4: use m4 quoting for AC_MSG_NOTICE calls, to keep
          these calls from thinking they have multiple arguments

        * codecs/ilbc, formats, utils/Makefile, agi/Makefile, Makefile,
          funcs, build_tools/mkdep (removed), codecs/lpc10, main/db1-ast,
          main, codecs/gsm, pbx, res, channels, codecs, utils, agi,
          main/Makefile, apps, Makefile.moddir_rules, Makefile.rules, cdr:
          simplify dependency tracking system, using the compiler's
          built-in method for generating them, and only doing dependency
          tracking if developer mode is enabled via the configure script

        * Makefile, include/asterisk.h, main/stdtime/localtime.c: since we
          really, really have to have autoconfig.h included before all
          other headers (especially system headers), the Makefile will now
          force it to happen (this will fix build problems with files like
          ast_expr2f.c, where we can't control the inclusion order in the
          file itself)

        * funcs/func_curl.c: instead of initializing the curl library every
          time the CURL() function is invoked, do it only once per thread
          (this allows multiple calls to CURL() in the dialplan for a
          channel to run much more quickly, and also to re-use connections
          to the server) (thanks to JerJer for frequently complaining about
          this performance problem)

2006-12-15 19:55 +0000 [r48502-48506]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: Turn payload_lock into bridge_lock and make it
          encompass all RTP structure contents that may relate to bridge
          information, including who we are bridged to.

        * channels/chan_iax2.c: Hold call structure lock in places where a
          qualify or peer action can destroy it.

        * channels/chan_iax2.c: Lock network retransmission queue in all
          places that it is used.

2006-12-15 10:55 +0000 [r48481-48487]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Issue #8592 - treat 504 as 503 (imported
          from 1.2)

        * channels/chan_sip.c: Update to latest IANA spec

2006-12-15 06:28 +0000 [r48461-48478]  Joshua Colp <jcolp@digium.com>

        * channels/chan_iax2.c: Use a wakeup variable so that we don't wait
          on IO indefinitely if packets need to be retransmitted.

        * main/rtp.c, include/asterisk/rtp.h: Payload values on the RTP
          structure can change AFTER a bridge has started. This comes from
          the packet handling of the SIP response when indication that it
          was answered has been sent. Therefore we need to protect this
          data with a lock when we read/write. (issue #8232 reported by
          tgrman)

        * main/rtp.c: Remove direct RTCP bridging. I've come to the
          conclusion that we should handle this through the core and not
          just forward it on. Should solve a few bugs.

2006-12-12  Kevin P. Fleming  <kpfleming@digium.com>

        * Asterisk 1.4.0-beta4 released.

2006-12-12 04:13 +0000 [r48401]  Joshua Colp <jcolp@digium.com>

        * apps/app_voicemail.c: Use S_OR in my previous app_voicemail. This
          is the way it should have been done.

2006-12-11 23:02 +0000 [r48396-48399]  Matt O'Gorman <mogorman@digium.com>

        * sounds/Makefile: new sounds package with 100% more silence

        * /, apps/app_externalivr.c: Merged revisions 48394 via svnmerge
          from https://svn.digium.com/svn/asterisk/branches/1.2 ........
          r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006)
          | 4 lines app_externalivr needs a real silence file, and
          additional changes to add silence files into core instead of
          extra patch provided by bug 8177 with minor additions. ........

2006-12-11 21:31 +0000 [r48391]  Joshua Colp <jcolp@digium.com>

        * apps/app_voicemail.c: Return non-existant callerid handling to
          that which it was before. In 1.4 and trunk callerid can be
          allocated but not have any contents so we have to use
          ast_strlen_zero before passing it to the relevant functions.
          (issue #8567 reported by pabelanger)

2006-12-11 05:37 +0000 [r48382]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * funcs/func_strings.c: STRFTIME() does not actually require an
          argument (issue 8540)

2006-12-11 05:36 +0000 [r48377-48381]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: Merge in my latest RTP changes. Break out RTP and
          RTCP callback functions so they no longer share a common one.

        * apps/app_meetme.c: Use the correct API call to say a device state
          changed. (Yes, I'm a nub.)

        * apps/app_meetme.c: Don't access the conference structure after it
          has been freed.

2006-12-11 00:47 +0000 [r48375]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_nbscat.c, /, apps/app_festival.c, apps/app_mp3.c,
          res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c,
          apps/app_ices.c, res/res_musiconhold.c: Merged revisions 48374
          via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006)
          | 5 lines When doing a fork() and exec(), two problems existed
          (Issue 8086): 1) Ignored signals stayed ignored after the exec().
          2) Signals could possibly fire between the fork() and exec(),
          causing Asterisk signal handlers within the child to execute,
          which caused nasty race conditions. ........

2006-12-10 03:04 +0000 [r48372]  Steve Murphy <murf@digium.com>

        * channels/chan_zap.c, /: Merged revisions 48371 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1
          line This version applies the patch suggested by stevens in bug
          7836 (make inbound channel RINGING state consistent with other
          channels). ........

2006-12-09 15:59 +0000 [r48362-48363]  Russell Bryant <russell@digium.com>

        * channels/chan_iax2.c: Use locking when accessing the
          registrations list. This list is not actually used very often, so
          the likelihood of there being a problem is pretty small, but
          still possible. For example, if the CLI command to list the
          registrations was called at the same time that a reload was
          occurring and the registrations list was getting destroyed and
          rebuilt, a crash could occur. In passing, go ahead and convert
          this list to use the linked list macros.

2006-12-07 18:17 +0000 [r48357]  Russell Bryant <russell@digium.com>

        * /, res/res_musiconhold.c: Merged revisions 48356 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07
          Dec 2006) | 3 lines Ensure that the file position is not
          incremented beyond the total number of files available for
          playback. (issue #8539, ulogic) ........

2006-12-07 15:33 +0000 [r48349]  Steve Murphy <murf@digium.com>

        * main/manager.c, UPGRADE.txt, CHANGES: Here lies the fixes that
          killed bug 8423 -- OriginateSuccess and OriginateError incomplete
          channel name. May it rest in peace.

2006-12-06 16:25 +0000 [r48326]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Issue #8258 - fix handling of 487 being
          retransmitted to Asterisk

2006-12-06 16:15 +0000 [r48323]  Russell Bryant <russell@digium.com>

        * configs/iax.conf.sample, /: Merged revisions 48322 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06
          Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option
          in the sample configuration file. (issue #8526, arkadia) ........

2006-12-06 12:27 +0000 [r48316-48317]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Don't send Contact on MESSAGE

2006-12-05 20:42 +0000 [r48279]  Jason Parker <jparker@digium.com>

        * configure.ac: Fix curl version number testing to be much more
          friendly to non-bash shells. Issue 8508, patch by me. This
          *SHOULD* be POSIX compliant now..

2006-12-05 17:29 +0000 [r48264-48270]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Merging the invitestate-1.4 branch after
          successful testing. Will check if I can solve this with less
          changes in 1.2.

        * configs/sip.conf.sample: Add missing s from another repository.
          (thanks jcmoore!)

        * configs/sip.conf.sample: Updating sip.conf.sample with
          information about T38 not working when chan_local or chan_agent
          is involved in the call. I don't know how big a fix that would be
          to solve, but this is the current state of affairs. (Chan_sip
          currently checks if the other side of the bridge has a SIP tech.
          We could/should implement another check, possibly for udptl_write
          or some flag in the ast_channel structure).

2006-12-05 01:41 +0000 [r48252-48254]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_voicemail.c: Oops, forgot to release the odbc handle

        * apps/app_voicemail.c, /: Merged revisions 48251 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006)
          | 6 lines If the recording in the database is too large, it will
          fail to retrieve with an mmap error. Not too sure why this
          doesn't happen when we put it in the database, also, but since
          that doesn't seem to be broken, I'm not going to fix it (at least
          until someone reports it). Solution is to ask for the file in
          smaller chunks. (Bug 8385) ........

2006-12-04 21:48 +0000 [r48237-48248]  Jason Parker <jparker@digium.com>

        * apps/app_voicemail.c: Fix an issue which didn't allow
          unavail/greet/busy/etc messages from being saved into ODBC (and
          probably IMAP).

2006-12-04 17:54 +0000 [r48228-48230]  Jason Parker <jparker@digium.com>

        * configs/voicemail.conf.sample: Add documentation to
          voicemail.conf.sample for ODBC storage. Issue 8499 - patch by
          blitzrage.

        * doc/snmp.txt: Attempt to document some of the dependencies that
          are needed for net-snmp Issue 8499 - initial patch by blitzrage.

2006-12-03 06:34 +0000 [r48223]  Russell Bryant <russell@digium.com>

        * sounds/Makefile: When "fetch" is in use, instead of "wget",
          --continue is not a valid option. (issue #8451)

2006-12-02 21:45 +0000 [r48199-48219]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: - Removing one of two pieces of code to
          handle 481 response on INVITE - Move handling of REFER response
          to handle_response_refer()

        * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h,
          configs/sip.conf.sample: - Disable RTP hold timers while T.38 fax
          transmission happens - Encapsulate RTP timers in the rtp
          structure so we have one for video and one for audio The video
          one is not used in 1.4, really. Will be used for RTP keepalives
          when we can send something that video phones support in the RTP
          stream. I now this is a big architectual change at this stage for
          1.4, but decided it was needed to avoid future bug reports. -
          Document the RTP NAT keepalive option in sip.conf.sample Issue
          7679 in the bug tracker. Please test.

2006-12-02 03:50 +0000 [r48195]  Russell Bryant <russell@digium.com>

        * include/asterisk/utils.h: Backport the comment containing the
          warning regarding the limitations on the usage of this function.
          It is thread safe, but not technically reentrant.

2006-12-01 23:37 +0000 [r48193]  Kevin P. Fleming <kpfleming@digium.com>

        * apps/app_dial.c, /: Merged revisions 48192 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006)
          | 2 lines if Dial() is going to send music-on-hold to the calling
          party, it has to send PROGRESS first to ensure that the reverse
          audio path has been setup first (BE-106) ........

2006-12-01 23:16 +0000 [r48190]  Russell Bryant <russell@digium.com>

        * Makefile, configure, configure.ac, makeopts.in, sounds/Makefile:
          FreeBSD 6.1 does not include wget by default. However, it has
          fetch which will work just fine for our purposes of downloading
          the sounds packages. So, check for both wget and fetch and the
          configure script and use what was found to download them. If
          neither one was found, and sound packages are selected that must
          be downloaded, the install process will print out an informative
          error message indicating the situation. Also, fix a couple places
          where "make" was hard coded into some output messages by
          replacing them with the $(MAKE) variable. (issue #8451, initial
          patch by pabelanger, with additional modifications by me)

2006-12-01 20:25 +0000 [r48184-48186]  Jason Parker <jparker@digium.com>

        * configs/extensions.conf.sample, /: Merged revisions 48183 via
          svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2
          lines Fix a small typo - issue 8848, reported by pabelanger
          ........

2006-12-01 19:38 +0000 [r48179]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * main/cli.c: Double-unlock error (reported by blitzrage on IRC)

2006-12-01 17:41 +0000 [r48177]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c, configs/sip.conf.sample: - Backport of the
          "limitonpeers" patch from trunk, to fix a lot of issues with
          queues and SIP device states - Remove support for T.38 early
          media, since it's impossible. (Two patches in one - extra friday
          evening offer due to being off line from svn today... :-)

2006-11-30 21:18 +0000 [r48168]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c, include/asterisk/rtp.h, channels/chan_gtalk.c: Do not
          do a partial bridge for Google Talk since we need to handle STUN.
          (issue #8448 reported by phsultan)

2006-11-30 20:51 +0000 [r48166]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Issue 8319 - change noncecount before
          using it.

2006-11-30 20:28 +0000 [r48143-48162]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_iax2.c: Merged revisions 48157 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2
          lines Only print out debug message if bridged channel is not
          NULL. (issue #8412 reported by jubilex) ........

        * /, res/res_features.c: Merged revisions 48154 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2
          lines Do not listen for DTMF on the bridge that comes into
          existence when ParkedCall is executed. This means native bridging
          can now occur for this. (issue #8406 reported by kebl0155)
          ........

        * main/cdr.c, /: Merged revisions 48151 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2
          lines Print certain CDR messages out at the NOTICE level versus
          WARNING since they can occur when used with the CDR applications
          and are perfectly fine. (issue #8367 reported by dartvader)
          ........

        * /, configs/sip.conf.sample: Merged revisions 48142 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov
          2006) | 2 lines Document 'port' for SIP peers, came up because of
          the current mailing list thread. (issue #8450 reported by
          blitzrage) ........

2006-11-30 14:29 +0000 [r48129-48135]  Olle Johansson <oej@edvina.net>

        * doc/manager.txt: Explain status reports and make codefreeze more
          happy :-)

        * /, channels/chan_sip.c: Clean up bad dialogs properly. Caused by
          GS 487 adapter without CSEQ on separate line in the REGISTER
          request. Imported from 1.2.

2006-11-29 21:05 +0000 [r48115]  Joshua Colp <jcolp@digium.com>

        * apps/app_voicemail.c: Use MAILTMPLEN instead of sizeof in
          mm_login. (issue #8420 reported by slimey)

2006-11-29 19:56 +0000 [r48113]  Olle Johansson <oej@edvina.net>

        * configs/sip.conf.sample: Explain the use device status system
          implemented in SIP for subscriptions, queues and manager a bit
          better. Like in 1.2, you will get more detailed information if
          you set a call limit for a device. When the call limit is
          reached, the status system will report a device as busy. For
          queues, setting a call limit per SIP device is propably a
          requirement. In most cases, it will work much better if you only
          use type=peer and not type=friend. We might decide to backport
          the new setting from trunk to apply all call limits to the peer
          part of a friend only.

2006-11-29 16:50 +0000 [r48107]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c, /: Merged revisions 48106 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2
          lines If the frame was duplicated before writing out then we need
          to free it. (issue #8429 reported by edguy3) ........

2006-11-29 08:03 +0000 [r48105]  Olle Johansson <oej@edvina.net>

        * configs/sip.conf.sample: Clarify RTP timers. Sorry, grandma.

2006-11-29 04:26 +0000 [r48101]  Joshua Colp <jcolp@digium.com>

        * apps/app_voicemail.c: Don't crash if the mailstream was not
          created.

2006-11-28 18:26 +0000 [r48095]  Jason Parker <jparker@digium.com>

        * Makefile: Export several more variables in top level Makefile.
          Inspired by issue 8438.

2006-11-28 16:57 +0000 [r48054-48088]  Joshua Colp <jcolp@digium.com>

        * channels/chan_phone.c, /: Merged revisions 48087 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov
          2006) | 2 lines According to the research I have done we never
          needed to include compiler.h in the first place so let's not!
          (issue #8430 reported by edguy3) ........

        * apps/app_voicemail.c, /: Merged revisions 48053 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2
          lines Use the proper function to get the new message count
          instead of always using the filesystem. (issue #8421 reported by
          slimey) ........

2006-11-27 17:20 +0000 [r48049]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * /, res/res_musiconhold.c: Merged revisions 48045 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27
          Nov 2006) | 2 lines Random MOH wasn't really random (bug 8381)
          ........

2006-11-27 17:17 +0000 [r48046]  Russell Bryant <russell@digium.com>

        * main/manager.c: Remove a couple of unused variables (issue #8380,
          casper)

2006-11-27 15:32 +0000 [r48038]  Joshua Colp <jcolp@digium.com>

        * pbx/pbx_spool.c, /: Merged revisions 48037 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2
          lines Do not reference the freed outgoing structure in the debug
          message. (issue #8425 reported by arkadia) ........

2006-11-27 06:41 +0000 [r48031]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Change logging message

2006-11-26 00:26 +0000 [r48015-48017]  Steve Murphy <murf@digium.com>

        * funcs/func_cdr.c: might as well also document the raw values of
          the flag vars

        * /, funcs/func_cdr.c: A little bit of func_cdr documentation
          upgrade-- no bug# involved, although 8221 may have inspired it.

2006-11-25 09:28 +0000 [r48002]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Not having a HINT is not an ERROR. In 1.4
          and future releases, you can disable subscription support totally
          or per peer in sip.conf with allowsubscribe = yes | no

2006-11-24 17:17 +0000 [r47992]  Steve Murphy <murf@digium.com>

        * main/translate.c: bug 8189 posted this fix for main/translate.c
          for PLC

2006-11-24 15:46 +0000 [r47989]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/misdn_config.c,
          channels/chan_misdn.c, /: Merged revisions 47968 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23
          Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE.
          beatufied some logs, changed some loglevels. changed the default
          value of block_on_alarm ........

2006-11-23 11:01 +0000 [r47959]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Don't allocate unused variable.

2006-11-22 21:47 +0000 [r47944]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: Video will never reach Packet2Packet bridging and can
          do more harm then good.

2006-11-21 17:32 +0000 [r47897]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: If we have the non standard G726-32 setting turned on
          we want to return G726-32 to the SDP, not our AAL2 string. (issue
          #8330 reported by voipgate)

2006-11-21 15:20 +0000 [r47892]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Apparently Exosip sends a 101 after a 100
          provisional response. Let's not treat that as early media.
          (discovered at the AVTF meeting in Paris).

2006-11-20 20:01 +0000 [r47863-47864]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_voicemail.c: Oops, merge missed release of odbc object

        * apps/app_voicemail.c, /: Merged revisions 47862 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r47862 | tilghman | 2006-11-20 13:59:07 -0600 (Mon, 20 Nov 2006)
          | 2 lines Failing to trap -1 error from mmap causes segfault
          (Issue 8385) ........

2006-11-20 19:51 +0000 [r47850-47860]  Joshua Colp <jcolp@digium.com>

        * main/frame.c, /: Merged revisions 47859 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2
          lines Don't forget to byte swap if we are exiting the smoother
          feed early. (issue #8287 reported by arturs) ........

2006-11-16 23:00 +0000 [r47777]  Kevin P. Fleming <kpfleming@digium.com>

        * /, doc/billing.txt: update documentation regarding IAX2 transfers
          and CDRs Merged revisions 47776 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006)
          | 2 lines update clearly wrong documentation regarding cdr_custom
          ........

2006-11-16 21:11 +0000 [r47762-47764]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Compare technology using the pointers
          instead of a straight comparison based on name. (issue #8228
          reported by dean bath)

2006-11-16 20:09 +0000 [r47758]  Kevin P. Fleming <kpfleming@digium.com>

        * configure, configure.ac: check for pre-1.4 versions of Zaptel and
          abort the configure script if found with an appropriate error
          message

2006-11-16 19:24 +0000 [r47755]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c, configs/sip.conf.sample: Make the HOLD
          notification optional, in order to avoid a lot of extra database
          lookups for all those realtime users out there.

2006-11-16 18:29 +0000 [r47748-47751]  Joshua Colp <jcolp@digium.com>

        * channels/chan_local.c, /: Merged revisions 47750 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov
          2006) | 2 lines Because of the way chan_local is written we
          should be extra careful and make sure our callback functions have
          a tech_pvt. (issue #8275 reported by mflorell) ........

        * apps/app_meetme.c: Don't unreference the SLA object if there is
          no SLA object in the devicestate callback. (issue #8354 reported
          by loloski)

2006-11-16 16:51 +0000 [r47733-47744]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Don't fixup if there's nothing to fixup

        * UPGRADE.txt: Warn users about change in canreinvite

        * channels/chan_sip.c, configs/sip.conf.sample: - CANCEL is never
          authenticated (according to the RFC) - Update docs on
          canreinvite. "nonat" is the recommended setting for most users
          with phones behind a NAT.

2006-11-15 22:31 +0000 [r47712]  Joshua Colp <jcolp@digium.com>

        * channels/chan_local.c, /: Merged revisions 47711 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov
          2006) | 2 lines Make sure that the pvt structure exists before
          trying to do fixup on Local channels. (issue #7937 reported by
          mada123, fix by alamantia with mods by me) ........

2006-11-15 21:56 +0000 [r47709]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_voicemail.c: Fix ODBC_STORAGE for when context is NULL

2006-11-15 21:33 +0000 [r47707]  Joshua Colp <jcolp@digium.com>

        * main/channel.c: We need to ensure timelimit stuff is included as
          well so warnings get played. (issue #8050 reported by KNK)

2006-11-15 20:50 +0000 [r47701]  Kevin P. Fleming <kpfleming@digium.com>

        * main/file.c: don't try to call fclose() if fopen() failed

2006-11-15 20:31 +0000 [r47698]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: - Improve SIP history - Never send reply to
          ACK (again...)

2006-11-15 20:31 +0000 [r47684-47697]  Kevin P. Fleming <kpfleming@digium.com>

        * apps/app_voicemail.c, /: Merged revisions 47677 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15 Nov 2006)
          | 4 lines ensure that message duration is included in email
          notifications for forwarded messages (BE-96, fix by me after
          corydon used his clue-bat on me) ensure that duration in the
          message metadata is updated if prepending is done during
          forwarding (related to BE-96) remove prototype for API call that
          does not exist ........

        * main/config.c, /: Merged revisions 47686,47688-47689 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r47686 | kpfleming | 2006-11-15 13:42:05 -0600 (Wed, 15
          Nov 2006) | 2 lines clear the category's variable tail pointer as
          well when variables are detached from it ........ r47688 |
          kpfleming | 2006-11-15 13:47:43 -0600 (Wed, 15 Nov 2006) | 2
          lines when appending a list of variable to a category, ensure the
          tail pointer points to the last variable in the list ........
          r47689 | kpfleming | 2006-11-15 13:58:46 -0600 (Wed, 15 Nov 2006)
          | 2 lines when re-writing the config file, don't repeat the path
          if it hasn't changed ........

        * main/config.c, /: Merged revisions 47682 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r47682 | kpfleming | 2006-11-15 12:39:47 -0600 (Wed, 15 Nov 2006)
          | 2 lines ouch... don't use printf, use ast_log/ast_verbose
          ........

2006-11-15 17:46 +0000 [r47672]  Luigi Rizzo <rizzo@icir.org>

        * main/cli.c: fix longest match search in find_cli. Trunk already
          fixed. 1.2 not affected (well, i have no idea, the code is
          totally different there).

2006-11-15 15:25 +0000 [r47649-47656]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Send error message when we can't allocate
          SIP dialog, possibly due to limitation of file descriptors.
          (imported from 1.2)

2006-11-15 04:45 +0000 [r47645]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: If NAT detection is turned on or already detected
          then say NAT is active when setting the remote RTP peer when
          doing early bridging. (issue #8365 reported by marcelbarbulescu)

2006-11-15 00:19 +0000 [r47641]  Kevin P. Fleming <kpfleming@digium.com>

        * main/term.c: more formatting cleanup, and avoid running off the
          end of the string

2006-11-15 00:14 +0000 [r47639]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: Turn notice about unknown RTCP packet type into a
          debug message instead.

2006-11-15 00:05 +0000 [r47635]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/misdn/isdn_lib.c: silence compiler warning on 64-bit
          platforms (this variable is an 'int' anyway, comparing it to
          'signed long' is not useful)

2006-11-14 22:17 +0000 [r47625-47632]  Joshua Colp <jcolp@digium.com>

        * apps/app_voicemail.c, /: Merged revisions 47631 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r47631 | file | 2006-11-14 17:15:10 -0500 (Tue, 14 Nov 2006) | 2
          lines Update copyright information in the ADSI logo blob.
          ........

        * channels/chan_sip.c: Only keep the video RTP structure around if
          1. Video support is enabled and 2. A video codec is enabled on
          the dialog

        * funcs/func_uri.c: Small documentation clarification for
          URIENCODE. (issue #8294 reported by salaud)

2006-11-14 18:54 +0000 [r47621]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_voicemail.c: Conversion of res_odbc API to include ast_
          prefix did not completely transition app_voicemail when
          ODBC_STORAGE is used (reported on IRC by caio1982, not in
          bugtracker)

2006-11-14 16:45 +0000 [r47617]  Joshua Colp <jcolp@digium.com>

        * apps/app_amd.c: Use LOG_DEBUG to print out the indication that
          app_amd is using default settings instead of using LOG_NOTICE.
          This stops needless logging of this information under normal
          circumstances. (issue #8361 reported by Seb7)

2006-11-14 16:22 +0000 [r47597-47613]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Update documentation to fit the
          implementation...

        * /, channels/chan_sip.c: Issue #8272 - Don't destroy dialog in
          retransmission system if it's an OPTION packet from peerpoke

2006-11-13 21:28 +0000 [r47584]  Joshua Colp <jcolp@digium.com>

        * /, cdr/cdr_pgsql.c: Merged revisions 47583 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r47583 | file | 2006-11-13 16:26:36 -0500 (Mon, 13 Nov 2006) | 2
          lines Initialize global pointers for connection and result to
          NULL. (issue #8356 reported by james) ........

2006-11-13 20:20 +0000 [r47581]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * /, channels/chan_sip.c: Merged revisions 47580 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 Nov 2006)
          | 2 lines Having more than 255 old messages caused corruption in
          the new/old count ........

2006-11-13 19:15 +0000 [r47576]  Steve Murphy <murf@digium.com>

        * main/config.c: This solves bug 8342, whereby a crash occurs under
          certain circumstances while reading a config file with comments--
          a call to CB_ADD shouldn't happen if withcomments is zero

2006-11-13 19:11 +0000 [r47573]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * main/cli.c, channels/chan_sip.c: Re-enable old deprecated
          commands

2006-11-13 19:10 +0000 [r47572]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: - Don't reply to INVITE already replied
          to when we get BYE - Declare errmsg as int. Oops.

2006-11-13 18:18 +0000 [r47564]  Steve Murphy <murf@digium.com>

        * pbx/ael/ael-test/ref.ael-test3: Eager people beat me to fixing
          the messed if, but we all forgot to update the regressions. Until
          now.

2006-11-13 17:13 +0000 [r47553]  Steve Murphy <murf@digium.com>

        * pbx/pbx_ael.c: AEL need not complain about parkedcalls not being
          found... just confuses users

2006-11-13 17:08 +0000 [r47542-47551]  Joshua Colp <jcolp@digium.com>

        * /, apps/app_sms.c: Merged revisions 47549 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r47549 | file | 2006-11-13 12:05:32 -0500 (Mon, 13 Nov 2006) | 2
          lines When sending an SMS with a user data header properly set
          the UDH flag in the first byte. (issue #8347 reported by
          hoffmeis) ........

        * main/cli.c: Free full command string upon unregistering of CLI
          command. Backported from revision 47536 from rizzo.

2006-11-13 16:00 +0000 [r47540]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Only produce error message about sip history
          once

2006-11-13 05:48 +0000 [r47527]  Russell Bryant <russell@digium.com>

        * configure, acinclude.m4: AC_PROG_SED is included in autoconf
          2.60, but apparently it is not included in 2.59. So, to maintain
          compatability with 2.59 since it is a small change, copy this
          macro into acinclude.m4 and rename it to AST_PROG_SED. (issue
          #8345)

2006-11-13 05:46 +0000 [r47523-47526]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * res/res_odbc.c, /: Merged revisions 47525 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r47525 | tilghman | 2006-11-12 23:45:11 -0600 (Sun, 12 Nov 2006)
          | 2 lines If the execute fails a second time, make sure that we
          don't pass back a stale handle ........

        * channels/chan_zap.c, /: Merged revisions 47522 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12 Nov 2006)
          | 2 lines Don't play dialtone if the seizing the channel fails
          (Bug 7754) ........

2006-11-12 16:12 +0000 [r47507-47513]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Issue 8314 - Restore auto-framing (Thanks
          DEA!!!)

        * channels/chan_sip.c: Part of issue 8078 - parse even if udptl is
          UDPTL in sdp...

        * channels/chan_sip.c: - Don't destroy SIP dialog because of a
          failed T.38 re-invite. Wait for a bye. Final response to a
          re-invite does not mean that the session dies, only that the
          re-invite fails. - Keep RTP active during processing of T.38
          re-invite. If the re-invite fails, RTP needs to remain as before
          the re-invite. Issue 8338 - darren1713. Please test.

        * channels/chan_sip.c: -Remove blocking of ptime: parsing in sdp
          -Add some comments to t.38 code

2006-11-12 06:23 +0000 [r47492-47497]  Russell Bryant <russell@digium.com>

        * /, channels/chan_iax2.c: Merged revisions 47496 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12 Nov 2006) |
          4 lines Only do the check to determine whether the channel
          calling this function is an IAX2 channel when getting the IP
          address using the special argument, CURRENTCHANNEL. (issue #8341,
          jcovert) ........

        * Makefile: Add the target "menuconfig" as an alias for the
          "menuselect" target. This is just a favor to users so that if you
          accidentally type "make menuconfig" instead of "make menuselect",
          it still works. (inspired by a comment on IRC from wangster
          calling me an "especially devious asterisk developer" for having
          it be menuselect instead of menuconfig. :) )

        * main/term.c: Tweak the formatting of this new function to better
          conform to coding guidelines.

2006-11-11 02:04 +0000 [r47490]  Matt O'Gorman <mogorman@digium.com>

        * main/term.c, /, main/logger.c, include/asterisk/term.h: woohoo
          safe output!

2006-11-10 22:23 +0000 [r47480]  Matt Frederickson <creslin@digium.com>

        * channels/chan_zap.c: Make sure we don't use 32 bits when we only
          need one bit.

2006-11-10 21:42 +0000 [r47463-47476]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: ...and make sure that the dialog is
          destroyed, even if we don't get any answer on the bye... This is
          the channel that remains dead after the SIP transfer

        * channels/chan_sip.c: Add debug output while trying to trace bug
          in bug report

        * channels/chan_sip.c: Make sure we destroy dialog...

        * /, channels/chan_sip.c: Small cleanup of handle_request_invite()
          - imported from 1.2 with changes

2006-11-10 19:47 +0000 [r47462]  Matt Frederickson <creslin@digium.com>

        * channels/chan_zap.c: Fix for #7321. Be able to explicitly hide
          callerid name for switches that bork on it.

2006-11-10 18:56 +0000 [r47454]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Issue 8010 - Fix support for multipart
          SDP (alphaque)

2006-11-10 17:13 +0000 [r47444]  Luigi Rizzo <rizzo@icir.org>

        * build_tools/prep_moduledeps: grep -m is not available on BSD, so
          use head -1 instead

2006-11-10 16:53 +0000 [r47437]  Joshua Colp <jcolp@digium.com>

        * apps/app_chanspy.c: Only split up extension and context if a
          value exists. (issue #8332 reported by loloski)

2006-11-10 16:51 +0000 [r47436]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * channels/chan_mgcp.c, main/cli.c, channels/chan_sip.c,
          channels/chan_skinny.c, channels/chan_h323.c,
          channels/chan_iax2.c: Discussion of these CLI changes resulted in
          more consistency (Bug 8236)

2006-11-10 16:36 +0000 [r47432-47433]  Kevin P. Fleming <kpfleming@digium.com>

        * apps/app_queue.c: if adding a queue member is LOG_NOTICE, then
          removing them should be LOG_NOTICE, not LOG_DEBUG

        * apps/app_queue.c: reflect addition/removal of dynamic queue
          members in queue_log, so that people using dialplan replacement
          for AgentCallbackLogin can still track login/logout (issue #7736,
          reported/patched by whoiswes but this commit was written by me
          and covers all three paths for AQM/RQM)

2006-11-10 13:04 +0000 [r47414-47418]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Rip out half implementation of 491 response
          support, since it wasn't implemented properly and caused memory
          leaks in the case of us getting 491's, which Asterisk actually
          sends... Since it is a bit too complicated to fix this, I'll rip
          it out of 1.4 and put it on the to-do-list for future releases.
          Now, we handle this as congestion, which it really is. Issue
          #8331

        * channels/chan_sip.c: Fix bit definition for SIP_PAG2_CALL_ONHOLD.
          Thanks fenlander!

2006-11-10 03:44 +0000 [r47398-47405]  Joshua Colp <jcolp@digium.com>

        * channels/chan_h323.c: Fix building of chan_h323 by completeing
          some structure definitions. (issue #8327 reported by Mithraen)

        * apps/app_voicemail.c: Do conversion in a more easier to read and
          working way for \r, \n, and \t. (issue #8324 reported by
          johnlange)

2006-11-09 21:26 +0000 [r47391]  Russell Bryant <russell@digium.com>

        * apps/app_voicemail.c, channels/chan_zap.c,
          build_tools/prep_moduledeps: Work around an issue that caused
          menuselect to display a bogus description for app_voicemail and
          chan_zap. These modules use some preprocessor directives to
          determine what it will report to Asterisk as its description.
          However, the way we extract this information from the source
          files for menuselect is not smart enough to figure this out.
          (issue #8326, #8328)

2006-11-09 16:53 +0000 [r47380]  Joshua Colp <jcolp@digium.com>

        * channels/chan_phone.c, /: Merged revisions 47379 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov
          2006) | 2 lines Don't include compiler.h on kernels 2.6.18 and
          higher as, well, it's apparently going to be removed. This should
          make all you FC6 fans happy as your Asterisk will now build
          without any mods. ........

2006-11-09 16:28 +0000 [r47352-47377]  Russell Bryant <russell@digium.com>

        * main/cli.c: fix tab completion for "core debug channel" and "core
          no debug channel"

        * main/cli.c: Fix "core show channel". Also, fix tab completion for
          both "core show channel" and "core show channels".

        * main/cli.c: Fix "core debug channel <whatever>". I guess someone
          needs to go through and audit every CLI command that changed
          number of arguments ...

        * main/asterisk.c: revert the previous change, which actually
          modified the deprecated command, "show profile". Now, actually
          apply the change to "core show profile".

        * main/asterisk.c: Fix argument parsing for the "core show profile"
          CLI command (fixed by rizzo in his branch, team/rizzo/astobj2)

        * main/cli.c: Fix another CLI command, "core show uptime" ...
          (issue #8323, reported by johnlange, fixed by myself)

        * main/asterisk.c: fix "core show version" to reflect the new
          number of arguments for this CLI command (issue #8316, kshumard)

2006-11-08 23:14 +0000 [r47344-47348]  Steve Murphy <murf@digium.com>

        * main/channel.c: This update fixes 7531

        * channels/chan_skinny.c: Committed in behalf of 8190.

2006-11-08 21:46 +0000 [r47333-47338]  Kevin P. Fleming <kpfleming@digium.com>

        * main/frame.c: the battle over CLI command formats has broken
          stuff...

        * channels/chan_sip.c: add simple fix for SDP to report proper
          sample rate for G.722 media sessions

2006-11-08 17:03 +0000 [r47323-47331]  Russell Bryant <russell@digium.com>

        * utils/streamplayer.c: I occasionally get email from users that
          are trying to figure out what this does, or due to some
          misunderstanding as to what it is supposed to do, can't get it to
          work. So, I have added some text here to hopefully explain what
          this application does and does not do.

        * channels/chan_gtalk.c: Make this module build again

        * configure, configure.ac, acinclude.m4: Copy the macros from
          libtool.m4 to our own acinclude.m4 such that libtool is no longer
          required to be installed to be able to generated the configure
          script.

2006-11-08 07:43 +0000 [r47309-47310]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo)

2006-11-07 23:46 +0000 [r47303]  Steve Murphy <murf@digium.com>

        * channels/chan_oss.c, main/channel.c, channels/chan_phone.c,
          channels/chan_misdn.c, channels/chan_skinny.c,
          channels/chan_features.c, channels/chan_h323.c,
          channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c,
          include/asterisk/stringfields.h, apps/app_voicemail.c,
          main/pbx.c, channels/chan_vpb.cc, channels/chan_local.c,
          channels/chan_zap.c, channels/chan_sip.c, res/res_features.c,
          channels/chan_agent.c, main/utils.c, include/asterisk/channel.h,
          channels/chan_gtalk.c, channels/chan_iax2.c: These mods are to
          solve the problem in bug 7506. It's a lot of rework to solve a
          fairly small problem... such is life.

2006-11-07 20:14 +0000 [r47284-47287]  Joshua Colp <jcolp@digium.com>

        * channels/chan_local.c: Make MOH work as it did before in
          chan_local, without this then it can go funky when transfers and
          MOH are involved. (issue #7671 reported by jmls)

2006-11-07 18:56 +0000 [r47279]  Kevin P. Fleming <kpfleming@digium.com>

        * configs/musiconhold.conf.sample: clean up sample config, and make
          native file playback the more obvious default choice

2006-11-07 18:38 +0000 [r47275]  Matt O'Gorman <mogorman@digium.com>

        * apps/app_voicemail.c: large overhaul to voicemail imap support.
          Allows support for more imap servers, also a better
          implementation of several parts of the original work. patch
          provided by 8033 with major upgrades.

2006-11-07 17:30 +0000 [r47268]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Issue 8303 (lrizzo) - break instead of
          continue.

2006-11-07 13:13 +0000 [r47250]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Fixing the attack shield so it doesn't
          produce attacks... Issue 8265 - never reply to an ACK

2006-11-07 01:25 +0000 [r47239]  Russell Bryant <russell@digium.com>

        * /, res/res_musiconhold.c: Merged revisions 47238 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06
          Nov 2006) | 5 lines If random order is enabled for files mode
          music on hold, set a random initial position, instead of always
          starting at the first file, and doing the random operation only
          when switching to the next file. (bug reported by John Lange on
          the asterisk-dev mailing list) ........

2006-11-04 18:32 +0000 [r47199]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Issue #8284: Fixes to Invite/replaces and
          transfer from "john" Thank you!

2006-11-04 18:10 +0000 [r47192-47196]  Russell Bryant <russell@digium.com>

        * main/cli.c: Fix another bug in "core set debug" ...

        * main/asterisk.c, main/cli.c: Really fix the "core set debug" and
          "core set verbose" CLI commands.

        * main/cli.c: fix the "atleast" option to the "core set verbose"
          and "core set debug" CLI commands

2006-11-03 23:17 +0000 [r47176]  Steve Murphy <murf@digium.com>

        * channels/chan_sip.c: This fix introduced via bug 8233

2006-11-03 17:53 +0000 [r47107-47108]  Luigi Rizzo <rizzo@icir.org>

        * bootstrap.sh: align bootstrap.sh with the version in trunk (needs
          to be blocked as it is already in trunk)

        * configure.ac: add proper environment vars to detect modules on
          freebsd. (already applied to trunk so it needs to be blocked
          there)

2006-11-02 23:49 +0000 [r47051-47053]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * main/rtp.c, main/udptl.c, channels/chan_skinny.c, res/res_agi.c,
          channels/chan_h323.c, apps/app_queue.c, res/res_jabber.c: More
          changes making the CLI more consistent with "category verb
          arguments" (continuation of issue 8236)

        * main/config.c, main/cli.c, main/channel.c, main/manager.c,
          channels/chan_skinny.c, channels/chan_features.c, res/res_agi.c,
          main/http.c, main/file.c, main/logger.c, main/image.c,
          res/res_indications.c, main/asterisk.c, res/res_odbc.c,
          channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c,
          channels/chan_local.c, main/frame.c, channels/chan_sip.c,
          res/res_features.c, channels/chan_agent.c, res/res_crypto.c,
          res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c:
          Reverse change of "show" to "list" and make several other
          commands more consistent with "category verb arguments"

2006-11-02 19:56 +0000 [r46992-47015]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Move check for codec translation to
          sip_call() instead of in add_sdp. No one bothers with the result
          of add_sdp anyway... Yet...

        * channels/chan_sip.c: Disable code for T38 over TCP and RTP since
          there's no trace of actual functionality for it :-)

2006-11-02 17:49 +0000 [r46965]  Russell Bryant <russell@digium.com>

        * /, res/res_musiconhold.c: Merged revisions 46964 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r46964 | russell | 2006-11-02 12:47:56 -0500 (Thu, 02
          Nov 2006) | 3 lines ignore files in a music on hold directory
          that begin with '.' (issue #8249, cboie) ........

2006-11-02 17:17 +0000 [r46963]  Nadi Sarrar <ns@beronet.com>

        * channels/misdn/isdn_lib.c: find_free_chan_in_stack usage fix

2006-11-02 16:45 +0000 [r46937]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_sip.c: don't send INVITE when we have determined
          that we can't offer any audio formats due to lack of transcoding
          support (or incorrect configuration)

2006-11-02 16:06 +0000 [r46930]  Joshua Colp <jcolp@digium.com>

        * /, channels/chan_sip.c: Merged revisions 46920 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2
          lines Repeat after me oej: I will at least make sure my code
          compiles before I commit it. ........

2006-11-02 15:24 +0000 [r46901]  Olle Johansson <oej@edvina.net>

        * /, channels/chan_sip.c: Dont overwrite pkt->flags (from 1.2)

2006-11-02 14:02 +0000 [r46845-46883]  Russell Bryant <russell@digium.com>

        * /, main/callerid.c: Add the missing call to free described in
          issue #8268. Also, add a bunch of missing calls to free in
          callerid_feed_jp().

        * main/say.c: fix saying one hundred and two hundred in hebrew
          (issue #7810, eldadran)

        * Makefile, configure, codecs/gsm/Makefile, configure.ac,
          build_tools/strip_nonapi, makeopts.in: Fixes for
          cross-compilation on mips (issue #8058, ywalther, with some
          modifications)

        * aclocal.m4, build_tools/menuselect-deps.in, configure,
          build_tools/embed_modules.xml, configure.ac: Add a check in the
          configure script to determine whether ld is GNU ld or not. This
          is needed because module embedding only works for gnu ld. GNU ld
          is now listed as a dependency for all of the module embedding
          options in menuselect. (issue #8143)

2006-11-01 20:35 +0000 [r46822]  Matt O'Gorman <mogorman@digium.com>

        * channels/chan_gtalk.c: bind address support from bug 8164

2006-11-01 19:49 +0000 [r46802]  Steve Murphy <murf@digium.com>

        * res/res_config_odbc.c: a fix for bug 8251; the var_val needs to
          accept longer strings or mass confusion and a lot of lost time is
          the result

2006-11-01 18:39 +0000 [r46780]  Joshua Colp <jcolp@digium.com>

        * main/Makefile: Force poll() emulation for Darwin to always be on.
          It's too broken to consider being used. This resolves the console
          issue OSX users have been seeing. I would have liked to autoconf
          this but I haven't been able to come up with a test case that
          works. Que sera.

2006-11-01 18:26 +0000 [r46778]  Russell Bryant <russell@digium.com>

        * res/res_monitor.c, /: Merged revisions 46776 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 Nov 2006) |
          9 lines soxmix and Asterisk expect different file extensions for
          certain formats. This was already handled for the wav49 format.
          However, it was not handled for ulaw and alaw. I fixed this in
          such a way that using the alternate extensions for ulaw and alaw
          will only happen if we know we're calling soxmix, and not a
          custom script defined using the MONITOR_EXEC variable. The wav49
          processing was left alone so that external scripts will see no
          behavior change. (issue #7550, reported by mnicholson, proposed
          patch by junky, committed fix is a bit different) ........

2006-11-01 18:21 +0000 [r46775]  Joshua Colp <jcolp@digium.com>

        * channels/chan_iax2.c: It's another round of chan_iax2 fixes!
          Should hopefully fix the deadlock issues people have been
          reporting. IAXtel now has qualify turned on for 800 peers and it
          is handling it fine.

2006-11-01 17:48 +0000 [r46760]  Steve Murphy <murf@digium.com>

        * main/config.c: Cleanups suggested by Russell.

2006-11-01 16:39 +0000 [r46744]  Russell Bryant <russell@digium.com>

        * channels/chan_zap.c: Prevent an infinite loop when config
          processing gets to a jitterbuffer option

2006-10-31 22:02 +0000 [r46716]  Jason Parker <jparker@digium.com>

        * main/translate.c: Fix "core show translation" output. Issue
          #8243, patch by Damin.

2006-10-31 21:47 +0000 [r46711-46714]  Kevin P. Fleming <kpfleming@digium.com>

        * include/asterisk/translate.h, main/translate.c: add an API so
          that translators can activate/deactivate themselves when needed

        * include/asterisk/translate.h, main/translate.c: revert changes
          that were the wrong way to address this... proper fix coming

        * main/translate.c: let's set the seen flag early enough to
          actually make a difference...

        * include/asterisk/translate.h, main/translate.c: don't re-do setup
          operations for translators that can dynamically register
          themselves

2006-10-31 10:56 +0000 [r46583-46631]  Olle Johansson <oej@edvina.net>

        * main/enum.c, funcs/func_enum.c, include/asterisk/enum.h: Issue
          #8089 - Fix the ENUM support (picking one record by number).
          Thanks otmar!

        * /, channels/chan_sip.c, configs/sip.conf.sample: Support ;rport
          when we're supposed to support ;rport. Issue #7473.

        * /, channels/chan_sip.c: If peer fails ACL check, fail peer at
          REGISTER

        * channels/chan_sip.c: Fix T38 too. Thanks, tgrman !

2006-10-31 06:30 +0000 [r46554-46563]  Russell Bryant <russell@digium.com>

        * contrib/init.d/rc.redhat.asterisk: Start Asterisk later in the
          boot process to ensure it starts after stuff like MySQL (issue
          #8253, Alric)

        * /, main/utils.c: Merged revisions 46560 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r46560 | russell | 2006-10-31 01:18:36 -0500 (Tue, 31 Oct 2006) |
          3 lines When handling the case where the hostname is just an IPV4
          numeric address, be sure to set the address type. (issue #8247,
          alexr) ........

        * /, res/res_agi.c: Merged revisions 46557 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r46557 | russell | 2006-10-31 01:13:09 -0500 (Tue, 31 Oct 2006) |
          3 lines fix some copy/paste bugs in the checking of arguments for
          the "control stream file" AGI command (issue #8255, mnicholson)
          ........

        * main/translate.c: Add a small tweak to the code that checks to
          see whether destination formats are translatable based on the
          source format. If we have already determined that there is no
          translation path in one direction, don't bother checking the
          other direction.

2006-10-30 22:19 +0000 [r46511-46526]  Kevin P. Fleming <kpfleming@digium.com>

        * main/translate.c: when unregistering a translator, don't rebuild
          the translation matrix unless needed when filtering formats out
          of an offer, ensure we check for translation ability in both
          directions

        * include/asterisk/linkedlists.h: ensure that items removed from a
          list are always unlinked from the list (next pointer set to NULL)

2006-10-30 21:09 +0000 [r46474-46506]  Joshua Colp <jcolp@digium.com>

        * configure, configure.ac: Don't explicitly link in crypt as it is
          not used on some platforms.

        * channels/chan_iax2.c: We need to lock the pvt structure during
          retransmission as another worker thread may be doing something as
          well.

2006-10-30 16:27 +0000 [r46382-46433]  Olle Johansson <oej@edvina.net>

        * main/asterisk.c, apps/app_voicemail.c, include/asterisk/file.h,
          include/asterisk/doxyref.h, channels/chan_sip.c,
          main/ast_expr2f.c, include/asterisk/module.h,
          formats/format_ogg_vorbis.c, main/app.c,
          include/asterisk/channel.h, include/asterisk/lock.h,
          include/asterisk/frame.h: Issue #8246 - Doxygen fixes from
          kshumard. An extra big thankyou is given to everyone that
          contributes to doxygen! THANK YOU!

        * main/rtp.c, /: Bind RTCP to the same IP as RTP

        * /, channels/chan_sip.c: Issue #7869 - Stop retransmission of 302
          redirects (imported from 1.2)

        * /, channels/chan_sip.c: Issue #7608 - Notifications sent with
          wrong content-type (imported from 1.2, modified)

        * channels/chan_sip.c, CHANGES: Backport of patch for #7828 that
          was reported for trunk, but obviously exists in 1.4 too.

        * channels/chan_sip.c: Restoring the old logic, since working
          around it and fixing it seemed too complicated. - The
          SIP_OUTGOING flag indicates the direction of the last transaction
          in the dialog. - The initreq stores the last request in the
          dialog, the request that opened the latest transaction. Please
          now retry all the 1.4 bug reports with mixed to/from headers,
          tags etc in ACK, BYE, CANCEL. Thanks!

        * channels/chan_sip.c: Accepting a message twice may be
          misinterpreted...

        * channels/chan_sip.c: - 183 is not reliable message... - Error
          should not have SDP

2006-10-28 16:37 +0000 [r46377]  Joshua Colp <jcolp@digium.com>

        * utils/Makefile: Don't build muted on OpenBSD, it is not
          supported.

2006-10-27 19:03 +0000 [r46370]  Russell Bryant <russell@digium.com>

        * channels/chan_zap.c: move the copy of the default settings to the
          global settings back out of process_zap, so that they aren't
          overwritten when process_zap is called multiple times

2006-10-27 18:29 +0000 [r46367]  Olle Johansson <oej@edvina.net>

        * contrib/asterisk-ng-doxygen: Put some doxygen pressure on
          Christian :-)

2006-10-27 17:39 +0000 [r46358-46363]  Russell Bryant <russell@digium.com>

        * main/asterisk.c, res/res_agi.c, apps/app_externalivr.c,
          res/res_musiconhold.c: We should always be using _exit() after a
          fork() or vfork() instead of exit(). This is because exit() does
          some extra cleanup which in some implementations of vfork(), for
          example, can actually modify the state of the parent process,
          causing very weird bugs or crashes. (issue #7971, Nick Gavrikov)

        * channels/chan_zap.c: Instead of iterating all of the options once
          to look for jitterbuffer options, and then again for everything
          else, move the processing of jitterbuffer options into the main
          loop so that there are no erroneous messages about ignoring
          unknown options. (issue #8226)

2006-10-27 10:03 +0000 [r46351-46353]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
          channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
          Merged revisions 46350 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) |
          1 line fixed a bug which caused chan_misdn to try to allocate 2
          times the same channel on high load, which then caused
          instability of mISDN. removed a useless function from isdn_lib.c
          ........

        * channels/misdn_config.c: fixed not compile issue, which was just
          introduced

        * channels/misdn_config.c, channels/chan_misdn.c, /,
          channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
          Merged revisions 46176 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) |
          1 line added nttimeout option to configure wether we disconnect
          calls on NT timeouts or not during an overlapdial session
          ........

2006-10-26 17:57 +0000 [r46335-46340]  Jason Parker <jparker@digium.com>

        * /, contrib/scripts/astgenkey.8: Merged revisions 46337 via
          svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r46337 | qwell | 2006-10-26 12:47:52 -0500 (Thu, 26 Oct 2006) | 2
          lines oops - somebody forgot to change this - long ago, probably.
          ........

        * CHANGES: grammar check

2006-10-26 16:38 +0000 [r46331]  Olle Johansson <oej@edvina.net>

        * CHANGES: Corrections to changes (Multiparking is not included)

2006-10-26 16:31 +0000 [r46329]  Russell Bryant <russell@digium.com>

        * main/translate.c: - If the source has no audio or no video
          portion, do not call powerof() to get the format index. - Don't
          run through the audio and video loops if there is no audio or
          video portion of the source If 0 is passed to powerof, it will
          return -1. This value of -1 was then being used as an array index
          in these loops, which caused a crash on some systems. Other than
          this issue, this code works as we expected it to. If a format is
          not in the source, and we have to translation path to it, it is
          not offered in the list of acceptable destination formats. (fixes
          issue #8231)

2006-10-26 12:15 +0000 [r46317]  Kevin P. Fleming <kpfleming@digium.com>

        * CHANGES: update to reflect G.722 addition

2006-10-26 04:18 +0000 [r46298]  Russell Bryant <russell@digium.com>

        * doc/backtrace.txt: update backtrace documentation to reflect
          changes in 1.4 (issue #8230, kshumard)

2006-10-26 01:37 +0000 [r46287]  Mark Spencer <markster@digium.com>

        * main/config.c, main/manager.c: Fix config comment code
          preservation code (thanks murf!)

2006-10-25 20:14 +0000 [r46276]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Old todo note - Don't add Contact header on
          BYE and Cancel

2006-10-25 19:24 +0000 [r46253-46255]  Russell Bryant <russell@digium.com>

        * configure.ac: fix error output when checking for openh323 to
          refer to openh323 instead of pwlib (issue #8222, misaksen)

2006-10-25 19:16 +0000 [r46252]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Somewhat ugly code to try to fix issue
          #7608. Since the problem was not very well defined, the fix is a
          bit fuzzy too... Thanks to Luigi for accidentally spotting the
          possible problem!

2006-10-25 19:08 +0000 [r46249]  Russell Bryant <russell@digium.com>

        * apps/app_queue.c: update warning message to include "agi" option
          (issue #8225, jmls)

2006-10-25 18:13 +0000 [r46237-46248]  Kevin P. Fleming <kpfleming@digium.com>

        * sounds/Makefile: use 1.4.3 extra sounds with corrected silence
          files

        * sounds/sounds.xml, sounds/Makefile: add support for prebuilt
          G.722 prompts and music on hold files

2006-10-25 15:56 +0000 [r46214-46216]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: show settings doesn't produce a list of
          similar objects, it should stay a "show"

2006-10-25 14:32 +0000 [r46200]  Kevin P. Fleming <kpfleming@digium.com>

        * main/cli.c, main/cdr.c, channels/chan_phone.c, pbx/pbx_spool.c,
          channels/chan_features.c, pbx/pbx_ael.c, channels/chan_h323.c,
          pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_sms.c,
          main/image.c, channels/chan_nbs.c, apps/app_rpt.c, main/db.c,
          cdr/cdr_custom.c, channels/chan_mgcp.c,
          apps/app_parkandannounce.c, apps/app_voicemail.c,
          channels/chan_sip.c, apps/app_softhangup.c, apps/app_record.c,
          res/res_adsi.c, main/utils.c, apps/app_ices.c,
          pbx/dundi-parser.c, channels/chan_iax2.c, apps/app_queue.c,
          apps/app_getcpeid.c: apparently developers are still not aware
          that they should be use ast_copy_string instead of strncpy... fix
          up many more users, and fix some bugs in the process

2006-10-25 04:58 +0000 [r46165]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * main/pbx.c: WaitExten truncates decimals of times to wait,
          instead of accepting them (Bug 8208)

2006-10-25 00:26 +0000 [r46152-46154]  Kevin P. Fleming <kpfleming@digium.com>

        * main/rtp.c, main/frame.c, main/translate.c, formats/format_pcm.c,
          channels/chan_h323.c, channels/chan_iax2.c,
          include/asterisk/frame.h: add passthrough and file format support
          for G.722 16KHz audio (issue #5084, original patch by andrew,
          updated by mithraen)

        * channels/chan_sip.c, main/translate.c: code zone experiment:
          don't offer formats in the outbound INVITE that aren't either
          passthrough or translatable

        * main/translate.c: if multiple translators are registered for the
          same source/dest combination, ensure that the lowest-cost one is
          always inserted earlier in the list

2006-10-24 20:30 +0000 [r46142]  Mark Spencer <markster@digium.com>

        * res/res_agi.c: Fix FastAGI when there is no pid (bug #7628,
          #8147)

2006-10-24 19:29 +0000 [r46130]  Joshua Colp <jcolp@digium.com>

        * channels/chan_iax2.c: We need to initialize our scheduler pthread
          condition... yes.

2006-10-24 08:34 +0000 [r46114-46117]  Luigi Rizzo <rizzo@icir.org>

        * main/http.c: merge 45152 don't leak descriptors in http.c

        * channels/chan_sip.c: merge 45966 refer_to_domain potentially
          containing options

        * channels/chan_sip.c: merge 46026 improper checks on get_header()
          return values

        * channels/chan_sip.c: merge 46045 prevent NULL args to
          ast_strdupa() in chan_sip.c

2006-10-24 05:23 +0000 [r46093]  Russell Bryant <russell@digium.com>

        * Makefile: Restore the ability to remove the firmware directory
          without causing the installation to fail (issue #8111)

2006-10-24 03:53 +0000 [r46080-46083]  Kevin P. Fleming <kpfleming@digium.com>

        * main/translate.c: ensure that the translation matrix is properly
          lock-protected every place it is used

        * include/asterisk/translate.h, main/translate.c: add an API call
          to allow channel drivers to determine which media formats are
          compatible (passthrough or transcode) with the format an existing
          channel is already using

        * doc/imapstorage.txt: simplify and correct voicemail IMAP storage
          build instructions

2006-10-24 03:01 +0000 [r46078]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * main/channel.c: Pass through a frame if we don't know what it is,
          rather than trying to pass a NULL, which will segfault a channel
          driver (Bug 8149)

2006-10-24 01:27 +0000 [r45999-46067]  Russell Bryant <russell@digium.com>

        * utils/muted.c, utils/ael_main.c: In muted.c, check the return
          value of strdup. In ael_main.c, check the return value of calloc.
          (issue #8157) In passing fix a few minor bugs in ael_main.c. The
          last argument to strncpy() was a hard-coded 100, where it should
          have been 99. I changed this to use sizeof() - 1.

        * apps/app_meetme.c: Fix the descriptions of some of the
          MeetMeAdmin options (issue #8098, mflorell)

        * res/res_jabber.c: don't crash when an incoming message has no
          "from" (issue #8205, jmls)

2006-10-23 00:27 +0000 [r45928]  Joshua Colp <jcolp@digium.com>

        * /, cdr/cdr_odbc.c: Merged revisions 45927 via svnmerge from
          https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
          r45927 | file | 2006-10-22 20:25:28 -0400 (Sun, 22 Oct 2006) | 2
          lines Don't leak memory mmmk? ........

2006-10-22 21:44 +0000 [r45916]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c, /: Merged revisions 45808 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r45808 | crichter | 2006-10-21 14:35:13 +0200 (Sat, 21
          Oct 2006) | 1 line fixed issue, that if chan_misdn is loaded and
          couldn't be initialized it would cause a segfault after 'reload'.
          Reported by Drew/Matt thx. ........

2006-10-21 18:49 +0000 [r45818]  Russell Bryant <russell@digium.com>

        * res/res_monitor.c: Add a couple missing unregistrations of
          manager actions and remove duplicate unregistrations of
          applications. (issue #8194, jmls)

2006-10-21 18:48 +0000 [r45775-45817]  Joshua Colp <jcolp@digium.com>

        * main/loader.c: Don't use promotion on Darwin because it doesn't
          seem to work quite right in all cases, this should solve the
          unresolved symbol issue people have been seeing.

        * Makefile: Pass DESTDIR and ASTSBINDIR so that the utilities get
          installed in the proper location (reported on asterisk-dev
          mailing list)

2006-10-20 07:44 +0000 [r45741]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Let's understand SIP: - REFER can create
          dialog, Asterisk does not support it yet - NOTIFY can create
          dialog in Asterisk's implementation (voicemail) even though we
          don't support the server side of it. In this case, the standard
          is a side issue ;-) - Added extened functionality for unsupported
          methods (PING, PUBLISH) so we don't create PVT's for those
          either. Russellb needs to judge what to do with this in 1.2, but
          I think the current implementation n 1.2 is a bug since we're
          sending bad replies to NOTIFY and REFER outside of dialogs

2006-10-19 17:24 +0000 [r45678-45694]  Joshua Colp <jcolp@digium.com>

        * res/res_jabber.c: Let's remember to unregister JabberStatus too
          (issue #8184 reported by jmls)

        * /, apps/app_externalivr.c: Merged revisions 45691 via svnmerge
          from https://origsvn.digium.com/svn/asterisk/branches/1.2
          ........ r45691 | file | 2006-10-19 13:16:37 -0400 (Thu, 19 Oct
          2006) | 2 lines Respect language selection when seeing if the
          file exists (issue #8178 reported by mnicholson) ........

        * channels/chan_sip.c: If the jitterbuffer is forced on then we
          can't partially bridge (reported by wangster on #asterisk-dev)

2006-10-19 00:59 +0000 [r45622]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: Don't leak the actual thread-specific
          sip_pvt struct

2006-10-18 23:49 +0000 [r45621]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_sip.c: don't leak memory when a chan_sip thread is
          destroyed that has a thread-local temp_pvt allocated

2006-10-18 21:03 +0000 [r45595]  Joshua Colp <jcolp@digium.com>

        * main/asterisk.c: Don't modify things if we are using vfork as
          this is very bad and may cause unexpected behavior (issue #7970
          reported by Nick Gavrikov)

2006-10-18 11:54 +0000 [r45517]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: remove duplicate declarations

2006-10-18 04:09 +0000 [r45464]  Luigi Rizzo <rizzo@icir.org>

        * main/http.c: merge from trunk: move ast_variables_destroy() to a
          better place in handle_uri() to avoid leaking memory on non
          existing files.

2006-10-18 03:02 +0000 [r45452]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: Don't segfault if you're using a channel driver that
          doesn't turn RTCP on

2006-10-18 02:41 +0000 [r45439-45441]  Russell Bryant <russell@digium.com>

        * main/channel.c: Don't attempt to access private data members of
          the pthread_mutex_t object, because this does not work on all
          linux systems. Instead, just access the reentrancy field in the
          ast_mutex_info struct when DEBUG_THREADS is enabled. If
          DEBUG_CHANNEL_LOCKS is enabled, the developer probably has
          DEBUG_THREADS on as well. (issue #8139, me)

        * configs/sip_notify.conf.sample: update entry to reboot a snom
          phone (issue #7850, pnlarsson)

2006-10-17  Kevin P. Fleming  <kpfleming@digium.com>

        * Asterisk 1.4.0-beta3 released.

2006-10-17 22:31 +0000 [r45408-45410]  Kevin P. Fleming <kpfleming@digium.com>

        * include/asterisk/stringfields.h, main/ast_expr2.c,
          main/channel.c, channels/chan_sip.c, channels/chan_iax2.c:
          optimize the 'quick response' code a bit more... no more malloc()
          or memset() for each response expand stringfields API a bit to
          allow reusing the stringfield pool on a structure when needed,
          and remove some unnecessary code when the structure was being
          freed

2006-10-17 20:38 +0000 [r45378-45381]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Don't create a "real" pvt structure for
          requests that shouldn't be able to create one. Instead use a
          temporary pvt and fill it with enough information so we can send
          a reply.

2006-10-17 17:39 +0000 [r45329]  Olle Johansson <oej@edvina.net>

        * configs/sip.conf.sample: Adding information about Marks
          direct-RTP hack to the docs...

2006-10-17 17:22 +0000 [r45327]  Kevin P. Fleming <kpfleming@digium.com>

        * LICENSE: provide licensing language for IAXy firmware file

2006-10-16 20:06 +0000 [r45246-45280]  Joshua Colp <jcolp@digium.com>

        * apps/app_dial.c, apps/app_directed_pickup.c: Backport of new
          directed pickup (BE-85).

2006-10-16 13:59 +0000 [r45196-45213]  Olle Johansson <oej@edvina.net>

        * CREDITS: Adding Inotel to credits for SIP transfers. Thanks for
          your support!

        * channels/chan_sip.c: Don't destroy dialog for unexpected REFER
          response...

2006-10-14 04:38 +0000 [r45143]  Steve Murphy <murf@digium.com>

        * funcs/func_rand.c: update the doc string for both AEL and
          extensions.conf users.

2006-10-13 23:02 +0000 [r45125]  Kevin P. Fleming <kpfleming@digium.com>

        * main/acl.c don't drop the entire permit/deny list when an attempt
          is made to add an invalid entry (BE-92)

2006-10-13 21:06 +0000 [r45104-45106]  Joshua Colp <jcolp@digium.com>

        * res/res_speech.c: Clear the quiet flag too since we are
          restarting a recognition again (reported on -dev by Stephan
          Edelman)

        * res/res_speech.c: Check return value from engine in case of
          failure (ie: out of licenses) (reported on -dev mailing list)

2006-10-13 20:52 +0000 [r45103]  Steve Murphy <murf@digium.com>

        * pbx/ael/ael-test/ref.ael-vtest17 (added),
          pbx/ael/ael-test/ael-vtest17/extensions.ael (added),
          pbx/ael/ael-test/ael-vtest17 (added),
          pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c: Bug 8128 fixed in
          this release via these changes

2006-10-13 19:19 +0000 [r45088]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c: avoiding warning, fixing potential bug

2006-10-13 18:42 +0000 [r45051-45079]  Joshua Colp <jcolp@digium.com>

        * codecs/lpc10/placev.c, codecs/lpc10/irc2pc.c,
          codecs/lpc10/decode.c, codecs/lpc10/dcbias.c,
          codecs/lpc10/pitsyn.c, codecs/lpc10/voicin.c,
          codecs/lpc10/difmag.c, codecs/lpc10/hp100.c,
          codecs/lpc10/synths.c, codecs/lpc10/preemp.c,
          codecs/lpc10/rcchk.c, codecs/lpc10/lpfilt.c,
          codecs/lpc10/mload.c, codecs/lpc10/lpcenc.c,
          codecs/lpc10/vparms.c, codecs/lpc10/dyptrk.c,
          codecs/lpc10/lpcini.c, codecs/lpc10/random.c,
          codecs/lpc10/ham84.c, codecs/lpc10/chanwr.c,
          codecs/lpc10/placea.c, codecs/lpc10/tbdm.c,
          codecs/lpc10/analys.c, codecs/lpc10/onset.c,
          codecs/lpc10/energy.c, codecs/lpc10/deemp.c,
          codecs/lpc10/lpcdec.c, codecs/lpc10/ivfilt.c,
          codecs/lpc10/median.c, codecs/lpc10/encode.c,
          codecs/lpc10/bsynz.c, codecs/lpc10/prepro.c,
          codecs/lpc10/invert.c: And file said... let the compiler warnings
          STOP!

        * apps/app_chanspy.c: Turn on volume adjustment if it needs to be on (issue #8136
          reported by mnicholson)

        * apps/app_playback.c: Move say.conf existence check to do_say
          function since it is called from multiple places (issue #8144
          reported by kshumard)

2006-10-13 16:19 +0000 [r45049]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_iax2.c: when sending a call to a peer, use the proper socket if
          we have multiple bindings (reported on asterisk-dev)

2006-10-13 16:01 +0000 [r45031-45040]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Complete merging in RPID screen changes
          (issue #8101 reported by hristo, patch by oej in revision 44757)

        * main/dnsmgr.c: Pass the right value to usleep for sleeping, and always add
          the background refresh item back into the scheduler if enabled
          since it is deleted during reload. (issue #8142 reported by
          p_lindheimer)

2006-10-13 15:41 +0000 [r45027]  Kevin P. Fleming <kpfleming@digium.com>

        * configure, include/asterisk/autoconfig.h.in, configure.ac,
          main/utils.c: use a configure script test for PMTU discovery
          control instead of just assuming it's available on Linux

2006-10-13 14:45 +0000 [r44994-45026]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed some
          echocandisable issues when bridged. this caused a kernel panic
          sometimes.. also some minor formatting fixes

        * channels/misdn/isdn_msg_parser.c: fixed issue that the hangupcause
          got a wrong isdn cause at RELEASE_COMPLETE

2006-10-12 22:07 +0000 [r44992]  Luigi Rizzo <rizzo@icir.org>

        * channels/chan_sip.c: merge formatting and minor code
          simplifications from trunk

2006-10-12 20:34 +0000 [r44982]  Matt O'Gorman <mogorman@digium.com>

        * channels/chan_gtalk.c: fix for bug 7764.

2006-10-12 19:14 +0000 [r44956-44971]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_sip.c: we can only send one 'a=ptime' attribute per
          media session, not one for each format

        * main/netsock.c, include/asterisk/utils.h, channels/chan_sip.c,
          main/utils.c: ensure that IAX2 and SIP sockets allow UDP
          fragmentation when running on Linux (thanks to Brian Candler on
          the asterisk-dev list for the tip)

2006-10-12 16:56 +0000 [r44945]  Russell Bryant <russell@digium.com>

        * main/manager.c: fix a silly typo in a comment that I saw while
          reading the commit list

2006-10-12 16:08 +0000 [r44942]  Joshua Colp <jcolp@digium.com>

        * Makefile: Pass off AUDIO_LIBS so muted can link on OSX (issue
          #8135 reported by ssokol)

2006-10-12 12:55 +0000 [r44921]  Nadi Sarrar <ns@beronet.com>

        * main/manager.c: append_event must be called while holding the
          session lock

2006-10-12 10:24 +0000 [r44911]  Russell Bryant <russell@digium.com>

        * res/res_jabber.c: change some debug output to use LOG_DEBUG
          instead of verbose output

2006-10-11 16:57 +0000 [r44888]  Jason Parker <jparker@digium.com>

        * main/db1-ast/Makefile: These are already set by the parent
          Makefile.. There is no need to have this here (it doesn't
          actually work anyways).

2006-10-11 09:18 +0000 [r44854]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c: removed warning because of missing
          prototype declaration

2006-10-10 19:23 +0000 [r44830]  Olle Johansson <oej@edvina.net>

        * channels/chan_sip.c: Do not set default/global values in the
          variable declaration, set it in reload_config()

2006-10-10 17:21 +0000 [r44819]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Move some stuff around so that a NOTIFY
          dialog won't hang around until the end of the world under certain
          circumstances

2006-10-10 16:44 +0000 [r44809]  Paul Cadach <paul@odt.east.telecom.kz>

        * main/channel.c, funcs/func_channel.c, include/asterisk/channel.h:
          CHANNEL() function sometime mix parameter and value

2006-10-10 16:42 +0000 [r44808]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * funcs/func_logic.c: Lost of a bit of logic when this was
          simplified between 1.2 and 1.4 (Bug 8117)

2006-10-10 16:30 +0000 [r44806]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Bail out if we have no refer structure and
          we get a refer response

2006-10-10 16:21 +0000 [r44805]  Luigi Rizzo <rizzo@icir.org>

        * channels/chan_sip.c: more merge from trunk (comments and change a
          static function name)

2006-10-10 15:23 +0000 [r44788]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Only set DTMF information if an RTP
          structure exists

2006-10-10 13:50 +0000 [r44786]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/chan_misdn.c: (re)added
          support of dynamically enabling hdlc on bchannels

2006-10-10 08:25 +0000 [r44776-44777]  Luigi Rizzo <rizzo@icir.org>

        * channels/chan_sip.c: whitespace changes related to previous
          commit

        * channels/chan_sip.c: merge a few code simplifications that have
          gone into trunk during last week, to reduce differences between
          the two branches and make porting fixes easier.

2006-10-09 16:12 +0000 [r44764]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Fix a problem where phones that go
          "missing" never got unregistered. Issue #8067, reported by pj,
          patch by Anthony LaMantia (with minor whitespace modifications)

2006-10-09 15:46 +0000 [r44759-44760]  Joshua Colp <jcolp@digium.com>

        * channels/chan_iax2.c: iaxs[callno] may go away if we try to avoid
          the deadlock

        * channels/chan_iax2.c: Properly avoid a collision with iax2_hangup
          (issue #8115 reported by vazir)

2006-10-08 14:14 +0000 [r44746]  Luigi Rizzo <rizzo@icir.org>

        * channels/chan_sip.c: do not dereference p if we
          know it is NULL

2006-10-07 14:39 +0000 [r44684]  Paul Cadach <paul@odt.east.telecom.kz>

        * channels/h323/ast_h323.cxx, channels/chan_h323.c,
          channels/h323/ast_h323.h, channels/h323/chan_h323.h: Propagate
          caller's transfer capability too

2006-10-07 11:37 +0000 [r44650-44665]  Luigi Rizzo <rizzo@icir.org>

        * channels/chan_sip.c: put common code in a
          function to avoid repetitions.

        * channels/chan_sip.c: remove hardwired usage of 5060, use
          DEFAULT_SIP_PORT instead

        * channels/chan_sip.c: option_debug checking
          before printing to debug channel.

        * channels/chan_sip.c: backport simplifications on sip_register,
          usage of ast_set2_flag(), and fixes to the handling of failed
          module loading.

        * channels/chan_sip.c: improve and document function
          get_in_brackets(), introducing a helper function
          find_closing_quote() of more general use.

2006-10-06 21:28 +0000 [r44629-44631]  Kevin P. Fleming <kpfleming@digium.com>

        * include/asterisk/linkedlists.h: ensure that mutex locks inside
          list heads are initialized properly on platforms that require
          constructor initialization (issue #8029, patch from timrobbins)

        * CHANGES: remove Jingle as per mog

2006-10-06 21:08 +0000 [r44628]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: Remove the seqno check for RFC2833, the handler is
          smart enough to not need it.

2006-10-06 21:07 +0000 [r44627]  Kevin P. Fleming <kpfleming@digium.com>

        * CHANGES: various cleanups

2006-10-06 18:46 +0000 [r44581-44605]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: When the sequence number rolls over then reset the
          recorded sequence number for DTMF (issue #8106 reported by
          bungalow)

        * main/file.c: Even more frames to treat as though the remote side
          disappeared (issue #8097 reported by eldadran)

2006-10-06 15:59 +0000 [r44567]  Luigi Rizzo <rizzo@icir.org>

        * main/manager.c, main/http.c: make sure sockets are blocking when
          they should be blocking.

2006-10-06 12:53 +0000 [r44559-44563]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c: fixed segfault which happens during
          hold/transfer action

        * channels/chan_misdn.c: if INFORMATION Message come with keypad
          instead of called party number, we just use the keypad as called
          party number.

        * channels/misdn/isdn_lib.c, channels/misdn_config.c,
          channels/misdn/isdn_lib.h, channels/chan_misdn.c,
          channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
          added the option 'reject_cause' to make it possible to set
          the RELEASE_COMPLETE - cause on the 3. incoming PMP channel,
          which is automatically rejected because chan_misdn does not
          support that kind of callwaiting. Therefore chan_misdn supports
          now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc
          now gets the info if the requested channel is incoming or
          outgoing to make the 3. channel possible

        * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
          channels/chan_misdn.c: fixed the hold/retrieve/transfer issues,
          removed a useless bc field, added setting of frame.delivery fields,
          some minor code cleanups

2006-10-05 19:57 +0000 [r44502]  Joshua Colp <jcolp@digium.com>

        * main/file.c: Treat busy control frames as hangup in the file streaming
          core (issue #8097 reported by eldadran)

2006-10-05 18:21 +0000 [r44488]  Steve Murphy <murf@digium.com>

        * pbx/pbx_ael.c: This mod fixes a problem pointed out by dgarstang.
          Many thanks to Doug!

2006-10-05 18:01 +0000 [r44486]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: One more T.38 fix! Don't leave a reinvite
          hanging by a thread if the other side is already setup with T.38

2006-10-05 16:10 +0000 [r44476]  Kevin P. Fleming <kpfleming@digium.com>

        * main/app.c: don't segfault when an argument without a close
          parenthesis is found stop parsing as soon as that situation
          occurs

2006-10-05 15:22 +0000 [r44465-44466]  Steve Murphy <murf@digium.com>

        * CHANGES: I put the accumulated changes from the commit logs and
          inspection, into CHANGES. Hope everyone approves!

        * configs/muted.conf.sample, utils/muted.c: Hang on a minute, the
          install process sticks muted.conf in /etc/asterisk, so that's
          where muted should look for it, right?

2006-10-05 02:40 +0000 [r44450]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Don't totally bail out if T.38 was
          negotiated

2006-10-05 01:42 +0000 [r44433-44436]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_sip.c: fix Polycom presence notification again

2006-10-04 22:52 +0000 [r44407-44409]  Luigi Rizzo <rizzo@icir.org>

        * utils/Makefile: as far as i can tell astman only uses newt...

        * Makefile: put linker flags in ASTLDFLAGS where they belong

2006-10-04 21:17 +0000 [r44390-44393]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_sip.c: remove workaround for old Polycom firmware SUBSCRIBE
          requests add workaround for new Polycom firmware SUBSCRIBE
          requests (bug is known to exist in 2.0.1 firmware)

        * include/asterisk.h, main/utils.c: make LOW_MEMORY builds actually
          work

2006-10-04 19:57 +0000 [r44380]  Steve Murphy <murf@digium.com>

        * pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael.tab.c,
          pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12,
          pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
          pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4,
          pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6,
          pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8,
          pbx/ael/ael-test/ael-test16/extensions.ael (added),
          pbx/ael/ael-test/ael-test16 (added), pbx/ael/ael.y,
          pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14,
          pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9,
          pbx/ael/ael-test/ref.ael-test16 (added): These changes fix the
          problems reported in bug 8090

2006-10-04 19:47 +0000 [r44378]  Kevin P. Fleming <kpfleming@digium.com>

        * channels/chan_oss.c, main/cdr.c, channels/chan_phone.c,
          main/manager.c, pbx/pbx_spool.c, res/res_smdi.c,
          channels/chan_skinny.c, channels/chan_h323.c, main/http.c,
          channels/chan_alsa.c, pbx/pbx_dundi.c, apps/app_mixmonitor.c,
          main/asterisk.c, channels/chan_mgcp.c, main/autoservice.c,
          include/asterisk/utils.h, main/dnsmgr.c, channels/chan_zap.c,
          channels/chan_sip.c, apps/app_meetme.c, res/res_snmp.c,
          main/devicestate.c, main/utils.c, res/res_musiconhold.c,
          channels/chan_iax2.c, apps/app_queue.c, res/res_jabber.c: update
          thread creation code a bit reduce standard thread stack size
          slightly to allow the pthreads library to allocate the stack+data
          and not overflow a power-of-2 allocation in the kernel and waste
          memory/address space add a new stack size for 'background'
          threads (those that don't handle PBX calls) when LOW_MEMORY is
          defined

2006-10-04 17:04 +0000 [r44337-44365]  Steve Murphy <murf@digium.com>

        * configs/muted.conf.sample: I've been meaning to add some
          explanation about muted... here it is

        * configs/manager.conf.sample: CLI reverbification update to this
          config file

        * apps/app_macro.c: In response to bug 7776, a Warning has been
          added to the doc string for Macro().

2006-10-04 00:25 +0000 [r44322]  Kevin P. Fleming <kpfleming@digium.com>

        * main/asterisk.c, main/loader.c, main/term.c, Makefile,
          include/asterisk.h: ensure that local include files are always
          used avoid a duplicate function name (term_init())

2006-10-03 22:35 +0000 [r44312]  Matt O'Gorman <mogorman@digium.com>

        * channels/chan_gtalk.c, res/res_jabber.c: fix issue with dialing
          client without resource.

2006-10-03 20:18 +0000 [r44298]  Kevin P. Fleming <kpfleming@digium.com>

        * apps/app_queue.c: fix a logic error in my previous fix to the queue
          reload code

2006-10-03 18:42 +0000 [r44286]  Paul Cadach <paul@odt.east.telecom.kz>

        * channels/h323/ast_h323.cxx: Change default presentation indicator
          to "user provided not screened" if octet 3a missed in
          CallingPartyNumber IE

2006-10-03 18:35 +0000 [r44284]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Use VideoSupport instead so it is considered
          a valid XML attribute name. (issue #8075 reported by renemendoza)

2006-10-03 18:30 +0000 [r44283]  Paul Cadach <paul@odt.east.telecom.kz>

        * channels/h323/ast_h323.cxx: Fix preparation of type and
          presentation of calling number

2006-10-03 00:01 +0000 [r44240]  Matt O'Gorman <mogorman@digium.com>

        * doc/jingle.txt, channels/chan_jingle.c (removed),
          include/asterisk/jabber.h, configs/jingle.conf.sample (removed),
          res/res_jabber.c: updated res_jabber for even better component
          support, soon will be jep-0100 compliant. also removed
          chan_jingle and infromed info from jingle.txt, chan_gtalk still
          works and should be used in this version.

2006-10-02 20:11 +0000 [r44199-44215]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Change the fd on the I/O context in case it
          changed during the reload, which is indeed possible. (issue #7943
          reported by eclubb)

        * contrib/init.d/rc.redhat.asterisk: We should be using $AST_SBIN
          instead of hardcoding the path for the error message (issue #7942
          reported by eclubb)

2006-10-02 18:52 +0000 [r44186]  Paul Cadach <paul@odt.east.telecom.kz>

        * configs/users.conf.sample, pbx/pbx_config.c: Missed part of
          userconf functionality for chan_h323

2006-10-02 17:25 +0000 [r44169]  Joshua Colp <jcolp@digium.com>

        * main/io.c: Shrink when current_ioc is unused. It is set to -1 when
          unused, not 0. (issue #7941 reported by eclubb)

2006-10-02 17:16 +0000 [r44166-44167]  Paul Cadach <paul@odt.east.telecom.kz>

        * doc/realtime.txt: Typo fix

        * channels/chan_h323.c: Optimization of oh323_indicate(): less
          locks - less problems, plus single exit point

2006-10-02 02:38 +0000 [r44146]  Mark Spencer <markster@digium.com>

        * channels/chan_sip.c, channels/chan_iax2.c: Don't use Channel when
          you're not talking about a channel :)

2006-10-01 19:32 +0000 [r44135]  Paul Cadach <paul@odt.east.telecom.kz>

        * channels/chan_h323.c: Do not simulate any audio tones if we got
          PROGRESS message

2006-10-01 18:30 +0000 [r44111-44125]  Russell Bryant <russell@digium.com>

        * Makefile: Fix a problem that cuased AST_DATA_DIR in defaults.h to
          be empty. The cause is that since ASTDATADIR is explicitly
          exported using "export ASTDATADIR" at the top of the Makefile,
          make no longer considers the variable "undefined", so the
          Makefile can't use ?= to set ASTDATADIR if not yet set. (issue
          #8063, reported by akohlsmith, fixed by me)

        * configs/queues.conf.sample: Fix the name of the "eventmemberstatus"
          option in the sample queues.conf (issue #8065, adamg)

2006-10-01 15:01 +0000 [r44109]  Luigi Rizzo <rizzo@icir.org>

        * channels/chan_sip.c: sync with trunk - move variable declarations
          to the beginning of a block.

2006-09-30 19:20 +0000 [r44090]  Paul Cadach <paul@odt.east.telecom.kz>

        * main/rtp.c: Allow one-way RTP streams (device->Asterisk)

2006-09-30 16:28 +0000 [r44080]  Luigi Rizzo <rizzo@icir.org>

        * codecs/lpc10/Makefile, Makefile, main/Makefile: fix two recent
          build problems: - with AST_DEVMODE, building codecs/lpc10 fails
          because of lots of warnings, and the configure step in editline
          fails as well. Fix this by removing the -Werror in these steps. -
          on FreeBSD (but probably on other platforms as well), the final
          link of asterisk fails because AST_LIBS was not exported to the
          subdirs Makefiles. Add a proper fix in the top-level Makefile (a
          possible alternative way is to add "export AST_LIBS" near the
          beginning of the file). With this fix, i believe that some of the
          platform-specific conditionals in main/Makefile are redundant
          (because they should be already dealt with in the top level
          Makefile) but i don't have a platform to check. Merging to head
          will happen in a moment.

2006-09-30 16:12 +0000 [r44068-44078]  Paul Cadach <paul@odt.east.telecom.kz>

        * channels/chan_sip.c: Fix issue #7928 correctly. Next is a comment
          of previous fix: Issue #7928 - Don't send both 404 and 503. Fix
          by phsultan with a small fix by me, myself or I. Thanks,
          Philippe! (This was caused by my changes to the transaction
          handling)

        * channels/chan_sip.c: Found some buggy SIP clients (phones Planet
          VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which
          sends ACK not on OK message only (when remote party answers) but
          on RINGING message too, so when we send 200 OK message, we get
          unidentified ACK message (because INVITE acknowledged on RINGING
          message already), so 200 OK retransmits within its retransmission
          interval then call gets dropped. If someone else knows how to
          provide workaround for such cases, please, fix it in correct way.
          Thanks to ssh from #asteriskru for provide access to his box to
          study and fix this case.

2006-09-29 22:51 +0000 [r44055-44057]  Kevin P. Fleming <kpfleming@digium.com>

        * agi, utils: ignore temporary files made by the Makefiles during a
          build

        * codecs/lpc10/Makefile, main/db1-ast/Makefile, agi/Makefile,
          codecs/Makefile, utils/Makefile, configure,
          build_tools/embed_modules.xml, codecs/gsm/Makefile, configure.ac,
          Makefile.moddir_rules, Makefile.rules, codecs/ilbc/Makefile,
          pbx/Makefile, res/Makefile, channels/Makefile: fix a few build
          system bugs, and convert Makefiles to be compatible with GNU make
          3.80

2006-09-29 22:35 +0000 [r44053]  Jason Parker <jparker@digium.com>

        * main/asterisk.c, main/cli.c: Fix a bug with the removal of
          'atleast' argument to 'core verbose' and 'core debug'. Add that
          argument back in.

2006-09-29 21:09 +0000 [r44022-44043]  Paul Cadach <paul@odt.east.telecom.kz>

        * channels/h323/ast_h323.cxx: Set TON/PRESENTATION information more
          carefully when no CallingNumber IE available

        * channels/h323/ast_h323.cxx: Fake display name by called number on
          incoming calls (until passing connected number/connected name is
          not implemented)

        * channels/h323/ast_h323.cxx: Ported code refers to H.450 - add
          includes

        * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Properly
          pass TON/PRESENTATION information - original
          H323Connection::SendSignalSetup() destroys Q.931 fields.

2006-09-29 18:49 +0000 [r44011-44012]  Kevin P. Fleming <kpfleming@digium.com>

        * main/Makefile: yet another place where we were not using the
          correct CFLAGS by default

        * main/Makefile: missed one conversion to ASTCFLAGS

2006-09-29 18:30 +0000 [r44009]  Paul Cadach <paul@odt.east.telecom.kz>

        * channels/h323/ast_h323.cxx, channels/chan_h323.c,
          channels/h323/ast_h323.h, channels/h323/chan_h323.h: Pass
          TON/PRESENTATION information too

2006-09-29 18:25 +0000 [r43952-44008]  Kevin P. Fleming <kpfleming@digium.com>

        * main/db1-ast/Makefile, Makefile, codecs/Makefile, utils/Makefile,
          main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules,
          Makefile.rules, pbx/Makefile, channels/Makefile: don't abuse
          CFLAGS and LDFLAGS for build of Asterisk components, because they
          are also then used for non-Asterisk components (like menuselect);
          use our own variables instead

        * configure, configure.ac: support --without-curl in configure
          script

        * Makefile.rules: another cross-compile fix

        * Makefile: a couple more environment settings that can't leak into
          the menuselect build

        * main/cli.c: proper fix for ast_group_t change

        * include/asterisk/lock.h: eliminate compiler warning when
          DEBUG_CHANNEL_LOCKS is enabled and users of this header file
          don't also include channel.h

2006-09-28 20:11 +0000 [r43944]  Jason Parker <jparker@digium.com>

        * apps/app_queue.c: Fix incorrect argument order for member names,
          on persisted members. Issue 8047, patch by jmls.

2006-09-28 18:05 +0000 [r43932-43933]  Joshua Colp <jcolp@digium.com>

        * apps/app_playback.c, res/res_monitor.c,
          include/asterisk/logger.h, channels/chan_misdn.c, res/res_smdi.c,
          channels/chan_skinny.c, apps/app_rpt.c, channels/chan_mgcp.c,
          main/udptl.c, main/frame.c, funcs/func_timeout.c,
          channels/chan_sip.c, apps/app_festival.c,
          channels/iax2-provision.c, apps/app_alarmreceiver.c,
          res/res_musiconhold.c, apps/app_followme.c, channels/chan_iax2.c:
          Put in missing \ns on the end of ast_logs (issue #7936 reported
          by wojtekka)

2006-09-28 17:35 +0000 [r43919]  Kevin P. Fleming <kpfleming@digium.com>

        * apps/app_queue.c: fix buggy (and overly complex) loop used during reload
          of app_queue for static member list updating

2006-09-28 17:34 +0000 [r43918]  Paul Cadach <paul@odt.east.telecom.kz>

        * channels/h323/ast_h323.cxx: Extend call establishment timeout

2006-09-28 17:31 +0000 [r43913-43915]  Joshua Colp <jcolp@digium.com>

        * channels/chan_iax2.c: Make sure the pvt exists before accessing
          it again as it may have gone away (issue #7562 reported by Seb7
          and issue #7939 reported by sorg)

        * main/cli.c: Warning be gone!

2006-09-28 16:41 +0000 [r43899]  BJ Weschke <bweschke@btwtech.com>

        * apps/app_queue.c: app_queue is comparing the device names incorrectly
          while checking their statuses. It's internal list of interfaces
          includes the dial string, while the argument passed to this
          function does not have the dial string (/n for a local channel).
          This causes it to ignore the device state changes because it
          thinks it belongs to none of its members. (#8040 reported and
          patch by tim_ringenbach)

2006-09-28 16:17 +0000 [r43893]  Joshua Colp <jcolp@digium.com>

        *  apps/app_meetme.c: Stop the stream after waitstream returns so that our
          formats get restored. (issue #7370 reported by kryptolus)

2006-09-28 15:56 +0000 [r43877]  Paul Cadach <paul@odt.east.telecom.kz>

        * channels/h323/ast_h323.cxx: Fix compiler warning

2006-09-28 15:29 +0000 [r43864-43873]  BJ Weschke <bweschke@btwtech.com>

        * apps/app_queue.c: Fix race conditioon crash with get_member_status (#7864 -
          tim_ringenbach reported and patched)

        * apps/app_queue.c: Autopause not working for queue members. (#8042
          - jmls reported and patch)

2006-09-28 12:58 +0000 [r43861-43862]  Paul Cadach <paul@odt.east.telecom.kz>

        * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Force
          remote side to start media on outgoing PROGRESS message

        * include/asterisk/compiler.h: Put attribute tag at correct place

2006-09-28 11:03 +0000 [r43852]  Christian Richter <christian.richter@beronet.com>

        * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
          channels/chan_misdn.c: fixed a bug which led to chan_list zombies,
          when the call could not be properly established in misdn_call.
          also removed the ACK_HDLC stuff which is not really needed.

2006-09-28 10:51 +0000 [r43843-43846]  Paul Cadach <paul@odt.east.telecom.kz>

        * channels/h323/ast_h323.cxx: Do not open transmit channel until
          TCS is received

        * main/file.c: Don't warn on HOLD/UNHOLD control frames

        * main/file.c: Don't treat unknown control frames as voice

2006-09-27 20:21 +0000 [r43816]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_voicemail.c: Avoid inability to lock directory log message by
          creating the directory ahead of time. (Issue 7631)

2006-09-27 19:44 +0000 [r43801-43803]  Jason Parker <jparker@digium.com>

        * apps/app_playback.c, main/pbx.c: Fix an issue with PLAYBACKSTATUS
          not being set under certain circumstances. Fix a minor issue, to
          make it use the filenames that were parsed, instead of the entire
          argument string. Fix Background() to return -1 like Playback(),
          if no args are specified.

2006-09-27 19:10 +0000 [r43783-43798]  Joshua Colp <jcolp@digium.com>

        * main/rtp.c: Compensate for out of order packets better if RFC2833
          compensation is turned on.

        * channels/chan_iax2.c: Get rid of two functions from a time now
          past (we THINK these are from pre-recursive lock time) that may
          be contributing to two open issues on the bug tracker (7562/7939)
          and that has the potential to just make bad things happen if the
          timing is right.

2006-09-27 16:55 +0000 [r43779]  Russell Bryant <russell@digium.com>

        * main/channel.c,res/res_features.c: Fix a problem that occurred if
          a user entered a digit
          that matched a bridge feature that was configured using multiple
          digits, and the digit that was pressed timed out in the feature
          digit timeout period. For example, if blind transfer is
          configured as '##', and a user presses just '#'. In this
          situation, the call would lock up and no longer pass any frames.
          (issue #7977 reported by festr, and issue #7982 reported by
          michaels and valuable input provided by mneuhauser and kuj. Fixed
          by me, with testing help and peer review from Joshua Colp). There
          are a couple of issues involved in this fix: 1) When
          ast_generic_bridge determines that there has been a timeout, it
          returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets
          this result, it calls ast_generic_bridge over again with the same
          timestamp for the next event. This results in an endless loop of
          nothing until the call is terminated. This is resolved by simply
          changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it
          sees a timeout. 2) I also changed ast_channel_bridge such that if
          in the process of calculating the time until the next event, it
          knows a timeout has already occured, to immediately return
          AST_BRIDGE_COMPLETE instead of attempting to bridge the channels
          anyway. 3) In the process of testing the previous two changes, I
          ran into a problem in res_features where ast_channel_bridge would
          return because it determined that there was a timeout. However,
          ast_bridge_call in res_features would then determine by its own
          calculation that there was still 1 ms before the timeout really
          occurs. It would then proceed, and since the bridge broke out and
          did *not* return a frame, it interpreted this as the call was
          over and hung up the channels. The reason for this was because
          ast_bridge_call in res_features and ast_channel_bridge in
          channel.c were using different times for their calculations.
          channel.c uses the start_time on the bridge config, which is the
          time that the feature digit was recieved. However, res_features
          had another time, 'start', which was set right before calling
          ast_channel_bridge. 'start' will always be slightly after
          start_time in the bridge config, and sometimes enough to round up
          to one ms. This is fixed by making ast_bridge_call use the same
          time as ast_channel_bridge for the timeout calculation. ........

2006-09-27 16:24 +0000 [r43775]  Christian Richter <christian.richter@beronet.com>

        * channels/chan_misdn.c, channels/Makefile: removed the chan_misdn
          versioning, since Asterisk has it's own

2006-09-27 16:23 +0000 [r43774]  Joshua Colp <jcolp@digium.com>

        * channels/chan_sip.c: Make rfc2833compensate a global option.

2006-09-27 04:35 +0000 [r43756]  Russell Bryant <russell@digium.com>

        * apps/app_voicemail.c: Backport revision 43754 from the trunk,
          which removes an unused buffer from mm_login to close bug 8038,
          as well as addresses some formatting and coding guidelines issues
          in passing. Originally, I did not commit this to 1.4 since it is
          not necessarily fixing a bug. However, since the IMAP storage
          code is brand new, I decided it would be better to make the
          change here as well, in case someone has to work on this code to
          address issues in the very near future. I don't want to make
          unnecessary merge problems going to the trunk.

2006-09-27 02:32 +0000 [r43739]  Steve Murphy <murf@digium.com>

        * configs/extensions.ael.sample: This change to extensions.ael was
          to fix bug 8031; the install scripts are causing it to be copied
          to /etc/asterisk/extensions.ael, and because it is a fairly
          direct conversion of the original extensions.conf, the macro and
          context names clash with the existing extensions.conf. So, I put
          an ael- in front of all macros and contexts, and checked every
          goto and macro call. Also, this file compiles under aelparse.

2006-09-26 20:56 +0000 [r43710]  Russell Bryant <russell@digium.com>

        * main/asterisk.c: Back in revision 4798, this message was changed from
          using ast_cli() to directly calling write(). During this change,
          checking if this was a remote console was removed. This caused
          this message about using "exit" or "quit" to exit an Asterisk
          console to come up in times where it did not make sense. This
          change restores the check to see if this is a remote console
          before printing the message. (fixes BE-65)

2006-09-26 20:47 +0000 [r43707]  Joshua Colp <jcolp@digium.com>

        * .cleancount, main/cli.c, channels/chan_sip.c,
          include/asterisk/channel.h: Use proper type to represent the group variable
          (issue #8025 reported by makoto)

2006-09-26 20:30 +0000 [r43700-43703]  Russell Bryant <russell@digium.com>

        * channels/chan_sip.c: Add missing newline character in the warning
          message about deprecated TOS values in configuration.

        * apps/app_voicemail.c: When parsing the sections of voicemail.conf that contain
          mailbox definitions, don't introduce a length limit on the
          definition by using a 256 byte temporary storage buffer. Instead,
          make the temporary buffer just as big as it needs to be to hold
          the entire mailbox definition. (fixes BE-68)

2006-09-26 20:19 +0000 [r43695-43697]  Joshua Colp <jcolp@digium.com>

        * channels/chan_local.c: Strip options off the argument passed for
          devicestate in chan_local. (issue #8034 reported by pcardozo)

        * apps/app_chanspy.c, main/channel.c, main/slinfactory.c: Slight
          overhaul of the whisper support. 1. We need to duplicate the
          frame from ast_translate 2. We need to ensure we always have
          signed linear coming in for signed linear combining. 3. We need
          to ensure we are always feeding signed linear out. 4. Properly
          store and restore write format when beeping on the channel we are
          whispering on. 5. Properly discontinue the stream on the channel
          for the beep. (issue #8019 reported by timkelly1980)

2006-09-26 18:34 +0000 [r43676]  Kevin P. Fleming <kpfleming@digium.com>

        * sounds/Makefile: update to use 1.4.3 core sounds, with corrected
          beep/beeperr/tt-monkeys files

2006-09-26 18:08 +0000 [r43650-43674]  Jason Parker <jparker@digium.com>

        * doc/rtp-packetization.txt, main/frame.c: Issue #8015, patch by
          Dan Austin. Maximum values were incorrect, which is why this is
          being put in 1.4

        * channels/chan_skinny.c: Add proper codec support to chan_skinny.
          Works with at least ulaw, alaw, and g729a. This is technically a
          "new feature", but there are justifications for it. I found a bug
          with the recent rtp packetization changes, which caused the media
          setup to fail under certain circumstances, particularly when
          using allow=all, or having no allow= statements (globally or on
          the device). I could have either removed the rtp packetization
          features, or I could add proper codec support (which, without, I
          think most people would consider to be a bug anyways).

2006-09-25 22:07 +0000 [r43640-43642]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_voicemail.c: Should have moved these lines up in the
          merge, instead of removing them

        * apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue 7824): 1)
          delete=yes was ignored 2) maxmessages was ignored

2006-09-25 21:26 +0000 [r43626-43635]  Paul Cadach <paul@odt.east.telecom.kz>

        * channels/h323/cisco-h225.cxx, channels/h323/cisco-h225.h,
          channels/h323/cisco-h225.asn: Fix ASN1 description of
          non-standard Cisco extensions

        * channels/h323/ast_h323.cxx, channels/chan_h323.c: Backport
          changes of trunk: 1) r43540: Avoid possible deadlock on channel
          destruction 2) r43590: Disable fastStart if requested by remote
          side

2006-09-25 15:23 +0000 [r43616]  Jason Parker <jparker@digium.com>

        * sounds/Makefile: One more fix for sounds installation - this time
          for portability. Reported to asterisk-dev mailing list.

2006-09-25 14:52 +0000 [r43605]  Steve Murphy <murf@digium.com>

        * formats/format_ogg_vorbis.c: This tiny fix prevents asterisk from
          crashing if trying to play an OGG moh file.

2006-09-25 06:15 +0000 [r43582]  Paul Cadach <paul@odt.east.telecom.kz>

        * channels/h323/caps_h323.cxx, channels/h323/compat_h323.h,
          channels/chan_h323.c: Merged revisions 43472,43495 from trunk

2006-09-24 14:58 +0000 [r43553-43564]  Russell Bryant <russell@digium.com>

        * channels/iax2-provision.c: Fix a CLI command registration issue
          where an erroneous message claiming that "iax2 show provisioning"
          was already registered. This was because this command was
          registering itself as both the command, as well as the command it
          is deprecating. (issue #8022, reported by bjweeks, fixed by
          myself)

        * channels/chan_iax2.c:Check to see if the channel that is activating the
          IAXPEER function is actually an IAX2 channel before proceeding to
          process it to avoid crashing. (issue #8017, reported by admott,
          fixed by myself)

2006-09-22 23:44 +0000 [r43524]  Kevin P. Fleming <kpfleming@digium.com>

        * Makefile: don't output the 'build complete' message when the
          target being run is already going to do an installation

2006-09-22 22:12 +0000 [r43518]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: Allow chan_skinny.so to be unloaded
          properly. Remove reload support, since it doesn't
          actually...work.

2006-09-22 21:36 +0000 [r43505-43508]  Steve Murphy <murf@digium.com>

        * pbx/pbx_ael.c: This commits a change to return
          MODULE_LOAD_FAILURE on error, and SUCCESS (instead of 0) when all
          goes well for bug 8004

        * pbx/pbx_ael.c: If the extensions.ael file not found, or
          unreadable, we return AST_MODULE_LOAD_DECLINE, as per bug # 8004.

2006-09-22 17:25 +0000 [r43492]  Jason Parker <jparker@digium.com>

        * main/cli.c: Make sure we explicitly set the CLI command to not be
          deprecated, if it isn't.

2006-09-22 16:42 +0000 [r43486-43489]  Kevin P. Fleming <kpfleming@digium.com>

        * sounds/Makefile: use rebuilt extra sounds

        * main/channel.c: all the Linux systems I have don't use
          '__m_count' for this field, so I don't know where this came
          from...

2006-09-22 15:47 +0000 [r43477-43484]  Russell Bryant <russell@digium.com>

        * include/asterisk/threadstorage.h: backport the compatability fix
          to use attribute_malloc instaed of __attribute__ ((malloc))

        * channels/chan_misdn.c: return AST_MODULE_LOAD_DECLIDE if mISDN
          could not be configured (issue #8006, Mithraen)

        * main/frame.c: Suppress a compiler warning about the use of a
          potentially uninitialized variable. It couldn't actually happen,
          though.

2006-09-22 03:01 +0000 [r43469]  Jason Parker <jparker@digium.com>

        * channels/chan_skinny.c: First shot at unload_module in
          chan_skinny.. More to come.

2006-09-21 23:50 +0000 [r43466]  Matt O'Gorman <mogorman@digium.com>

        * include/asterisk/jabber.h, channels/chan_gtalk.c,
          res/res_jabber.c: updates for better compontent support

2006-09-21 23:24 +0000 [r43464]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * res/res_odbc.c, configs/res_odbc.conf.sample: Twould help if we
          actually documented how the new features in res_odbc actually
          work. (Oops)

2006-09-21 22:21 +0000 [r43454-43456]  Joshua Colp <jcolp@digium.com>

        * channels/chan_oss.c: Some more clean up in the load function for
          chan_oss (issue #8002 reported by Mithraen with minor mods by
          moi)

        * channels/chan_mgcp.c: Clean up chan_mgcp's module load function
          (issue #8001 reported by Mithraen with mods by moi)

2006-09-21 21:21 +0000 [r43450]  Kevin P. Fleming <kpfleming@digium.com>

        * main/Makefile, build_tools/strip_nonapi (added): add another
          attempt to strip non-API symbols from the final binary... script
          will need to be extended to work on non-Linux systems

2006-09-21 20:22 +0000 [r43410-43445]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

        * apps/app_url.c: Fix documentation to reflect how Url() really
          works

        * cdr/cdr_tds.c, configure, configure.ac: TDS 0.64 updates

2006-09-21  Kevin P. Fleming  <kpfleming@digium.com>

        * Asterisk 1.4.0-beta2 released.

2006-09-21 16:08 +0000 [r43404-43405]  Kevin P. Fleming <kpfleming@digium.com>

        * main/Makefile: remove this change... it requires binutils 2.17

2006-09-20 23:19 +0000 [r43396]  Jason Parker <jparker@digium.com>

        * build_tools/make_version: fix minor typo in the way version is
          handled

2006-09-20  Kevin P. Fleming  <kpfleming@digium.com>

        * Asterisk 1.4.0-beta1 released.