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1 ;
2 ; SIP Configuration example for Asterisk
3 ;
4 ; SIP dial strings
5 ;-----------------------------------------------------------
6 ; In the dialplan (extensions.conf) you can use several
7 ; syntaxes for dialing SIP devices.
8 ; SIP/devicename
9 ; SIP/username@domain (SIP uri)
10 ; SIP/username[:password[:md5secret[:authname]]]@host[:port]
11 ; SIP/devicename/extension
12 ;
13 ;
14 ; Devicename
15 ; devicename is defined as a peer in a section below.
16 ;
17 ; username@domain
18 ; Call any SIP user on the Internet
19 ; (Don't forget to enable DNS SRV records if you want to use this)
20 ;
21 ; devicename/extension
22 ; If you define a SIP proxy as a peer below, you may call
23 ; SIP/proxyhostname/user or SIP/user@proxyhostname
24 ; where the proxyhostname is defined in a section below
25 ; This syntax also works with ATA's with FXO ports
26 ;
27 ; SIP/username[:password[:md5secret[:authname]]]@host[:port]
28 ; This form allows you to specify password or md5secret and authname
29 ; without altering any authentication data in config.
30 ; Examples:
31 ;
32 ; SIP/*98@mysipproxy
33 ; SIP/sales:topsecret::account02@domain.com:5062
34 ; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
35 ;
36 ; All of these dial strings specify the SIP request URI.
37 ; In addition, you can specify a specific To: header by adding an
38 ; exclamation mark after the dial string, like
39 ;
40 ; SIP/sales@mysipproxy!sales@edvina.net
41 ;
42 ; CLI Commands
43 ; -------------------------------------------------------------
44 ; Useful CLI commands to check peers/users:
45 ; sip show peers Show all SIP peers (including friends)
46 ; sip show users Show all SIP users (including friends)
47 ; sip show registry Show status of hosts we register with
48 ;
49 ; sip set debug Show all SIP messages
50 ;
51 ; sip reload Reload configuration file
52 ; Active SIP peers will not be reconfigured
53 ;
55 ; ** Deprecated configuration options **
56 ; The "call-limit" configuation option is deprecated. It still works in
57 ; this version of Asterisk, but will disappear in the next version.
58 ; You are encouraged to use the dialplan groupcount functionality
59 ; to enforce call limits instead of using this channel-specific method.
60 ;
61 ; You can still set limits per device in sip.conf or in a database by using
62 ; "setvar" to set variables that can be used in the dialplan for various limits.
64 [general]
65 context=default ; Default context for incoming calls
66 ;allowguest=no ; Allow or reject guest calls (default is yes)
67 ;match_auth_username=yes ; if available, match user entry using the
68 ; 'username' field from the authentication line
69 ; instead of the From: field.
70 ;;
71 ;; hash table sizes. For maximum efficiency, adjust the following
72 ;; values to be slightly larger than the maximum number of users/peers.
73 ;; Too large, and space is wasted. Too small, and things will run slower.
74 ;; 563 is probably way too big for small (home) applications, but it
75 ;; should cover most small/medium sites.
76 ;; it is recommended to make the sizes be a prime number!
77 ;; This was internally set to 17 for small-memory applications...
78 ;; All tables default to 563, except when compiled in LOW_MEMORY mode,
79 ;; in which case, they default to 17. You can override this by uncommenting
80 ;; the following, and changing the values.
81 ;hash_users=563
82 ;hash_peers=563
83 ;hash_dialogs=563
85 allowoverlap=no ; Disable overlap dialing support. (Default is yes)
86 ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
87 ; Default is enabled
88 ;realm=mydomain.tld ; Realm for digest authentication
89 ; defaults to "asterisk". If you set a system name in
90 ; asterisk.conf, it defaults to that system name
91 ; Realms MUST be globally unique according to RFC 3261
92 ; Set this to your host name or domain name
93 bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
94 ; bindport is the local UDP port that Asterisk will listen on
95 bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
97 ;
98 ; Note that the TCP and TLS support for chan_sip is currently considered
99 ; experimental. Since it is new, all of the related configuration options are
100 ; subject to change in any release. If they are changed, the changes will
101 ; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
102 ;
103 tcpenable=yes ; Enable server for incoming TCP connections (default is yes)
104 tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
105 ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
107 ;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
108 ;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
109 ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
110 ; Remember that the IP address must match the common name (hostname) in the
111 ; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
113 ;tlscertfile=asterisk.pem ; Certificate file (*.pem only) to use for TLS connections
114 ; default is to look for "asterisk.pem" in current directory
116 ;tlscafile=</path/to/certificate>
117 ; If the server your connecting to uses a self signed certificate
118 ; you should have their certificate installed here so the code can
119 ; verify the authenticity of their certificate.
121 ;tlscadir=</path/to/ca/dir>
122 ; A directory full of CA certificates. The files must be named with
123 ; the CA subject name hash value.
124 ; (see man SSL_CTX_load_verify_locations for more info)
126 ;tlsdontverifyserver=[yes|no]
127 ; If set to yes, don't verify the servers certificate when acting as
128 ; a client. If you don't have the server's CA certificate you can
129 ; set this and it will connect without requiring tlscafile to be set.
130 ; Default is no.
132 ;tlscipher=<SSL cipher string>
133 ; A string specifying which SSL ciphers to use or not use
134 ; A list of valid SSL cipher strings can be found at:
135 ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
137 srvlookup=yes ; Enable DNS SRV lookups on outbound calls
138 ; Note: Asterisk only uses the first host
139 ; in SRV records
140 ; Disabling DNS SRV lookups disables the
141 ; ability to place SIP calls based on domain
142 ; names to some other SIP users on the Internet
144 ;domain=mydomain.tld ; Set default domain for this host
145 ; If configured, Asterisk will only allow
146 ; INVITE and REFER to non-local domains
147 ; Use "sip show domains" to list local domains
148 ;pedantic=yes ; Enable checking of tags in headers,
149 ; international character conversions in URIs
150 ; and multiline formatted headers for strict
151 ; SIP compatibility (defaults to "no")
153 ; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
154 ;tos_sip=cs3 ; Sets TOS for SIP packets.
155 ;tos_audio=ef ; Sets TOS for RTP audio packets.
156 ;tos_video=af41 ; Sets TOS for RTP video packets.
157 ;tos_text=af41 ; Sets TOS for RTP text packets.
159 ;cos_sip=3 ; Sets 802.1p priority for SIP packets.
160 ;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
161 ;cos_video=4 ; Sets 802.1p priority for RTP video packets.
162 ;cos_text=3 ; Sets 802.1p priority for RTP text packets.
164 ;maxexpiry=3600 ; Maximum allowed time of incoming registrations
165 ; and subscriptions (seconds)
166 ;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
167 ;defaultexpiry=120 ; Default length of incoming/outgoing registration
168 ;qualifyfreq=60 ; Qualification: How often to check for the
169 ; host to be up in seconds
170 ; Set to low value if you use low timeout for
171 ; NAT of UDP sessions
172 ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
173 ;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
174 ; fully. Enable this option to not get error messages
175 ; when sending MWI to phones with this bug.
176 ;vmexten=voicemail ; dialplan extension to reach mailbox sets the
177 ; Message-Account in the MWI notify message
178 ; defaults to "asterisk"
179 ;disallow=all ; First disallow all codecs
180 ;allow=ulaw ; Allow codecs in order of preference
181 ;allow=ilbc ; see doc/rtp-packetization for framing options
182 ;
183 ; This option specifies a preference for which music on hold class this channel
184 ; should listen to when put on hold if the music class has not been set on the
185 ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
186 ; channel putting this one on hold did not suggest a music class.
187 ;
188 ; This option may be specified globally, or on a per-user or per-peer basis.
189 ;
190 ;mohinterpret=default
191 ;
192 ; This option specifies which music on hold class to suggest to the peer channel
193 ; when this channel places the peer on hold. It may be specified globally or on
194 ; a per-user or per-peer basis.
195 ;
196 ;mohsuggest=default
197 ;
198 ;parkinglot=plaza ; Sets the default parking lot for call parking
199 ; This may also be set for individual users/peers
200 ; Parkinglots are configured in features.conf
201 ;language=en ; Default language setting for all users/peers
202 ; This may also be set for individual users/peers
203 ;relaxdtmf=yes ; Relax dtmf handling
204 ;trustrpid = no ; If Remote-Party-ID should be trusted
205 ;sendrpid = yes ; If Remote-Party-ID should be sent
206 ;progressinband=never ; If we should generate in-band ringing always
207 ; use 'never' to never use in-band signalling, even in cases
208 ; where some buggy devices might not render it
209 ; Valid values: yes, no, never Default: never
210 ;useragent=Asterisk PBX ; Allows you to change the user agent string
211 ; The default user agent string also contains the Asterisk
212 ; version. If you don't want to expose this, change the
213 ; useragent string.
214 ;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
215 ; Like the useragent parameter, the default user agent string
216 ; also contains the Asterisk version.
217 ;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
218 ; This field MUST NOT contain spaces
219 ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
220 ; Note that promiscredir when redirects are made to the
221 ; local system will cause loops since Asterisk is incapable
222 ; of performing a "hairpin" call.
223 ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
224 ; a valid phone number
225 ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
226 ; Other options:
227 ; info : SIP INFO messages (application/dtmf-relay)
228 ; shortinfo : SIP INFO messages (application/dtmf)
229 ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
230 ; auto : Use rfc2833 if offered, inband otherwise
232 ;compactheaders = yes ; send compact sip headers.
233 ;
234 ;videosupport=yes ; Turn on support for SIP video. You need to turn this on
235 ; in the this section to get any video support at all.
236 ; You can turn it off on a per peer basis if the general
237 ; video support is enabled, but you can't enable it for
238 ; one peer only without enabling in the general section.
239 ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
240 ; Videosupport and maxcallbitrate is settable
241 ; for peers and users as well
242 ;callevents=no ; generate manager events when sip ua
243 ; performs events (e.g. hold)
244 ;authfailureevents=no ; generate manager "peerstatus" events when peer can't
245 ; authenticate with Asterisk. Peerstatus will be "rejected".
246 ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
247 ; for any reason, always reject with '401 Unauthorized'
248 ; instead of letting the requester know whether there was
249 ; a matching user or peer for their request
251 ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
252 ; order instead of RFC3551 packing order (this is required
253 ; for Sipura and Grandstream ATAs, among others). This is
254 ; contrary to the RFC3551 specification, the peer _should_
255 ; be negotiating AAL2-G726-32 instead :-(
256 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
257 ;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
258 ;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
259 ;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
260 ; your localnet setting. Unless you have some sort of strange network
261 ; setup you will not need to enable this.
263 ;
264 ; If regcontext is specified, Asterisk will dynamically create and destroy a
265 ; NoOp priority 1 extension for a given peer who registers or unregisters with
266 ; us and have a "regexten=" configuration item.
267 ; Multiple contexts may be specified by separating them with '&'. The
268 ; actual extension is the 'regexten' parameter of the registering peer or its
269 ; name if 'regexten' is not provided. If more than one context is provided,
270 ; the context must be specified within regexten by appending the desired
271 ; context after '@'. More than one regexten may be supplied if they are
272 ; separated by '&'. Patterns may be used in regexten.
273 ;
274 ;regcontext=sipregistrations
275 ;regextenonqualify=yes ; Default "no"
276 ; If you have qualify on and the peer becomes unreachable
277 ; this setting will enforce inactivation of the regexten
278 ; extension for the peer
279 ;
280 ;--------------------------- SIP timers ----------------------------------------------------
281 ; These timers are used primarily in INVITE transactions.
282 ; The default for Timer T1 is 500 ms or the measured run-trip time between
283 ; Asterisk and the device if you have qualify=yes for the device.
284 ;
285 ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
286 ; Defaults to 100 ms
287 ;timert1=500 ; Default T1 timer
288 ; Defaults to 500 ms or the measured round-trip
289 ; time to a peer (qualify=yes).
290 ;timerb=32000 ; Call setup timer. If a provisional response is not received
291 ; in this amount of time, the call will autocongest
292 ; Defaults to 64*timert1
294 ;--------------------------- RTP timers ----------------------------------------------------
295 ; These timers are currently used for both audio and video streams. The RTP timeouts
296 ; are only applied to the audio channel.
297 ; The settings are settable in the global section as well as per device
298 ;
299 ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
300 ; on the audio channel
301 ; when we're not on hold. This is to be able to hangup
302 ; a call in the case of a phone disappearing from the net,
303 ; like a powerloss or grandma tripping over a cable.
304 ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
305 ; on the audio channel
306 ; when we're on hold (must be > rtptimeout)
307 ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
308 ; (default is off - zero)
310 ;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
311 ; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
312 ; This mechanism can detect and reclaim SIP channels that do not terminate through normal
313 ; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
314 ; The operation of Session-Timers is driven by the following configuration parameters:
315 ;
316 ; * session-timers - Session-Timers feature operates in the following three modes:
317 ; originate : Request and run session-timers always
318 ; accept : Run session-timers only when requested by other UA
319 ; refuse : Do not run session timers in any case
320 ; The default mode of operation is 'accept'.
321 ; * session-expires - Maximum session refresh interval in seconds. Defaults to 1800 secs.
322 ; * session-minse - Minimum session refresh interval in seconds. Defualts to 90 secs.
323 ; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
324 ;
325 ;session-timers=originate
326 ;session-expires=600
327 ;session-minse=90
328 ;session-refresher=uas
331 ;--------------------------- SIP DEBUGGING ---------------------------------------------------
332 ;sipdebug = yes ; Turn on SIP debugging by default, from
333 ; the moment the channel loads this configuration
334 ;recordhistory=yes ; Record SIP history by default
335 ; (see sip history / sip no history)
336 ;dumphistory=yes ; Dump SIP history at end of SIP dialogue
337 ; SIP history is output to the DEBUG logging channel
340 ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
341 ; You can subscribe to the status of extensions with a "hint" priority
342 ; (See extensions.conf.sample for examples)
343 ; chan_sip support two major formats for notifications: dialog-info and SIMPLE
344 ;
345 ; You will get more detailed reports (busy etc) if you have a call counter enabled
346 ; for a device.
347 ;
348 ; If you set the busylevel, we will indicate busy when we have a number of calls that
349 ; matches the busylevel treshold.
350 ;
351 ; For queues, you will need this level of detail in status reporting, regardless
352 ; if you use SIP subscriptions. Queues and manager use the same internal interface
353 ; for reading status information.
354 ;
355 ; Note: Subscriptions does not work if you have a realtime dialplan and use the
356 ; realtime switch.
357 ;
358 ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
359 ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
360 ; Useful to limit subscriptions to local extensions
361 ; Settable per peer/user also
362 ;notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
363 ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
364 ; Turning on notifyringing and notifyhold will add a lot
365 ; more database transactions if you are using realtime.
366 ;callcounter = yes ; Enable call counters on devices. This can be set per
367 ; device too.
368 ;counteronpeer = yes ; Apply call counting on peers only. This will improve
369 ; status notification when you are using type=friend
370 ; Inbound calls, that really apply to the user part
371 ; of a friend will now be added to and compared with
372 ; the peer counter instead of applying two call counters,
373 ; one for the peer and one for the user.
374 ; "sip show inuse" will only show active calls on
375 ; the peer side of a "type=friend" object if this
376 ; setting is turned on.
378 ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
379 ;
380 ; This setting is available in the [general] section as well as in device configurations.
381 ; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
382 ; both parties have T38 support enabled in their Asterisk configuration
383 ; This has to be enabled in the general section for all devices to work. You can then
384 ; disable it on a per device basis.
385 ;
386 ; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.
387 ;
388 ; t38pt_udptl = yes ; Default false
389 ;
390 ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
391 ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
392 ; Format for the register statement is:
393 ; register => [transport://]user[:secret[:authuser]]@host[:port][/extension][~expiry]
394 ;
395 ;
396 ;
397 ; If no extension is given, the 's' extension is used. The extension needs to
398 ; be defined in extensions.conf to be able to accept calls from this SIP proxy
399 ; (provider).
400 ;
401 ; host is either a host name defined in DNS or the name of a section defined
402 ; below.
403 ;
404 ; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
405 ; this is equivalent to having the following line in the general section:
406 ;
407 ; register => username:secret@host/callbackextension
408 ;
409 ; and more readable because you don't have to write the parameters in two places
410 ; (note that the "port" is ignored - this is a bug that should be fixed).
411 ;
412 ; Examples:
413 ;
414 ;register => 1234:password@mysipprovider.com
415 ;
416 ; This will pass incoming calls to the 's' extension
417 ;
418 ;
419 ;register => 2345:password@sip_proxy/1234
420 ;
421 ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
422 ; connect to local extension 1234 in extensions.conf, default context,
423 ; unless you configure a [sip_proxy] section below, and configure a
424 ; context.
425 ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
426 ; Tip 2: Use separate type=peer and type=user sections for SIP providers
427 ; (instead of type=friend) if you have calls in both directions
429 ;registertimeout=20 ; retry registration calls every 20 seconds (default)
430 ;registerattempts=10 ; Number of registration attempts before we give up
431 ; 0 = continue forever, hammering the other server
432 ; until it accepts the registration
433 ; Default is 0 tries, continue forever
435 ;----------------------------------------- NAT SUPPORT ------------------------
436 ;
437 ; WARNING: SIP operation behind a NAT is tricky and you really need
438 ; to read and understand well the following section.
439 ;
440 ; When Asterisk is behind a NAT device, the "local" address (and port) that
441 ; a socket is bound to has different values when seen from the inside or
442 ; from the outside of the NATted network. Unfortunately this address must
443 ; be communicated to the outside (e.g. in SIP and SDP messages), and in
444 ; order to determine the correct value Asterisk needs to know:
445 ;
446 ; + whether it is talking to someone "inside" or "outside" of the NATted network.
447 ; This is configured by assigning the "localnet" parameter with a list
448 ; of network addresses that are considered "inside" of the NATted network.
449 ; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
450 ; Multiple entries are allowed, e.g. a reasonable set is the following:
451 ;
452 ; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
453 ; localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
454 ; localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
455 ; localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
456 ;
457 ; + the "externally visible" address and port number to be used when talking
458 ; to a host outside the NAT. This information is derived by one of the
459 ; following (mutually exclusive) config file parameters:
460 ;
461 ; a. "externip = hostname[:port]" specifies a static address[:port] to
462 ; be used in SIP and SDP messages.
463 ; The hostname is looked up only once, when [re]loading sip.conf .
464 ; If a port number is not present, use the "bindport" value (which is
465 ; not guaranteed to work correctly, because a NAT box might remap the
466 ; port number as well as the address).
467 ; This approach can be useful if you have a NAT device where you can
468 ; configure the mapping statically. Examples:
469 ;
470 ; externip = 12.34.56.78 ; use this address.
471 ; externip = 12.34.56.78:9900 ; use this address and port.
472 ; externip = mynat.my.org:12600 ; Public address of my nat box.
473 ;
474 ; b. "externhost = hostname[:port]" is similar to "externip" except
475 ; that the hostname is looked up every "externrefresh" seconds
476 ; (default 10s). This can be useful when your NAT device lets you choose
477 ; the port mapping, but the IP address is dynamic.
478 ; Beware, you might suffer from service disruption when the name server
479 ; resolution fails. Examples:
480 ;
481 ; externhost=foo.dyndns.net ; refreshed periodically
482 ; externrefresh=180 ; change the refresh interval
483 ;
484 ; c. "stunaddr = stun.server[:port]" queries the STUN server specified
485 ; as an argument to obtain the external address/port.
486 ; Queries are also sent periodically every "externrefresh" seconds
487 ; (as a side effect, sending the query also acts as a keepalive for
488 ; the state entry on the nat box):
489 ;
490 ; stunaddr = foo.stun.com:3478
491 ; externrefresh = 15
492 ;
493 ; Note that at the moment all these mechanism work only for the SIP socket.
494 ; The IP address discovered with externip/externhost/STUN is reused for
495 ; media sessions as well, but the port numbers are not remapped so you
496 ; may still experience problems.
497 ;
498 ; NOTE 1: in some cases, NAT boxes will use different port numbers in
499 ; the internal<->external mapping. In these cases, the "externip" and
500 ; "externhost" might not help you configure addresses properly, and you
501 ; really need to use STUN.
502 ;
503 ; NOTE 2: when using "externip" or "externhost", the address part is
504 ; also used as the external address for media sessions.
505 ; If you use "stunaddr", STUN queries will be sent to the same server
506 ; also from media sockets, and this should permit a correct mapping of
507 ; the port numbers as well.
508 ;
509 ; In addition to the above, Asterisk has an additional "nat" parameter to
510 ; address NAT-related issues in incoming SIP or media sessions.
511 ; In particular, depending on the 'nat= ' settings described below, Asterisk
512 ; may override the address/port information specified in the SIP/SDP messages,
513 ; and use the information (sender address) supplied by the network stack instead.
514 ; However, this is only useful if the external traffic can reach us.
515 ; The following settings are allowed (both globally and in individual sections):
516 ;
517 ; nat = no ; default. Use NAT mode only according to RFC3581 (;rport)
518 ; nat = yes ; Always ignore info and assume NAT
519 ; nat = never ; Never attempt NAT mode or RFC3581 support
520 ; nat = route ; route = Assume NAT, don't send rport
521 ; ; (work around more UNIDEN bugs)
523 ;----------------------------------- MEDIA HANDLING --------------------------------
524 ; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
525 ; no reason for Asterisk to stay in the media path, the media will be redirected.
526 ; This does not really work with in the case where Asterisk is outside and have
527 ; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
528 ;
529 ;canreinvite=yes ; Asterisk by default tries to redirect the
530 ; RTP media stream (audio) to go directly from
531 ; the caller to the callee. Some devices do not
532 ; support this (especially if one of them is behind a NAT).
533 ; The default setting is YES. If you have all clients
534 ; behind a NAT, or for some other reason wants Asterisk to
535 ; stay in the audio path, you may want to turn this off.
537 ; This setting also affect direct RTP
538 ; at call setup (a new feature in 1.4 - setting up the
539 ; call directly between the endpoints instead of sending
540 ; a re-INVITE).
542 ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
543 ; the call directly with media peer-2-peer without re-invites.
544 ; Will not work for video and cases where the callee sends
545 ; RTP payloads and fmtp headers in the 200 OK that does not match the
546 ; callers INVITE. This will also fail if canreinvite is enabled when
547 ; the device is actually behind NAT.
549 ;canreinvite=nonat ; An additional option is to allow media path redirection
550 ; (reinvite) but only when the peer where the media is being
551 ; sent is known to not be behind a NAT (as the RTP core can
552 ; determine it based on the apparent IP address the media
553 ; arrives from).
555 ;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
556 ; instead of INVITE. This can be combined with 'nonat', as
557 ; 'canreinvite=update,nonat'. It implies 'yes'.
559 ;----------------------------------------- REALTIME SUPPORT ------------------------
560 ; For additional information on ARA, the Asterisk Realtime Architecture,
561 ; please read realtime.txt and extconfig.txt in the /doc directory of the
562 ; source code.
563 ;
564 ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
565 ; just like friends added from the config file only on a
566 ; as-needed basis? (yes|no)
568 ;rtsavesysname=yes ; Save systemname in realtime database at registration
569 ; Default= no
571 ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
572 ; If set to yes, when a SIP UA registers successfully, the ip address,
573 ; the origination port, the registration period, and the username of
574 ; the UA will be set to database via realtime.
575 ; If not present, defaults to 'yes'.
576 ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
577 ; as if it had just registered? (yes|no|<seconds>)
578 ; If set to yes, when the registration expires, the friend will
579 ; vanish from the configuration until requested again. If set
580 ; to an integer, friends expire within this number of seconds
581 ; instead of the registration interval.
583 ;ignoreregexpire=yes ; Enabling this setting has two functions:
584 ;
585 ; For non-realtime peers, when their registration expires, the
586 ; information will _not_ be removed from memory or the Asterisk database
587 ; if you attempt to place a call to the peer, the existing information
588 ; will be used in spite of it having expired
589 ;
590 ; For realtime peers, when the peer is retrieved from realtime storage,
591 ; the registration information will be used regardless of whether
592 ; it has expired or not; if it expires while the realtime peer
593 ; is still in memory (due to caching or other reasons), the
594 ; information will not be removed from realtime storage
596 ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
597 ; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
598 ; domains, each of which can direct the call to a specific context if desired.
599 ; By default, all domains are accepted and sent to the default context or the
600 ; context associated with the user/peer placing the call.
601 ; Domains can be specified using:
602 ; domain=<domain>[,<context>]
603 ; Examples:
604 ; domain=myasterisk.dom
605 ; domain=customer.com,customer-context
606 ;
607 ; In addition, all the 'default' domains associated with a server should be
608 ; added if incoming request filtering is desired.
609 ; autodomain=yes
610 ;
611 ; To disallow requests for domains not serviced by this server:
612 ; allowexternaldomains=no
614 ;domain=mydomain.tld,mydomain-incoming
615 ; Add domain and configure incoming context
616 ; for external calls to this domain
617 ;domain=1.2.3.4 ; Add IP address as local domain
618 ; You can have several "domain" settings
619 ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
620 ; Default is yes
621 ;autodomain=yes ; Turn this on to have Asterisk add local host
622 ; name and local IP to domain list.
624 ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
625 ; non-peers, use your primary domain "identity"
626 ; for From: headers instead of just your IP
627 ; address. This is to be polite and
628 ; it may be a mandatory requirement for some
629 ; destinations which do not have a prior
630 ; account relationship with your server.
632 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
633 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
634 ; SIP channel. Defaults to "no". An enabled jitterbuffer will
635 ; be used only if the sending side can create and the receiving
636 ; side can not accept jitter. The SIP channel can accept jitter,
637 ; thus a jitterbuffer on the receive SIP side will be used only
638 ; if it is forced and enabled.
640 ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
641 ; channel. Defaults to "no".
643 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
645 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
646 ; resynchronized. Useful to improve the quality of the voice, with
647 ; big jumps in/broken timestamps, usually sent from exotic devices
648 ; and programs. Defaults to 1000.
650 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
651 ; channel. Two implementations are currently available - "fixed"
652 ; (with size always equals to jbmaxsize) and "adaptive" (with
653 ; variable size, actually the new jb of IAX2). Defaults to fixed.
655 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
656 ;-----------------------------------------------------------------------------------
658 [authentication]
659 ; Global credentials for outbound calls, i.e. when a proxy challenges your
660 ; Asterisk server for authentication. These credentials override
661 ; any credentials in peer/register definition if realm is matched.
662 ;
663 ; This way, Asterisk can authenticate for outbound calls to other
664 ; realms. We match realm on the proxy challenge and pick an set of
665 ; credentials from this list
666 ; Syntax:
667 ; auth = <user>:<secret>@<realm>
668 ; auth = <user>#<md5secret>@<realm>
669 ; Example:
670 ;auth=mark:topsecret@digium.com
671 ;
672 ; You may also add auth= statements to [peer] definitions
673 ; Peer auth= override all other authentication settings if we match on realm
675 ;------------------------------------------------------------------------------
676 ; Users and peers have different settings available. Friends have all settings,
677 ; since a friend is both a peer and a user
678 ;
679 ; User config options: Peer configuration:
680 ; -------------------- -------------------
681 ; context context
682 ; callingpres callingpres
683 ; permit permit
684 ; deny deny
685 ; secret secret
686 ; md5secret md5secret
687 ; dtmfmode dtmfmode
688 ; canreinvite canreinvite
689 ; nat nat
690 ; callgroup callgroup
691 ; pickupgroup pickupgroup
692 ; language language
693 ; allow allow
694 ; disallow disallow
695 ; insecure insecure
696 ; trustrpid trustrpid
697 ; progressinband progressinband
698 ; promiscredir promiscredir
699 ; useclientcode useclientcode
700 ; accountcode accountcode
701 ; setvar setvar
702 ; callerid callerid
703 ; amaflags amaflags
704 ; call-limit call-limit (deprecated)
705 ; callcounter callcounter
706 ; allowoverlap allowoverlap
707 ; allowsubscribe allowsubscribe
708 ; allowtransfer allowtransfer
709 ; subscribecontext subscribecontext
710 ; videosupport videosupport
711 ; maxcallbitrate maxcallbitrate
712 ; rfc2833compensate mailbox
713 ; session-timers busylevel
714 ; session-expires
715 ; session-minse template
716 ; session-refresher fromdomain
717 ; t38pt_usertpsource regexten
718 ; fromuser
719 ; host
720 ; port
721 ; qualify
722 ; defaultip
723 ; defaultuser
724 ; rtptimeout
725 ; rtpholdtimeout
726 ; sendrpid
727 ; outboundproxy
728 ; rfc2833compensate
729 ; callbackextension
730 ; registertrying
731 ; session-timers
732 ; session-expires
733 ; session-minse
734 ; session-refresher
735 ; timert1
736 ; timerb
737 ; qualifyfreq
738 ; t38pt_usertpsource
740 ;[sip_proxy]
741 ; For incoming calls only. Example: FWD (Free World Dialup)
742 ; We match on IP address of the proxy for incoming calls
743 ; since we can not match on username (caller id)
744 ;type=peer
745 ;context=from-fwd
746 ;host=fwd.pulver.com
748 ;[sip_proxy-out]
749 ;type=peer ; we only want to call out, not be called
750 ;secret=guessit
751 ;defaultuser=yourusername ; Authentication user for outbound proxies
752 ;fromuser=yourusername ; Many SIP providers require this!
753 ;fromdomain=provider.sip.domain
754 ;host=box.provider.com
755 ;usereqphone=yes ; This provider requires ";user=phone" on URI
756 ;callcounter=yes ; Enable call counter
757 ;busylevel=2 ; Signal busy at 2 or more calls
758 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
759 ;port=80 ; The port number we want to connect to on the remote side
760 ; Also used as "defaultport" in combination with "defaultip" settings
762 ;--- sample definition for a provider
763 ;[provider1]
764 ;type=peer
765 ;host=sip.provider1.com
766 ;fromuser=4015552299 ; how your provider knows you
767 ;secret=youwillneverguessit
768 ;callbackextension=123 ; Register with this server and require calls coming back to this extension
770 ;------------------------------------------------------------------------------
771 ; Definitions of locally connected SIP devices
772 ;
773 ; type = user a device that authenticates to us by "from" field to place calls
774 ; type = peer a device we place calls to or that calls us and we match by host
775 ; type = friend two configurations (peer+user) in one
776 ;
777 ; For device names, we recommend using only a-z, numerics (0-9) and underscore
778 ;
779 ; For local phones, type=friend works most of the time
780 ;
781 ; If you have one-way audio, you probably have NAT problems.
782 ; If Asterisk is on a public IP, and the phone is inside of a NAT device
783 ; you will need to configure nat option for those phones.
784 ; Also, turn on qualify=yes to keep the nat session open
785 ;
786 ; Because you might have a large number of similar sections, it is generally
787 ; convenient to use templates for the common parameters, and add them
788 ; the the various sections. Examples are below, and we can even leave
789 ; the templates uncommented as they will not harm:
791 [basic-options](!) ; a template
792 dtmfmode=rfc2833
793 context=from-office
794 type=friend
796 [natted-phone](!,basic-options) ; another template inheriting basic-options
797 nat=yes
798 canreinvite=no
799 host=dynamic
801 [public-phone](!,basic-options) ; another template inheriting basic-options
802 nat=no
803 canreinvite=yes
805 [my-codecs](!) ; a template for my preferred codecs
806 disallow=all
807 allow=ilbc
808 allow=g729
809 allow=gsm
810 allow=g723
811 allow=ulaw
813 [ulaw-phone](!) ; and another one for ulaw-only
814 disallow=all
815 allow=ulaw
817 ; and finally instantiate a few phones
818 ;
819 ; [2133](natted-phone,my-codecs)
820 ; secret = peekaboo
821 ; [2134](natted-phone,ulaw-phone)
822 ; secret = not_very_secret
823 ; [2136](public-phone,ulaw-phone)
824 ; secret = not_very_secret_either
825 ; ...
826 ;
828 ; Standard configurations not using templates look like this:
829 ;
830 ;[grandstream1]
831 ;type=friend
832 ;context=from-sip ; Where to start in the dialplan when this phone calls
833 ;callerid=John Doe <1234> ; Full caller ID, to override the phones config
834 ; on incoming calls to Asterisk
835 ;host=192.168.0.23 ; we have a static but private IP address
836 ; No registration allowed
837 ;nat=no ; there is not NAT between phone and Asterisk
838 ;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
839 ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
840 ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
841 ; from the phone to asterisk (deprecated)
842 ; 1 for the explicit peer, 1 for the explicit user,
843 ; remember that a friend equals 1 peer and 1 user in
844 ; memory
845 ; There is no combined call counter for a "friend"
846 ; so there's currently no way in sip.conf to limit
847 ; to one inbound or outbound call per phone. Use
848 ; the group counters in the dial plan for that.
849 ;
850 ;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
851 ;disallow=all ; need to disallow=all before we can use allow=
852 ;allow=ulaw ; Note: In user sections the order of codecs
853 ; listed with allow= does NOT matter!
854 ;allow=alaw
855 ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
856 ;allow=g729 ; Pass-thru only unless g729 license obtained
857 ;callingpres=allowed_passed_screen ; Set caller ID presentation
858 ; See README.callingpres for more information
860 ;[xlite1]
861 ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
862 ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
863 ;type=friend
864 ;regexten=1234 ; When they register, create extension 1234
865 ;callerid="Jane Smith" <5678>
866 ;host=dynamic ; This device needs to register
867 ;nat=yes ; X-Lite is behind a NAT router
868 ;canreinvite=no ; Typically set to NO if behind NAT
869 ;disallow=all
870 ;allow=gsm ; GSM consumes far less bandwidth than ulaw
871 ;allow=ulaw
872 ;allow=alaw
873 ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
874 ;registertrying=yes ; Send a 100 Trying when the device registers.
876 ;[snom]
877 ;type=friend ; Friends place calls and receive calls
878 ;context=from-sip ; Context for incoming calls from this user
879 ;secret=blah
880 ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
881 ;language=de ; Use German prompts for this user
882 ;host=dynamic ; This peer register with us
883 ;dtmfmode=inband ; Choices are inband, rfc2833, or info
884 ;defaultip=192.168.0.59 ; IP used until peer registers
885 ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
886 ;subscribemwi=yes ; Only send notifications if this phone
887 ; subscribes for mailbox notification
888 ;vmexten=voicemail ; dialplan extension to reach mailbox
889 ; sets the Message-Account in the MWI notify message
890 ; defaults to global vmexten which defaults to "asterisk"
891 ;disallow=all
892 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
895 ;[polycom]
896 ;type=friend ; Friends place calls and receive calls
897 ;context=from-sip ; Context for incoming calls from this user
898 ;secret=blahpoly
899 ;host=dynamic ; This peer register with us
900 ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
901 ;defaultuser=polly ; Username to use in INVITE until peer registers
902 ;defaultip=192.168.40.123
903 ; Normally you do NOT need to set this parameter
904 ;disallow=all
905 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
906 ;progressinband=no ; Polycom phones don't work properly with "never"
909 ;[pingtel]
910 ;type=friend
911 ;secret=blah
912 ;host=dynamic
913 ;insecure=port ; Allow matching of peer by IP address without
914 ; matching port number
915 ;insecure=invite ; Do not require authentication of incoming INVITEs
916 ;insecure=port,invite ; (both)
917 ;qualify=1000 ; Consider it down if it's 1 second to reply
918 ; Helps with NAT session
919 ; qualify=yes uses default value
920 ;qualifyfreq=60 ; Qualification: How often to check for the
921 ; host to be up in seconds
922 ; Set to low value if you use low timeout for
923 ; NAT of UDP sessions
924 ;
925 ; Call group and Pickup group should be in the range from 0 to 63
926 ;
927 ;callgroup=1,3-4 ; We are in caller groups 1,3,4
928 ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
929 ;defaultip=192.168.0.60 ; IP address to use if peer has not registered
930 ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
931 ;permit=192.168.0.60/255.255.255.0
933 ;[cisco1]
934 ;type=friend
935 ;secret=blah
936 ;qualify=200 ; Qualify peer is no more than 200ms away
937 ;nat=yes ; This phone may be natted
938 ; Send SIP and RTP to the IP address that packet is
939 ; received from instead of trusting SIP headers
940 ;host=dynamic ; This device registers with us
941 ;canreinvite=no ; Asterisk by default tries to redirect the
942 ; RTP media stream (audio) to go directly from
943 ; the caller to the callee. Some devices do not
944 ; support this (especially if one of them is
945 ; behind a NAT).
946 ;defaultip=192.168.0.4 ; IP address to use until registration
947 ;defaultuser=goran ; Username to use when calling this device before registration
948 ; Normally you do NOT need to set this parameter
949 ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
950 ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will cause the given audio file to be played
951 ; upon completion of an attended transfer
953 ;[pre14-asterisk]
954 ;type=friend
955 ;secret=digium
956 ;host=dynamic
957 ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
958 ; You must have this turned on or DTMF reception will work improperly.
959 ;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
960 ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
961 ; external IP address of the remote device. If port forwarding is done at the client side
962 ; then UDPTL will flow to the remote device.