2 ; SIP Configuration example for Asterisk
4 ; Syntax for specifying a SIP device in extensions.conf is
5 ; SIP/devicename where devicename is defined in a section below.
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)
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
15 ; Useful CLI commands to check peers/users:
16 ; sip list peers Show all SIP peers (including friends)
17 ; sip list users Show all SIP users (including friends)
18 ; sip list registry Show status of hosts we register with
20 ; sip debug Show all SIP messages
22 ; module reload chan_sip.so Reload configuration file
23 ; Active SIP peers will not be reconfigured
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)
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 bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
39 srvlookup=yes ; Enable DNS SRV lookups on outbound calls
40 ; Note: Asterisk only uses the first host
42 ; Disabling DNS SRV lookups disables the
43 ; ability to place SIP calls based on domain
44 ; names to some other SIP users on the Internet
46 ;domain=mydomain.tld ; Set default domain for this host
47 ; If configured, Asterisk will only allow
48 ; INVITE and REFER to non-local domains
49 ; Use "sip show domains" to list local domains
50 ;pedantic=yes ; Enable checking of tags in headers,
51 ; international character conversions in URIs
52 ; and multiline formatted headers for strict
53 ; SIP compatibility (defaults to "no")
55 ; See doc/README.tos for a description of these parameters.
56 ;tos_sip=cs3 ; Sets TOS for SIP packets.
57 ;tos_audio=ef ; Sets TOS for RTP audio packets.
58 ;tos_video=af41 ; Sets TOS for RTP video packets.
60 ;maxexpiry=3600 ; Maximum allowed time of incoming registrations
61 ; and subscriptions (seconds)
62 ;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
63 ;defaultexpiry=120 ; Default length of incoming/outgoing registration
64 ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
66 ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
67 ;checkmwi=10 ; Default time between mailbox checks for peers
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.
80 ; This option may be specified globally, or on a per-user or per-peer basis.
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.
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 ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
94 ; when we're not on hold
95 ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity
96 ; when we're on hold (must be > rtptimeout)
97 ;trustrpid = no ; If Remote-Party-ID should be trusted
98 ;sendrpid = yes ; If Remote-Party-ID should be sent
99 ;progressinband=never ; If we should generate in-band ringing always
100 ; use 'never' to never use in-band signalling, even in cases
101 ; where some buggy devices might not render it
102 ; Valid values: yes, no, never Default: never
103 ;useragent=Asterisk PBX ; Allows you to change the user agent string
104 ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
105 ; Note that promiscredir when redirects are made to the
106 ; local system will cause loops since Asterisk is incapable
107 ; of performing a "hairpin" call.
108 ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
109 ; a valid phone number
110 ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
112 ; info : SIP INFO messages
113 ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
114 ; auto : Use rfc2833 if offered, inband otherwise
116 ;compactheaders = yes ; send compact sip headers.
118 ;videosupport=yes ; Turn on support for SIP video
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 :-(
136 ; If regcontext is specified, Asterisk will dynamically create and destroy a
137 ; NoOp priority 1 extension for a given peer who registers or unregisters with
138 ; us and have a "regexten=" configuration item.
139 ; Multiple contexts may be specified by separating them with '&'. The
140 ; actual extension is the 'regexten' parameter of the registering peer or its
141 ; name if 'regexten' is not provided. If more than one context is provided,
142 ; the context must be specified within regexten by appending the desired
143 ; context after '@'. More than one regexten may be supplied if they are
144 ; separated by '&'. Patterns may be used in regexten.
146 ;regcontext=sipregistrations
148 ;--------------------------- SIP DEBUGGING ---------------------------------------------------
149 ;sipdebug = yes ; Turn on SIP debugging by default, from
150 ; the moment the channel loads this configuration
151 ;recordhistory=yes ; Record SIP history by default
152 ; (see sip history / sip no history)
153 ;dumphistory=yes ; Dump SIP history at end of SIP dialogue
154 ; SIP history is output to the DEBUG logging channel
157 ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
158 ; You can subscribe to the status of extensions with a "hint" priority
159 ; (See extensions.conf.sample for examples)
160 ; chan_sip support two major formats for notifications: dialog-info and SIMPLE
161 ; Note: Subscriptions does not work if you have a realtime dialplan and use the
164 ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
165 ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
166 ; Useful to limit subscriptions to local extensions
167 ; Settable per peer/user also
168 ;notifyringing = yes ; Notify subscriptions on RINGING state
169 ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
171 ; This setting is available in the [general] section as well as in device configurations.
172 ; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
173 ; both parties have T38 support enabled in their Asterisk configuration (either general or
174 ; peer/user/friend sections)
176 ; t38pt_udptl = yes ; Default false
178 ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
179 ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
180 ; Format for the register statement is:
181 ; register => user[:secret[:authuser]]@host[:port][/extension]
183 ; If no extension is given, the 's' extension is used. The extension needs to
184 ; be defined in extensions.conf to be able to accept calls from this SIP proxy
187 ; host is either a host name defined in DNS or the name of a section defined
192 ;register => 1234:password@mysipprovider.com
194 ; This will pass incoming calls to the 's' extension
197 ;register => 2345:password@sip_proxy/1234
199 ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
200 ; connect to local extension 1234 in extensions.conf, default context,
201 ; unless you configure a [sip_proxy] section below, and configure a
203 ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
204 ; Tip 2: Use separate type=peer and type=user sections for SIP providers
205 ; (instead of type=friend) if you have calls in both directions
207 ;registertimeout=20 ; retry registration calls every 20 seconds (default)
208 ;registerattempts=10 ; Number of registration attempts before we give up
209 ; 0 = continue forever, hammering the other server
210 ; until it accepts the registration
211 ; Default is 0 tries, continue forever
213 ;----------------------------------------- NAT SUPPORT ------------------------
214 ; The externip, externhost and localnet settings are used if you use Asterisk
215 ; behind a NAT device to communicate with services on the outside.
217 ;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP
218 ; messages if we're behind a NAT
220 ; The externip and localnet is used
221 ; when registering and communicating with other proxies
222 ; that we're registered with
223 ;externhost=foo.dyndns.net ; Alternatively you can specify an
224 ; external host, and Asterisk will
225 ; perform DNS queries periodically. Not
226 ; recommended for production
227 ; environments! Use externip instead
228 ;externrefresh=10 ; How often to refresh externhost if
230 ; You may add multiple local networks. A reasonable
231 ; set of defaults are:
232 ;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
233 ;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
234 ;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
235 ;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
237 ; The nat= setting is used when Asterisk is on a public IP, communicating with
238 ; devices hidden behind a NAT device (broadband router). If you have one-way
239 ; audio problems, you usually have problems with your NAT configuration or your
240 ; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
241 ; ports for incoming audio in rtp.conf
243 ;nat=no ; Global NAT settings (Affects all peers and users)
244 ; yes = Always ignore info and assume NAT
245 ; no = Use NAT mode only according to RFC3581
246 ; never = Never attempt NAT mode or RFC3581 support
247 ; route = Assume NAT, don't send rport
248 ; (work around more UNIDEN bugs)
250 ;canreinvite=yes ; Asterisk by default tries to redirect the
251 ; RTP media stream (audio) to go directly from
252 ; the caller to the callee. Some devices do not
253 ; support this (especially if one of them is behind a NAT).
254 ; The default setting is YES. If you have all clients
255 ; behind a NAT, or for some other reason wants Asterisk to
256 ; stay in the audio path, you may want to turn this off.
258 ;canreinvite=nonat ; An additional option is to allow media path redirection
259 ; (reinvite) but only when the peer where the media is being
260 ; sent is known to not be behind a NAT (as the RTP core can
261 ; determine it based on the apparent IP address the media
264 ;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
265 ; instead of INVITE. This can be combined with 'nonat', as
266 ; 'canreinvite=update,nonat'. It implies 'yes'.
268 ;----------------------------------------- REALTIME SUPPORT ------------------------
269 ; For additional information on ARA, the Asterisk Realtime Architecture,
270 ; please read realtime.txt and extconfig.txt in the /doc directory of the
273 ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
274 ; just like friends added from the config file only on a
275 ; as-needed basis? (yes|no)
277 ;rtsavesysname=yes ; Save systemname in realtime database at registration
280 ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
281 ; If set to yes, when a SIP UA registers successfully, the ip address,
282 ; the origination port, the registration period, and the username of
283 ; the UA will be set to database via realtime.
284 ; If not present, defaults to 'yes'.
285 ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
286 ; as if it had just registered? (yes|no|<seconds>)
287 ; If set to yes, when the registration expires, the friend will
288 ; vanish from the configuration until requested again. If set
289 ; to an integer, friends expire within this number of seconds
290 ; instead of the registration interval.
292 ;ignoreregexpire=yes ; Enabling this setting has two functions:
294 ; For non-realtime peers, when their registration expires, the
295 ; information will _not_ be removed from memory or the Asterisk database
296 ; if you attempt to place a call to the peer, the existing information
297 ; will be used in spite of it having expired
299 ; For realtime peers, when the peer is retrieved from realtime storage,
300 ; the registration information will be used regardless of whether
301 ; it has expired or not; if it expires while the realtime peer
302 ; is still in memory (due to caching or other reasons), the
303 ; information will not be removed from realtime storage
305 ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
306 ; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
307 ; domains, each of which can direct the call to a specific context if desired.
308 ; By default, all domains are accepted and sent to the default context or the
309 ; context associated with the user/peer placing the call.
310 ; Domains can be specified using:
311 ; domain=<domain>[,<context>]
313 ; domain=myasterisk.dom
314 ; domain=customer.com,customer-context
316 ; In addition, all the 'default' domains associated with a server should be
317 ; added if incoming request filtering is desired.
320 ; To disallow requests for domains not serviced by this server:
321 ; allowexternaldomains=no
323 ;domain=mydomain.tld,mydomain-incoming
324 ; Add domain and configure incoming context
325 ; for external calls to this domain
326 ;domain=1.2.3.4 ; Add IP address as local domain
327 ; You can have several "domain" settings
328 ;allowexternalinvites=no ; Disable INVITE and REFER to non-local domains
330 ;autodomain=yes ; Turn this on to have Asterisk add local host
331 ; name and local IP to domain list.
333 ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
334 ; non-peers, use your primary domain "identity"
335 ; for From: headers instead of just your IP
336 ; address. This is to be polite and
337 ; it may be a mandatory requirement for some
338 ; destinations which do not have a prior
339 ; account relationship with your server.
341 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
342 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
343 ; SIP channel. Defaults to "no". An enabled jitterbuffer will
344 ; be used only if the sending side can create and the receiving
345 ; side can not accept jitter. The SIP channel can accept jitter,
346 ; thus a jitterbuffer on the receive SIP side will be used only
347 ; if it is forced and enabled.
349 ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
350 ; channel. Defaults to "no".
352 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
354 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
355 ; resynchronized. Useful to improve the quality of the voice, with
356 ; big jumps in/broken timestamps, usually sent from exotic devices
357 ; and programs. Defaults to 1000.
359 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
360 ; channel. Two implementations are currently available - "fixed"
361 ; (with size always equals to jbmaxsize) and "adaptive" (with
362 ; variable size, actually the new jb of IAX2). Defaults to fixed.
364 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
365 ;-----------------------------------------------------------------------------------
368 ; Global credentials for outbound calls, i.e. when a proxy challenges your
369 ; Asterisk server for authentication. These credentials override
370 ; any credentials in peer/register definition if realm is matched.
372 ; This way, Asterisk can authenticate for outbound calls to other
373 ; realms. We match realm on the proxy challenge and pick an set of
374 ; credentials from this list
376 ; auth = <user>:<secret>@<realm>
377 ; auth = <user>#<md5secret>@<realm>
379 ;auth=mark:topsecret@digium.com
381 ; You may also add auth= statements to [peer] definitions
382 ; Peer auth= override all other authentication settings if we match on realm
384 ;------------------------------------------------------------------------------
385 ; Users and peers have different settings available. Friends have all settings,
386 ; since a friend is both a peer and a user
388 ; User config options: Peer configuration:
389 ; -------------------- -------------------
391 ; callingpres callingpres
395 ; md5secret md5secret
397 ; canreinvite canreinvite
399 ; callgroup callgroup
400 ; pickupgroup pickupgroup
405 ; trustrpid trustrpid
406 ; progressinband progressinband
407 ; promiscredir promiscredir
408 ; useclientcode useclientcode
409 ; accountcode accountcode
413 ; call-limit call-limit
414 ; allowoverlap allowoverlap
415 ; allowsubscribe allowsubscribe
416 ; allowtransfer allowtransfer
417 ; subscribecontext subscribecontext
418 ; videosupport videosupport
419 ; maxcallbitrate maxcallbitrate
420 ; rfc2833compensate mailbox
437 ; For incoming calls only. Example: FWD (Free World Dialup)
438 ; We match on IP address of the proxy for incoming calls
439 ; since we can not match on username (caller id)
445 ;type=peer ; we only want to call out, not be called
447 ;username=yourusername ; Authentication user for outbound proxies
448 ;fromuser=yourusername ; Many SIP providers require this!
449 ;fromdomain=provider.sip.domain
450 ;host=box.provider.com
451 ;usereqphone=yes ; This provider requires ";user=phone" on URI
452 ;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer
453 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
454 ; Call-limits will not be enforced on real-time peers,
455 ; since they are not stored in-memory
457 ;------------------------------------------------------------------------------
458 ; Definitions of locally connected SIP devices
460 ; type = user a device that authenticates to us by "from" field to place calls
461 ; type = peer a device we place calls to or that calls us and we match by host
462 ; type = friend two configurations (peer+user) in one
464 ; For device names, we recommend using only a-z, numerics (0-9) and underscore
466 ; For local phones, type=friend works most of the time
468 ; If you have one-way audio, you probably have NAT problems.
469 ; If Asterisk is on a public IP, and the phone is inside of a NAT device
470 ; you will need to configure nat option for those phones.
471 ; Also, turn on qualify=yes to keep the nat session open
475 ;context=from-sip ; Where to start in the dialplan when this phone calls
476 ;callerid=John Doe <1234> ; Full caller ID, to override the phones config
477 ; on incoming calls to Asterisk
478 ;host=192.168.0.23 ; we have a static but private IP address
479 ; No registration allowed
480 ;nat=no ; there is not NAT between phone and Asterisk
481 ;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
482 ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
483 ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
484 ; from the phone to asterisk
485 ; 1 for the explicit peer, 1 for the explicit user,
486 ; remember that a friend equals 1 peer and 1 user in
488 ; This will affect your subscriptions as well.
489 ; There is no combined call counter for a "friend"
490 ; so there's currently no way in sip.conf to limit
491 ; to one inbound or outbound call per phone. Use
492 ; the group counters in the dial plan for that.
494 ;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
495 ;disallow=all ; need to disallow=all before we can use allow=
496 ;allow=ulaw ; Note: In user sections the order of codecs
497 ; listed with allow= does NOT matter!
499 ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
500 ;allow=g729 ; Pass-thru only unless g729 license obtained
501 ;callingpres=allowed_passed_screen ; Set caller ID presentation
502 ; See README.callingpres for more information
506 ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
507 ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
509 ;regexten=1234 ; When they register, create extension 1234
510 ;callerid="Jane Smith" <5678>
511 ;host=dynamic ; This device needs to register
512 ;nat=yes ; X-Lite is behind a NAT router
513 ;canreinvite=no ; Typically set to NO if behind NAT
515 ;allow=gsm ; GSM consumes far less bandwidth than ulaw
518 ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
522 ;type=friend ; Friends place calls and receive calls
523 ;context=from-sip ; Context for incoming calls from this user
525 ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
526 ;language=de ; Use German prompts for this user
527 ;host=dynamic ; This peer register with us
528 ;dtmfmode=inband ; Choices are inband, rfc2833, or info
529 ;defaultip=192.168.0.59 ; IP used until peer registers
530 ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
531 ;subscribemwi=yes ; Only send notifications if this phone
532 ; subscribes for mailbox notification
533 ;vmexten=voicemail ; dialplan extension to reach mailbox
534 ; sets the Message-Account in the MWI notify message
535 ; defaults to global vmexten which defaults to "asterisk"
537 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
541 ;type=friend ; Friends place calls and receive calls
542 ;context=from-sip ; Context for incoming calls from this user
544 ;host=dynamic ; This peer register with us
545 ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
546 ;username=polly ; Username to use in INVITE until peer registers
547 ; Normally you do NOT need to set this parameter
549 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
550 ;progressinband=no ; Polycom phones don't work properly with "never"
557 ;insecure=port ; Allow matching of peer by IP address without
558 ; matching port number
559 ;insecure=invite ; Do not require authentication of incoming INVITEs
560 ;insecure=port,invite ; (both)
561 ;qualify=1000 ; Consider it down if it's 1 second to reply
562 ; Helps with NAT session
563 ; qualify=yes uses default value
565 ; Call group and Pickup group should be in the range from 0 to 63
567 ;callgroup=1,3-4 ; We are in caller groups 1,3,4
568 ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
569 ;defaultip=192.168.0.60 ; IP address to use if peer has not registered
570 ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
571 ;permit=192.168.0.60/255.255.255.0
576 ;qualify=200 ; Qualify peer is no more than 200ms away
577 ;nat=yes ; This phone may be natted
578 ; Send SIP and RTP to the IP address that packet is
579 ; received from instead of trusting SIP headers
580 ;host=dynamic ; This device registers with us
581 ;canreinvite=no ; Asterisk by default tries to redirect the
582 ; RTP media stream (audio) to go directly from
583 ; the caller to the callee. Some devices do not
584 ; support this (especially if one of them is
586 ;defaultip=192.168.0.4 ; IP address to use until registration
587 ;username=goran ; Username to use when calling this device before registration
588 ; Normally you do NOT need to set this parameter
589 ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
595 ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
596 ; You must have this turned on or DTMF reception will work improperly.
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