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1 ;
2 ; SIP Configuration example for Asterisk
3 ;
4 ; Syntax for specifying a SIP device in extensions.conf is
5 ; SIP/devicename where devicename is defined in a section below.
6 ;
7 ; You may also use
8 ; SIP/username@domain to call any SIP user on the Internet
9 ; (Don't forget to enable DNS SRV records if you want to use this)
10 ;
11 ; If you define a SIP proxy as a peer below, you may call
12 ; SIP/proxyhostname/user or SIP/user@proxyhostname
13 ; where the proxyhostname is defined in a section below
14 ;
15 ; Useful CLI commands to check peers/users:
16 ; sip show peers Show all SIP peers (including friends)
17 ; sip show users Show all SIP users (including friends)
18 ; sip show registry Show status of hosts we register with
19 ;
20 ; sip debug Show all SIP messages
21 ;
22 ; module reload chan_sip.so Reload configuration file
23 ; Active SIP peers will not be reconfigured
24 ;
26 [general]
27 context=default ; Default context for incoming calls
28 ;allowguest=no ; Allow or reject guest calls (default is yes)
29 allowoverlap=no ; Disable overlap dialing support. (Default is yes)
30 ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
31 ; Default is enabled
32 ;realm=mydomain.tld ; Realm for digest authentication
33 ; defaults to "asterisk". If you set a system name in
34 ; asterisk.conf, it defaults to that system name
35 ; Realms MUST be globally unique according to RFC 3261
36 ; Set this to your host name or domain name
37 bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
38 ; bindport is the local UDP port that Asterisk will listen on
39 bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
40 srvlookup=yes ; Enable DNS SRV lookups on outbound calls
41 ; Note: Asterisk only uses the first host
42 ; in SRV records
43 ; Disabling DNS SRV lookups disables the
44 ; ability to place SIP calls based on domain
45 ; names to some other SIP users on the Internet
47 ;pedantic=yes ; Enable checking of tags in headers,
48 ; international character conversions in URIs
49 ; and multiline formatted headers for strict
50 ; SIP compatibility (defaults to "no")
52 ; See doc/ip-tos.txt for a description of these parameters.
53 ;tos_sip=cs3 ; Sets TOS for SIP packets.
54 ;tos_audio=ef ; Sets TOS for RTP audio packets.
55 ;tos_video=af41 ; Sets TOS for RTP video packets.
57 ;maxexpiry=3600 ; Maximum allowed time of incoming registrations
58 ; and subscriptions (seconds)
59 ;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
60 ;defaultexpiry=120 ; Default length of incoming/outgoing registration
61 ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
62 ; Defaults to 100 ms
63 ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
64 ;checkmwi=10 ; Default time between mailbox checks for peers
65 ;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
66 ; fully. Enable this option to not get error messages
67 ; when sending MWI to phones with this bug.
68 ;vmexten=voicemail ; dialplan extension to reach mailbox sets the
69 ; Message-Account in the MWI notify message
70 ; defaults to "asterisk"
71 ;disallow=all ; First disallow all codecs
72 ;allow=ulaw ; Allow codecs in order of preference
73 ;allow=ilbc ; see doc/rtp-packetization for framing options
75 ; This option specifies a preference for which music on hold class this channel
76 ; should listen to when put on hold if the music class has not been set on the
77 ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
78 ; channel putting this one on hold did not suggest a music class.
79 ;
80 ; This option may be specified globally, or on a per-user or per-peer basis.
81 ;
82 ;mohinterpret=default
83 ;
84 ; This option specifies which music on hold class to suggest to the peer channel
85 ; when this channel places the peer on hold. It may be specified globally or on
86 ; a per-user or per-peer basis.
87 ;
88 ;mohsuggest=default
89 ;
90 ;language=en ; Default language setting for all users/peers
91 ; This may also be set for individual users/peers
92 ;relaxdtmf=yes ; Relax dtmf handling
93 ;trustrpid = no ; If Remote-Party-ID should be trusted
94 ;sendrpid = yes ; If Remote-Party-ID should be sent
95 ;progressinband=never ; If we should generate in-band ringing always
96 ; use 'never' to never use in-band signalling, even in cases
97 ; where some buggy devices might not render it
98 ; Valid values: yes, no, never Default: never
99 ;useragent=Asterisk PBX ; Allows you to change the user agent string
100 ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
101 ; Note that promiscredir when redirects are made to the
102 ; local system will cause loops since Asterisk is incapable
103 ; of performing a "hairpin" call.
104 ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
105 ; a valid phone number
106 ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
107 ; Other options:
108 ; info : SIP INFO messages
109 ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
110 ; auto : Use rfc2833 if offered, inband otherwise
112 ;compactheaders = yes ; send compact sip headers.
113 ;
114 ;videosupport=yes ; Turn on support for SIP video. You need to turn this on
115 ; in the this section to get any video support at all.
116 ; You can turn it off on a per peer basis if the general
117 ; video support is enabled, but you can't enable it for
118 ; one peer only without enabling in the general section.
119 ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
120 ; Videosupport and maxcallbitrate is settable
121 ; for peers and users as well
122 ;callevents=no ; generate manager events when sip ua
123 ; performs events (e.g. hold)
124 ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
125 ; for any reason, always reject with '401 Unauthorized'
126 ; instead of letting the requester know whether there was
127 ; a matching user or peer for their request
129 ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
130 ; order instead of RFC3551 packing order (this is required
131 ; for Sipura and Grandstream ATAs, among others). This is
132 ; contrary to the RFC3551 specification, the peer _should_
133 ; be negotiating AAL2-G726-32 instead :-(
135 ;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
136 ; your localnet setting. Unless you have some sort of strange network
137 ; setup you will not need to enable this.
139 ;
140 ; If regcontext is specified, Asterisk will dynamically create and destroy a
141 ; NoOp priority 1 extension for a given peer who registers or unregisters with
142 ; us and have a "regexten=" configuration item.
143 ; Multiple contexts may be specified by separating them with '&'. The
144 ; actual extension is the 'regexten' parameter of the registering peer or its
145 ; name if 'regexten' is not provided. If more than one context is provided,
146 ; the context must be specified within regexten by appending the desired
147 ; context after '@'. More than one regexten may be supplied if they are
148 ; separated by '&'. Patterns may be used in regexten.
149 ;
150 ;regcontext=sipregistrations
151 ;
152 ;--------------------------- RTP timers ----------------------------------------------------
153 ; These timers are currently used for both audio and video streams. The RTP timeouts
154 ; are only applied to the audio channel.
155 ; The settings are settable in the global section as well as per device
156 ;
157 ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
158 ; on the audio channel
159 ; when we're not on hold. This is to be able to hangup
160 ; a call in the case of a phone disappearing from the net,
161 ; like a powerloss or grandma tripping over a cable.
162 ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
163 ; on the audio channel
164 ; when we're on hold (must be > rtptimeout)
165 ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
166 ; (default is off - zero)
167 ;--------------------------- SIP DEBUGGING ---------------------------------------------------
168 ;sipdebug = yes ; Turn on SIP debugging by default, from
169 ; the moment the channel loads this configuration
170 ;recordhistory=yes ; Record SIP history by default
171 ; (see sip history / sip no history)
172 ;dumphistory=yes ; Dump SIP history at end of SIP dialogue
173 ; SIP history is output to the DEBUG logging channel
176 ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
177 ; You can subscribe to the status of extensions with a "hint" priority
178 ; (See extensions.conf.sample for examples)
179 ; chan_sip support two major formats for notifications: dialog-info and SIMPLE
180 ;
181 ; You will get more detailed reports (busy etc) if you have a call limit set
182 ; for a device. When the call limit is filled, we will indicate busy. Note that
183 ; you need at least 2 in order to be able to do attended transfers.
184 ;
185 ; For queues, you will need this level of detail in status reporting, regardless
186 ; if you use SIP subscriptions. Queues and manager use the same internal interface
187 ; for reading status information.
188 ;
189 ; Note: Subscriptions does not work if you have a realtime dialplan and use the
190 ; realtime switch.
191 ;
192 ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
193 ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
194 ; Useful to limit subscriptions to local extensions
195 ; Settable per peer/user also
196 ;notifyringing = yes ; Control whether subscriptions already INUSE get sent
197 ; RINGING when another call is sent (default: no)
198 ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
199 ; Turning on notifyringing and notifyhold will add a lot
200 ; more database transactions if you are using realtime.
201 ;limitonpeers = yes ; Apply call limits on peers only. This will improve
202 ; status notification when you are using type=friend
203 ; Inbound calls, that really apply to the user part
204 ; of a friend will now be added to and compared with
205 ; the peer limit instead of applying two call limits,
206 ; one for the peer and one for the user.
207 ; "sip show inuse" will only show active calls on
208 ; the peer side of a "type=friend" object if this
209 ; setting is turned on.
211 ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
212 ;
213 ; This setting is available in the [general] section as well as in device configurations.
214 ; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
215 ; both parties have T38 support enabled in their Asterisk configuration
216 ; This has to be enabled in the general section for all devices to work. You can then
217 ; disable it on a per device basis.
218 ;
219 ; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.
220 ;
221 ; t38pt_udptl = yes ; Default false
222 ;
223 ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
224 ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
225 ; Format for the register statement is:
226 ; register => user[:secret[:authuser]]@host[:port][/extension]
227 ;
228 ; If no extension is given, the 's' extension is used. The extension needs to
229 ; be defined in extensions.conf to be able to accept calls from this SIP proxy
230 ; (provider).
231 ;
232 ; host is either a host name defined in DNS or the name of a section defined
233 ; below.
234 ;
235 ; Examples:
236 ;
237 ;register => 1234:password@mysipprovider.com
238 ;
239 ; This will pass incoming calls to the 's' extension
240 ;
241 ;
242 ;register => 2345:password@sip_proxy/1234
243 ;
244 ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
245 ; connect to local extension 1234 in extensions.conf, default context,
246 ; unless you configure a [sip_proxy] section below, and configure a
247 ; context.
248 ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
249 ; Tip 2: Use separate type=peer and type=user sections for SIP providers
250 ; (instead of type=friend) if you have calls in both directions
252 ;registertimeout=20 ; retry registration calls every 20 seconds (default)
253 ;registerattempts=10 ; Number of registration attempts before we give up
254 ; 0 = continue forever, hammering the other server
255 ; until it accepts the registration
256 ; Default is 0 tries, continue forever
258 ;----------------------------------------- NAT SUPPORT ------------------------
259 ; The externip, externhost and localnet settings are used if you use Asterisk
260 ; behind a NAT device to communicate with services on the outside.
262 ;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP
263 ; messages if we're behind a NAT
265 ; The externip and localnet is used
266 ; when registering and communicating with other proxies
267 ; that we're registered with
268 ;externhost=foo.dyndns.net ; Alternatively you can specify an
269 ; external host, and Asterisk will
270 ; perform DNS queries periodically. Not
271 ; recommended for production
272 ; environments! Use externip instead
273 ;externrefresh=10 ; How often to refresh externhost if
274 ; used
275 ; You may add multiple local networks. A reasonable
276 ; set of defaults are:
277 ;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
278 ;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
279 ;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
280 ;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
282 ; The nat= setting is used when Asterisk is on a public IP, communicating with
283 ; devices hidden behind a NAT device (broadband router). If you have one-way
284 ; audio problems, you usually have problems with your NAT configuration or your
285 ; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
286 ; ports for incoming audio in rtp.conf
287 ;
288 ;nat=no ; Global NAT settings (Affects all peers and users)
289 ; yes = Always ignore info and assume NAT
290 ; no = Use NAT mode only according to RFC3581 (;rport)
291 ; never = Never attempt NAT mode or RFC3581 support
292 ; route = Assume NAT, don't send rport
293 ; (work around more UNIDEN bugs)
295 ;----------------------------------- MEDIA HANDLING --------------------------------
296 ; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
297 ; no reason for Asterisk to stay in the media path, the media will be redirected.
298 ; This does not really work with in the case where Asterisk is outside and have
299 ; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
300 ;
301 ;canreinvite=yes ; Asterisk by default tries to redirect the
302 ; RTP media stream (audio) to go directly from
303 ; the caller to the callee. Some devices do not
304 ; support this (especially if one of them is behind a NAT).
305 ; The default setting is YES. If you have all clients
306 ; behind a NAT, or for some other reason wants Asterisk to
307 ; stay in the audio path, you may want to turn this off.
309 ; In Asterisk 1.4 this setting also affect direct RTP
310 ; at call setup (a new feature in 1.4 - setting up the
311 ; call directly between the endpoints instead of sending
312 ; a re-INVITE).
314 ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
315 ; the call directly with media peer-2-peer without re-invites.
316 ; Will not work for video and cases where the callee sends
317 ; RTP payloads and fmtp headers in the 200 OK that does not match the
318 ; callers INVITE. This will also fail if canreinvite is enabled when
319 ; the device is actually behind NAT.
321 ;canreinvite=nonat ; An additional option is to allow media path redirection
322 ; (reinvite) but only when the peer where the media is being
323 ; sent is known to not be behind a NAT (as the RTP core can
324 ; determine it based on the apparent IP address the media
325 ; arrives from).
327 ;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
328 ; instead of INVITE. This can be combined with 'nonat', as
329 ; 'canreinvite=update,nonat'. It implies 'yes'.
331 ;----------------------------------------- REALTIME SUPPORT ------------------------
332 ; For additional information on ARA, the Asterisk Realtime Architecture,
333 ; please read realtime.txt and extconfig.txt in the /doc directory of the
334 ; source code.
335 ;
336 ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
337 ; just like friends added from the config file only on a
338 ; as-needed basis? (yes|no)
340 ;rtsavesysname=yes ; Save systemname in realtime database at registration
341 ; Default= no
343 ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
344 ; If set to yes, when a SIP UA registers successfully, the ip address,
345 ; the origination port, the registration period, and the username of
346 ; the UA will be set to database via realtime.
347 ; If not present, defaults to 'yes'. Note: realtime peers will
348 ; probably not function across reloads in the way that you expect, if
349 ; you turn this option off.
350 ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
351 ; as if it had just registered? (yes|no|<seconds>)
352 ; If set to yes, when the registration expires, the friend will
353 ; vanish from the configuration until requested again. If set
354 ; to an integer, friends expire within this number of seconds
355 ; instead of the registration interval.
357 ;ignoreregexpire=yes ; Enabling this setting has two functions:
358 ;
359 ; For non-realtime peers, when their registration expires, the
360 ; information will _not_ be removed from memory or the Asterisk database
361 ; if you attempt to place a call to the peer, the existing information
362 ; will be used in spite of it having expired
363 ;
364 ; For realtime peers, when the peer is retrieved from realtime storage,
365 ; the registration information will be used regardless of whether
366 ; it has expired or not; if it expires while the realtime peer
367 ; is still in memory (due to caching or other reasons), the
368 ; information will not be removed from realtime storage
370 ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
371 ; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
372 ; domains, each of which can direct the call to a specific context if desired.
373 ; By default, all domains are accepted and sent to the default context or the
374 ; context associated with the user/peer placing the call.
375 ; REGISTER to non-local domains will be automatically denied if a domain
376 ; list is configured.
377 ;
378 ; Domains can be specified using:
379 ; domain=<domain>[,<context>]
380 ; Examples:
381 ; domain=myasterisk.dom
382 ; domain=customer.com,customer-context
383 ;
384 ; In addition, all the 'default' domains associated with a server should be
385 ; added if incoming request filtering is desired.
386 ; autodomain=yes
387 ;
388 ; To disallow requests for domains not serviced by this server:
389 ; allowexternaldomains=no
391 ;domain=mydomain.tld,mydomain-incoming
392 ; Add domain and configure incoming context
393 ; for external calls to this domain
394 ;domain=1.2.3.4 ; Add IP address as local domain
395 ; You can have several "domain" settings
396 ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
397 ; Default is yes
398 ;autodomain=yes ; Turn this on to have Asterisk add local host
399 ; name and local IP to domain list.
401 ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
402 ; non-peers, use your primary domain "identity"
403 ; for From: headers instead of just your IP
404 ; address. This is to be polite and
405 ; it may be a mandatory requirement for some
406 ; destinations which do not have a prior
407 ; account relationship with your server.
409 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
410 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
411 ; SIP channel. Defaults to "no". An enabled jitterbuffer will
412 ; be used only if the sending side can create and the receiving
413 ; side can not accept jitter. The SIP channel can accept jitter,
414 ; thus a jitterbuffer on the receive SIP side will be used only
415 ; if it is forced and enabled.
417 ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
418 ; channel. Defaults to "no".
420 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
422 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
423 ; resynchronized. Useful to improve the quality of the voice, with
424 ; big jumps in/broken timestamps, usually sent from exotic devices
425 ; and programs. Defaults to 1000.
427 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
428 ; channel. Two implementations are currently available - "fixed"
429 ; (with size always equals to jbmaxsize) and "adaptive" (with
430 ; variable size, actually the new jb of IAX2). Defaults to fixed.
432 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
433 ;-----------------------------------------------------------------------------------
435 [authentication]
436 ; Global credentials for outbound calls, i.e. when a proxy challenges your
437 ; Asterisk server for authentication. These credentials override
438 ; any credentials in peer/register definition if realm is matched.
439 ;
440 ; This way, Asterisk can authenticate for outbound calls to other
441 ; realms. We match realm on the proxy challenge and pick an set of
442 ; credentials from this list
443 ; Syntax:
444 ; auth = <user>:<secret>@<realm>
445 ; auth = <user>#<md5secret>@<realm>
446 ; Example:
447 ;auth=mark:topsecret@digium.com
448 ;
449 ; You may also add auth= statements to [peer] definitions
450 ; Peer auth= override all other authentication settings if we match on realm
452 ;------------------------------------------------------------------------------
453 ; Users and peers have different settings available. Friends have all settings,
454 ; since a friend is both a peer and a user
455 ;
456 ; User config options: Peer configuration:
457 ; -------------------- -------------------
458 ; context context
459 ; callingpres callingpres
460 ; permit permit
461 ; deny deny
462 ; secret secret
463 ; md5secret md5secret
464 ; dtmfmode dtmfmode
465 ; canreinvite canreinvite
466 ; nat nat
467 ; callgroup callgroup
468 ; pickupgroup pickupgroup
469 ; language language
470 ; allow allow
471 ; disallow disallow
472 ; insecure insecure
473 ; trustrpid trustrpid
474 ; progressinband progressinband
475 ; promiscredir promiscredir
476 ; useclientcode useclientcode
477 ; accountcode accountcode
478 ; setvar setvar
479 ; callerid callerid
480 ; amaflags amaflags
481 ; call-limit call-limit
482 ; allowoverlap allowoverlap
483 ; allowsubscribe allowsubscribe
484 ; allowtransfer allowtransfer
485 ; subscribecontext subscribecontext
486 ; videosupport videosupport
487 ; maxcallbitrate maxcallbitrate
488 ; rfc2833compensate mailbox
489 ; t38pt_usertpsource username
490 ; template
491 ; fromdomain
492 ; regexten
493 ; fromuser
494 ; host
495 ; port
496 ; qualify
497 ; defaultip
498 ; rtptimeout
499 ; rtpholdtimeout
500 ; sendrpid
501 ; outboundproxy
502 ; rfc2833compensate
503 ; t38pt_usertpsource
505 ;[sip_proxy]
506 ; For incoming calls only. Example: FWD (Free World Dialup)
507 ; We match on IP address of the proxy for incoming calls
508 ; since we can not match on username (caller id)
509 ;type=peer
510 ;context=from-fwd
511 ;host=fwd.pulver.com
513 ;[sip_proxy-out]
514 ;type=peer ; we only want to call out, not be called
515 ;secret=guessit
516 ;username=yourusername ; Authentication user for outbound proxies
517 ;fromuser=yourusername ; Many SIP providers require this!
518 ;fromdomain=provider.sip.domain
519 ;host=box.provider.com
520 ;usereqphone=yes ; This provider requires ";user=phone" on URI
521 ;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer
522 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
523 ; Call-limits will not be enforced on real-time peers,
524 ; since they are not stored in-memory
525 ;port=80 ; The port number we want to connect to on the remote side
526 ; Also used as "defaultport" in combination with "defaultip" settings
528 ;------------------------------------------------------------------------------
529 ; Definitions of locally connected SIP devices
530 ;
531 ; type = user a device that authenticates to us by "from" field to place calls
532 ; type = peer a device we place calls to or that calls us and we match by host
533 ; type = friend two configurations (peer+user) in one
534 ;
535 ; For device names, we recommend using only a-z, numerics (0-9) and underscore
536 ;
537 ; For local phones, type=friend works most of the time
538 ;
539 ; If you have one-way audio, you probably have NAT problems.
540 ; If Asterisk is on a public IP, and the phone is inside of a NAT device
541 ; you will need to configure nat option for those phones.
542 ; Also, turn on qualify=yes to keep the nat session open
544 ;[grandstream1]
545 ;type=friend
546 ;context=from-sip ; Where to start in the dialplan when this phone calls
547 ;callerid=John Doe <1234> ; Full caller ID, to override the phones config
548 ; on incoming calls to Asterisk
549 ;host=192.168.0.23 ; we have a static but private IP address
550 ; No registration allowed
551 ;nat=no ; there is not NAT between phone and Asterisk
552 ;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
553 ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
554 ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
555 ; from the phone to asterisk
556 ; 1 for the explicit peer, 1 for the explicit user,
557 ; remember that a friend equals 1 peer and 1 user in
558 ; memory
559 ; This will affect your subscriptions as well.
560 ; There is no combined call counter for a "friend"
561 ; so there's currently no way in sip.conf to limit
562 ; to one inbound or outbound call per phone. Use
563 ; the group counters in the dial plan for that.
564 ;
565 ;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
566 ;disallow=all ; need to disallow=all before we can use allow=
567 ;allow=ulaw ; Note: In user sections the order of codecs
568 ; listed with allow= does NOT matter!
569 ;allow=alaw
570 ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
571 ;allow=g729 ; Pass-thru only unless g729 license obtained
572 ;callingpres=allowed_passed_screen ; Set caller ID presentation
573 ; See doc/callingpres.txt for more information
576 ;[xlite1]
577 ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
578 ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
579 ;type=friend
580 ;regexten=1234 ; When they register, create extension 1234
581 ;callerid="Jane Smith" <5678>
582 ;host=dynamic ; This device needs to register
583 ;nat=yes ; X-Lite is behind a NAT router
584 ;canreinvite=no ; Typically set to NO if behind NAT
585 ;disallow=all
586 ;allow=gsm ; GSM consumes far less bandwidth than ulaw
587 ;allow=ulaw
588 ;allow=alaw
589 ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
592 ;[snom]
593 ;type=friend ; Friends place calls and receive calls
594 ;context=from-sip ; Context for incoming calls from this user
595 ;secret=blah
596 ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
597 ;language=de ; Use German prompts for this user
598 ;host=dynamic ; This peer register with us
599 ;dtmfmode=inband ; Choices are inband, rfc2833, or info
600 ;defaultip=192.168.0.59 ; IP used until peer registers
601 ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
602 ;subscribemwi=yes ; Only send notifications if this phone
603 ; subscribes for mailbox notification
604 ;vmexten=voicemail ; dialplan extension to reach mailbox
605 ; sets the Message-Account in the MWI notify message
606 ; defaults to global vmexten which defaults to "asterisk"
607 ;disallow=all
608 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
611 ;[polycom]
612 ;type=friend ; Friends place calls and receive calls
613 ;context=from-sip ; Context for incoming calls from this user
614 ;secret=blahpoly
615 ;host=dynamic ; This peer register with us
616 ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
617 ;username=polly ; Username to use in INVITE until peer registers
618 ; Normally you do NOT need to set this parameter
619 ;disallow=all
620 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
621 ;progressinband=no ; Polycom phones don't work properly with "never"
624 ;[pingtel]
625 ;type=friend
626 ;secret=blah
627 ;host=dynamic
628 ;insecure=port ; Allow matching of peer by IP address without
629 ; matching port number
630 ;insecure=invite ; Do not require authentication of incoming INVITEs
631 ;insecure=port,invite ; (both)
632 ;qualify=1000 ; Consider it down if it's 1 second to reply
633 ; Helps with NAT session
634 ; qualify=yes uses default value
635 ;
636 ; Call group and Pickup group should be in the range from 0 to 63
637 ;
638 ;callgroup=1,3-4 ; We are in caller groups 1,3,4
639 ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
640 ;defaultip=192.168.0.60 ; IP address to use if peer has not registered
641 ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
642 ;permit=192.168.0.60/255.255.255.0
644 ;[cisco1]
645 ;type=friend
646 ;secret=blah
647 ;qualify=200 ; Qualify peer is no more than 200ms away
648 ;nat=yes ; This phone may be natted
649 ; Send SIP and RTP to the IP address that packet is
650 ; received from instead of trusting SIP headers
651 ;host=dynamic ; This device registers with us
652 ;canreinvite=no ; Asterisk by default tries to redirect the
653 ; RTP media stream (audio) to go directly from
654 ; the caller to the callee. Some devices do not
655 ; support this (especially if one of them is
656 ; behind a NAT).
657 ;defaultip=192.168.0.4 ; IP address to use until registration
658 ;username=goran ; Username to use when calling this device before registration
659 ; Normally you do NOT need to set this parameter
660 ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
662 ;[pre14-asterisk]
663 ;type=friend
664 ;secret=digium
665 ;host=dynamic
666 ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
667 ; You must have this turned on or DTMF reception will work improperly.
668 ;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
669 ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
670 ; external IP address of the remote device. If port forwarding is done at the client side
671 ; then UDPTL will flow to the remote device.