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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 ;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering
140 ; as any IP address used for staticly defined
141 ; hosts. This helps avoid the configuration
142 ; error of allowing your users to register at
143 ; the same address as a SIP provider.
145 ;contactdeny=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to
146 ;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may
147 ; register their phones.
149 ;
150 ; If regcontext is specified, Asterisk will dynamically create and destroy a
151 ; NoOp priority 1 extension for a given peer who registers or unregisters with
152 ; us and have a "regexten=" configuration item.
153 ; Multiple contexts may be specified by separating them with '&'. The
154 ; actual extension is the 'regexten' parameter of the registering peer or its
155 ; name if 'regexten' is not provided. If more than one context is provided,
156 ; the context must be specified within regexten by appending the desired
157 ; context after '@'. More than one regexten may be supplied if they are
158 ; separated by '&'. Patterns may be used in regexten.
159 ;
160 ;regcontext=sipregistrations
161 ;
162 ;--------------------------- RTP timers ----------------------------------------------------
163 ; These timers are currently used for both audio and video streams. The RTP timeouts
164 ; are only applied to the audio channel.
165 ; The settings are settable in the global section as well as per device
166 ;
167 ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
168 ; on the audio channel
169 ; when we're not on hold. This is to be able to hangup
170 ; a call in the case of a phone disappearing from the net,
171 ; like a powerloss or grandma tripping over a cable.
172 ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
173 ; on the audio channel
174 ; when we're on hold (must be > rtptimeout)
175 ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
176 ; (default is off - zero)
177 ;--------------------------- SIP DEBUGGING ---------------------------------------------------
178 ;sipdebug = yes ; Turn on SIP debugging by default, from
179 ; the moment the channel loads this configuration
180 ;recordhistory=yes ; Record SIP history by default
181 ; (see sip history / sip no history)
182 ;dumphistory=yes ; Dump SIP history at end of SIP dialogue
183 ; SIP history is output to the DEBUG logging channel
186 ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
187 ; You can subscribe to the status of extensions with a "hint" priority
188 ; (See extensions.conf.sample for examples)
189 ; chan_sip support two major formats for notifications: dialog-info and SIMPLE
190 ;
191 ; You will get more detailed reports (busy etc) if you have a call limit set
192 ; for a device. When the call limit is filled, we will indicate busy. Note that
193 ; you need at least 2 in order to be able to do attended transfers.
194 ;
195 ; For queues, you will need this level of detail in status reporting, regardless
196 ; if you use SIP subscriptions. Queues and manager use the same internal interface
197 ; for reading status information.
198 ;
199 ; Note: Subscriptions does not work if you have a realtime dialplan and use the
200 ; realtime switch.
201 ;
202 ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
203 ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
204 ; Useful to limit subscriptions to local extensions
205 ; Settable per peer/user also
206 ;notifyringing = yes ; Control whether subscriptions already INUSE get sent
207 ; RINGING when another call is sent (default: no)
208 ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
209 ; Turning on notifyringing and notifyhold will add a lot
210 ; more database transactions if you are using realtime.
211 ;limitonpeers = yes ; Apply call limits on peers only. This will improve
212 ; status notification when you are using type=friend
213 ; Inbound calls, that really apply to the user part
214 ; of a friend will now be added to and compared with
215 ; the peer limit instead of applying two call limits,
216 ; one for the peer and one for the user.
217 ; "sip show inuse" will only show active calls on
218 ; the peer side of a "type=friend" object if this
219 ; setting is turned on.
221 ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
222 ;
223 ; This setting is available in the [general] section as well as in device configurations.
224 ; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
225 ; both parties have T38 support enabled in their Asterisk configuration
226 ; This has to be enabled in the general section for all devices to work. You can then
227 ; disable it on a per device basis.
228 ;
229 ; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.
230 ;
231 ; t38pt_udptl = yes ; Default false
232 ;
233 ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
234 ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
235 ; Format for the register statement is:
236 ; register => user[:secret[:authuser]]@host[:port][/extension]
237 ;
238 ; If no extension is given, the 's' extension is used. The extension needs to
239 ; be defined in extensions.conf to be able to accept calls from this SIP proxy
240 ; (provider).
241 ;
242 ; host is either a host name defined in DNS or the name of a section defined
243 ; below.
244 ;
245 ; Examples:
246 ;
247 ;register => 1234:password@mysipprovider.com
248 ;
249 ; This will pass incoming calls to the 's' extension
250 ;
251 ;
252 ;register => 2345:password@sip_proxy/1234
253 ;
254 ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
255 ; connect to local extension 1234 in extensions.conf, default context,
256 ; unless you configure a [sip_proxy] section below, and configure a
257 ; context.
258 ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
259 ; Tip 2: Use separate type=peer and type=user sections for SIP providers
260 ; (instead of type=friend) if you have calls in both directions
262 ;registertimeout=20 ; retry registration calls every 20 seconds (default)
263 ;registerattempts=10 ; Number of registration attempts before we give up
264 ; 0 = continue forever, hammering the other server
265 ; until it accepts the registration
266 ; Default is 0 tries, continue forever
268 ;----------------------------------------- NAT SUPPORT ------------------------
269 ; The externip, externhost and localnet settings are used if you use Asterisk
270 ; behind a NAT device to communicate with services on the outside.
272 ;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP
273 ; messages if we're behind a NAT
275 ; The externip and localnet is used
276 ; when registering and communicating with other proxies
277 ; that we're registered with
278 ;externhost=foo.dyndns.net ; Alternatively you can specify an
279 ; external host, and Asterisk will
280 ; perform DNS queries periodically. Not
281 ; recommended for production
282 ; environments! Use externip instead
283 ;externrefresh=10 ; How often to refresh externhost if
284 ; used
285 ; You may add multiple local networks. A reasonable
286 ; set of defaults are:
287 ;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
288 ;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
289 ;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
290 ;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
292 ; The nat= setting is used when Asterisk is on a public IP, communicating with
293 ; devices hidden behind a NAT device (broadband router). If you have one-way
294 ; audio problems, you usually have problems with your NAT configuration or your
295 ; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
296 ; ports for incoming audio in rtp.conf
297 ;
298 ;nat=no ; Global NAT settings (Affects all peers and users)
299 ; yes = Always ignore info and assume NAT
300 ; no = Use NAT mode only according to RFC3581 (;rport)
301 ; never = Never attempt NAT mode or RFC3581 support
302 ; route = Assume NAT, don't send rport
303 ; (work around more UNIDEN bugs)
305 ;----------------------------------- MEDIA HANDLING --------------------------------
306 ; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
307 ; no reason for Asterisk to stay in the media path, the media will be redirected.
308 ; This does not really work with in the case where Asterisk is outside and have
309 ; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
310 ;
311 ;canreinvite=yes ; Asterisk by default tries to redirect the
312 ; RTP media stream (audio) to go directly from
313 ; the caller to the callee. Some devices do not
314 ; support this (especially if one of them is behind a NAT).
315 ; The default setting is YES. If you have all clients
316 ; behind a NAT, or for some other reason wants Asterisk to
317 ; stay in the audio path, you may want to turn this off.
319 ; In Asterisk 1.4 this setting also affect direct RTP
320 ; at call setup (a new feature in 1.4 - setting up the
321 ; call directly between the endpoints instead of sending
322 ; a re-INVITE).
324 ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
325 ; the call directly with media peer-2-peer without re-invites.
326 ; Will not work for video and cases where the callee sends
327 ; RTP payloads and fmtp headers in the 200 OK that does not match the
328 ; callers INVITE. This will also fail if canreinvite is enabled when
329 ; the device is actually behind NAT.
331 ;canreinvite=nonat ; An additional option is to allow media path redirection
332 ; (reinvite) but only when the peer where the media is being
333 ; sent is known to not be behind a NAT (as the RTP core can
334 ; determine it based on the apparent IP address the media
335 ; arrives from).
337 ;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
338 ; instead of INVITE. This can be combined with 'nonat', as
339 ; 'canreinvite=update,nonat'. It implies 'yes'.
341 ;----------------------------------------- REALTIME SUPPORT ------------------------
342 ; For additional information on ARA, the Asterisk Realtime Architecture,
343 ; please read realtime.txt and extconfig.txt in the /doc directory of the
344 ; source code.
345 ;
346 ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
347 ; just like friends added from the config file only on a
348 ; as-needed basis? (yes|no)
350 ;rtsavesysname=yes ; Save systemname in realtime database at registration
351 ; Default= no
353 ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
354 ; If set to yes, when a SIP UA registers successfully, the ip address,
355 ; the origination port, the registration period, and the username of
356 ; the UA will be set to database via realtime.
357 ; If not present, defaults to 'yes'. Note: realtime peers will
358 ; probably not function across reloads in the way that you expect, if
359 ; you turn this option off.
360 ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
361 ; as if it had just registered? (yes|no|<seconds>)
362 ; If set to yes, when the registration expires, the friend will
363 ; vanish from the configuration until requested again. If set
364 ; to an integer, friends expire within this number of seconds
365 ; instead of the registration interval.
367 ;ignoreregexpire=yes ; Enabling this setting has two functions:
368 ;
369 ; For non-realtime peers, when their registration expires, the
370 ; information will _not_ be removed from memory or the Asterisk database
371 ; if you attempt to place a call to the peer, the existing information
372 ; will be used in spite of it having expired
373 ;
374 ; For realtime peers, when the peer is retrieved from realtime storage,
375 ; the registration information will be used regardless of whether
376 ; it has expired or not; if it expires while the realtime peer
377 ; is still in memory (due to caching or other reasons), the
378 ; information will not be removed from realtime storage
380 ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
381 ; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
382 ; domains, each of which can direct the call to a specific context if desired.
383 ; By default, all domains are accepted and sent to the default context or the
384 ; context associated with the user/peer placing the call.
385 ; REGISTER to non-local domains will be automatically denied if a domain
386 ; list is configured.
387 ;
388 ; Domains can be specified using:
389 ; domain=<domain>[,<context>]
390 ; Examples:
391 ; domain=myasterisk.dom
392 ; domain=customer.com,customer-context
393 ;
394 ; In addition, all the 'default' domains associated with a server should be
395 ; added if incoming request filtering is desired.
396 ; autodomain=yes
397 ;
398 ; To disallow requests for domains not serviced by this server:
399 ; allowexternaldomains=no
401 ;domain=mydomain.tld,mydomain-incoming
402 ; Add domain and configure incoming context
403 ; for external calls to this domain
404 ;domain=1.2.3.4 ; Add IP address as local domain
405 ; You can have several "domain" settings
406 ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
407 ; Default is yes
408 ;autodomain=yes ; Turn this on to have Asterisk add local host
409 ; name and local IP to domain list.
411 ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
412 ; non-peers, use your primary domain "identity"
413 ; for From: headers instead of just your IP
414 ; address. This is to be polite and
415 ; it may be a mandatory requirement for some
416 ; destinations which do not have a prior
417 ; account relationship with your server.
419 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
420 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
421 ; SIP channel. Defaults to "no". An enabled jitterbuffer will
422 ; be used only if the sending side can create and the receiving
423 ; side can not accept jitter. The SIP channel can accept jitter,
424 ; thus a jitterbuffer on the receive SIP side will be used only
425 ; if it is forced and enabled.
427 ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
428 ; channel. Defaults to "no".
430 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
432 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
433 ; resynchronized. Useful to improve the quality of the voice, with
434 ; big jumps in/broken timestamps, usually sent from exotic devices
435 ; and programs. Defaults to 1000.
437 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
438 ; channel. Two implementations are currently available - "fixed"
439 ; (with size always equals to jbmaxsize) and "adaptive" (with
440 ; variable size, actually the new jb of IAX2). Defaults to fixed.
442 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
443 ;-----------------------------------------------------------------------------------
445 [authentication]
446 ; Global credentials for outbound calls, i.e. when a proxy challenges your
447 ; Asterisk server for authentication. These credentials override
448 ; any credentials in peer/register definition if realm is matched.
449 ;
450 ; This way, Asterisk can authenticate for outbound calls to other
451 ; realms. We match realm on the proxy challenge and pick an set of
452 ; credentials from this list
453 ; Syntax:
454 ; auth = <user>:<secret>@<realm>
455 ; auth = <user>#<md5secret>@<realm>
456 ; Example:
457 ;auth=mark:topsecret@digium.com
458 ;
459 ; You may also add auth= statements to [peer] definitions
460 ; Peer auth= override all other authentication settings if we match on realm
462 ;------------------------------------------------------------------------------
463 ; Users and peers have different settings available. Friends have all settings,
464 ; since a friend is both a peer and a user
465 ;
466 ; User config options: Peer configuration:
467 ; -------------------- -------------------
468 ; context context
469 ; callingpres callingpres
470 ; permit permit
471 ; deny deny
472 ; secret secret
473 ; md5secret md5secret
474 ; dtmfmode dtmfmode
475 ; canreinvite canreinvite
476 ; nat nat
477 ; callgroup callgroup
478 ; pickupgroup pickupgroup
479 ; language language
480 ; allow allow
481 ; disallow disallow
482 ; insecure insecure
483 ; trustrpid trustrpid
484 ; progressinband progressinband
485 ; promiscredir promiscredir
486 ; useclientcode useclientcode
487 ; accountcode accountcode
488 ; setvar setvar
489 ; callerid callerid
490 ; amaflags amaflags
491 ; call-limit call-limit
492 ; allowoverlap allowoverlap
493 ; allowsubscribe allowsubscribe
494 ; allowtransfer allowtransfer
495 ; subscribecontext subscribecontext
496 ; videosupport videosupport
497 ; maxcallbitrate maxcallbitrate
498 ; rfc2833compensate mailbox
499 ; t38pt_usertpsource username
500 ; template
501 ; fromdomain
502 ; regexten
503 ; fromuser
504 ; host
505 ; port
506 ; qualify
507 ; defaultip
508 ; rtptimeout
509 ; rtpholdtimeout
510 ; sendrpid
511 ; outboundproxy
512 ; rfc2833compensate
513 ; t38pt_usertpsource
514 ; contactpermit ; Limit what a host may register as (a neat trick
515 ; contactdeny ; is to register at the same IP as a SIP provider,
516 ; ; then call oneself, and get redirected to that
517 ; ; same location).
519 ;[sip_proxy]
520 ; For incoming calls only. Example: FWD (Free World Dialup)
521 ; We match on IP address of the proxy for incoming calls
522 ; since we can not match on username (caller id)
523 ;type=peer
524 ;context=from-fwd
525 ;host=fwd.pulver.com
527 ;[sip_proxy-out]
528 ;type=peer ; we only want to call out, not be called
529 ;secret=guessit
530 ;username=yourusername ; Authentication user for outbound proxies
531 ;fromuser=yourusername ; Many SIP providers require this!
532 ;fromdomain=provider.sip.domain
533 ;host=box.provider.com
534 ;usereqphone=yes ; This provider requires ";user=phone" on URI
535 ;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer
536 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
537 ; Call-limits will not be enforced on real-time peers,
538 ; since they are not stored in-memory
539 ;port=80 ; The port number we want to connect to on the remote side
540 ; Also used as "defaultport" in combination with "defaultip" settings
542 ;------------------------------------------------------------------------------
543 ; Definitions of locally connected SIP devices
544 ;
545 ; type = user a device that authenticates to us by "from" field to place calls
546 ; type = peer a device we place calls to or that calls us and we match by host
547 ; type = friend two configurations (peer+user) in one
548 ;
549 ; For device names, we recommend using only a-z, numerics (0-9) and underscore
550 ;
551 ; For local phones, type=friend works most of the time
552 ;
553 ; If you have one-way audio, you probably have NAT problems.
554 ; If Asterisk is on a public IP, and the phone is inside of a NAT device
555 ; you will need to configure nat option for those phones.
556 ; Also, turn on qualify=yes to keep the nat session open
558 ;[grandstream1]
559 ;type=friend
560 ;context=from-sip ; Where to start in the dialplan when this phone calls
561 ;callerid=John Doe <1234> ; Full caller ID, to override the phones config
562 ; on incoming calls to Asterisk
563 ;host=192.168.0.23 ; we have a static but private IP address
564 ; No registration allowed
565 ;nat=no ; there is not NAT between phone and Asterisk
566 ;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
567 ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
568 ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
569 ; from the phone to asterisk
570 ; 1 for the explicit peer, 1 for the explicit user,
571 ; remember that a friend equals 1 peer and 1 user in
572 ; memory
573 ; This will affect your subscriptions as well.
574 ; There is no combined call counter for a "friend"
575 ; so there's currently no way in sip.conf to limit
576 ; to one inbound or outbound call per phone. Use
577 ; the group counters in the dial plan for that.
578 ;
579 ;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
580 ;disallow=all ; need to disallow=all before we can use allow=
581 ;allow=ulaw ; Note: In user sections the order of codecs
582 ; listed with allow= does NOT matter!
583 ;allow=alaw
584 ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
585 ;allow=g729 ; Pass-thru only unless g729 license obtained
586 ;callingpres=allowed_passed_screen ; Set caller ID presentation
587 ; See doc/callingpres.txt for more information
590 ;[xlite1]
591 ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
592 ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
593 ;type=friend
594 ;regexten=1234 ; When they register, create extension 1234
595 ;callerid="Jane Smith" <5678>
596 ;host=dynamic ; This device needs to register
597 ;nat=yes ; X-Lite is behind a NAT router
598 ;canreinvite=no ; Typically set to NO if behind NAT
599 ;disallow=all
600 ;allow=gsm ; GSM consumes far less bandwidth than ulaw
601 ;allow=ulaw
602 ;allow=alaw
603 ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
606 ;[snom]
607 ;type=friend ; Friends place calls and receive calls
608 ;context=from-sip ; Context for incoming calls from this user
609 ;secret=blah
610 ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
611 ;language=de ; Use German prompts for this user
612 ;host=dynamic ; This peer register with us
613 ;dtmfmode=inband ; Choices are inband, rfc2833, or info
614 ;defaultip=192.168.0.59 ; IP used until peer registers
615 ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
616 ;subscribemwi=yes ; Only send notifications if this phone
617 ; subscribes for mailbox notification
618 ;vmexten=voicemail ; dialplan extension to reach mailbox
619 ; sets the Message-Account in the MWI notify message
620 ; defaults to global vmexten which defaults to "asterisk"
621 ;disallow=all
622 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
625 ;[polycom]
626 ;type=friend ; Friends place calls and receive calls
627 ;context=from-sip ; Context for incoming calls from this user
628 ;secret=blahpoly
629 ;host=dynamic ; This peer register with us
630 ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
631 ;username=polly ; Username to use in INVITE until peer registers
632 ; Normally you do NOT need to set this parameter
633 ;disallow=all
634 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
635 ;progressinband=no ; Polycom phones don't work properly with "never"
638 ;[pingtel]
639 ;type=friend
640 ;secret=blah
641 ;host=dynamic
642 ;insecure=port ; Allow matching of peer by IP address without
643 ; matching port number
644 ;insecure=invite ; Do not require authentication of incoming INVITEs
645 ;insecure=port,invite ; (both)
646 ;qualify=1000 ; Consider it down if it's 1 second to reply
647 ; Helps with NAT session
648 ; qualify=yes uses default value
649 ;
650 ; Call group and Pickup group should be in the range from 0 to 63
651 ;
652 ;callgroup=1,3-4 ; We are in caller groups 1,3,4
653 ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
654 ;defaultip=192.168.0.60 ; IP address to use if peer has not registered
655 ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
656 ;permit=192.168.0.60/255.255.255.0
658 ;[cisco1]
659 ;type=friend
660 ;secret=blah
661 ;qualify=200 ; Qualify peer is no more than 200ms away
662 ;nat=yes ; This phone may be natted
663 ; Send SIP and RTP to the IP address that packet is
664 ; received from instead of trusting SIP headers
665 ;host=dynamic ; This device registers with us
666 ;canreinvite=no ; Asterisk by default tries to redirect the
667 ; RTP media stream (audio) to go directly from
668 ; the caller to the callee. Some devices do not
669 ; support this (especially if one of them is
670 ; behind a NAT).
671 ;defaultip=192.168.0.4 ; IP address to use until registration
672 ;username=goran ; Username to use when calling this device before registration
673 ; Normally you do NOT need to set this parameter
674 ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
676 ;[pre14-asterisk]
677 ;type=friend
678 ;secret=digium
679 ;host=dynamic
680 ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
681 ; You must have this turned on or DTMF reception will work improperly.
682 ;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
683 ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
684 ; external IP address of the remote device. If port forwarding is done at the client side
685 ; then UDPTL will flow to the remote device.