premier commit (version 1.4-24)
[asterisk-config-auf.git] / etc-asterisk / sip.conf
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.
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)
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 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
20; sip debug Show all SIP messages
22; reload Reload configuration file
23; Active SIP peers will not be reconfigured
27context=default ; Default context for incoming calls
28;allowguest=no ; Allow or reject guest calls (default is yes)
29allowoverlap=yes ; Disable overlap dialing support. (Default is yes)
30allowtransfer=yes ; 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
37bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
38 ; bindport is the local UDP port that Asterisk will listen on
39bindaddr= ; IP address to bind to ( binds to all)
40srvlookup=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;domain=mydomain.tld ; Set default domain for this host
48 ; If configured, Asterisk will only allow
49 ; INVITE and REFER to non-local domains
50 ; Use "sip show domains" to list local domains
51;pedantic=yes ; Enable checking of tags in headers,
52 ; international character conversions in URIs
53 ; and multiline formatted headers for strict
54 ; SIP compatibility (defaults to "no")
56; See doc/README.tos for a description of these parameters.
57tos_sip=cs3 ; Sets TOS for SIP packets.
58tos_audio=ef ; Sets TOS for RTP audio packets.
59tos_video=af41 ; Sets TOS for RTP video packets.
61;maxexpiry=3600 ; Maximum allowed time of incoming registrations
62 ; and subscriptions (seconds)
63;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
64;defaultexpiry=120 ; Default length of incoming/outgoing registration
65;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
66 ; Defaults to 100 ms
67;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
68;checkmwi=10 ; Default time between mailbox checks for peers
69;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
70 ; fully. Enable this option to not get error messages
71 ; when sending MWI to phones with this bug.
72;vmexten=voicemail ; dialplan extension to reach mailbox sets the
73 ; Message-Account in the MWI notify message
74 ; defaults to "asterisk"
75;disallow=all ; First disallow all codecs
76;allow=ulaw ; Allow codecs in order of preference
77;allow=ilbc ; see doc/rtp-packetization for framing options
88; This option specifies a preference for which music on hold class this channel
89; should listen to when put on hold if the music class has not been set on the
90; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
91; channel putting this one on hold did not suggest a music class.
93; This option may be specified globally, or on a per-user or per-peer basis.
97; This option specifies which music on hold class to suggest to the peer channel
98; when this channel places the peer on hold. It may be specified globally or on
99; a per-user or per-peer basis.
103language=fr ; Default language setting for all users/peers
104 ; This may also be set for individual users/peers
105;relaxdtmf=yes ; Relax dtmf handling
106;trustrpid = no ; If Remote-Party-ID should be trusted
107;sendrpid = yes ; If Remote-Party-ID should be sent
108;progressinband=never ; If we should generate in-band ringing always
109 ; use 'never' to never use in-band signalling, even in cases
110 ; where some buggy devices might not render it
111 ; Valid values: yes, no, never Default: never
112;useragent=Asterisk PBX ; Allows you to change the user agent string
113;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
114 ; Note that promiscredir when redirects are made to the
115 ; local system will cause loops since Asterisk is incapable
116 ; of performing a "hairpin" call.
117;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
118 ; a valid phone number
119;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
120 ; Other options:
121 ; info : SIP INFO messages
122 ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
123 ; auto : Use rfc2833 if offered, inband otherwise
126;compactheaders = yes ; send compact sip headers.
128videosupport=yes ; Turn on support for SIP video. You need to turn this on
129 ; in the this section to get any video support at all.
130 ; You can turn it off on a per peer basis if the general
131 ; video support is enabled, but you can't enable it for
132 ; one peer only without enabling in the general section.
133;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
134 ; Videosupport and maxcallbitrate is settable
135 ; for peers and users as well
136;callevents=no ; generate manager events when sip ua
137 ; performs events (e.g. hold)
138;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
139 ; for any reason, always reject with '401 Unauthorized'
140 ; instead of letting the requester know whether there was
141 ; a matching user or peer for their request
143;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
144 ; order instead of RFC3551 packing order (this is required
145 ; for Sipura and Grandstream ATAs, among others). This is
146 ; contrary to the RFC3551 specification, the peer _should_
147 ; be negotiating AAL2-G726-32 instead :-(
149;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
150 ; your localnet setting. Unless you have some sort of strange network
151 ; setup you will not need to enable this.
154; If regcontext is specified, Asterisk will dynamically create and destroy a
155; NoOp priority 1 extension for a given peer who registers or unregisters with
156; us and have a "regexten=" configuration item.
157; Multiple contexts may be specified by separating them with '&'. The
158; actual extension is the 'regexten' parameter of the registering peer or its
159; name if 'regexten' is not provided. If more than one context is provided,
160; the context must be specified within regexten by appending the desired
161; context after '@'. More than one regexten may be supplied if they are
162; separated by '&'. Patterns may be used in regexten.
166;--------------------------- RTP timers ----------------------------------------------------
167; These timers are currently used for both audio and video streams. The RTP timeouts
168; are only applied to the audio channel.
169; The settings are settable in the global section as well as per device
171;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
172 ; on the audio channel
173 ; when we're not on hold. This is to be able to hangup
174 ; a call in the case of a phone disappearing from the net,
175 ; like a powerloss or grandma tripping over a cable.
176;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
177 ; on the audio channel
178 ; when we're on hold (must be > rtptimeout)
179;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
180 ; (default is off - zero)
181;--------------------------- SIP DEBUGGING ---------------------------------------------------
182;sipdebug = yes ; Turn on SIP debugging by default, from
183 ; the moment the channel loads this configuration
184;recordhistory=yes ; Record SIP history by default
185 ; (see sip history / sip no history)
186;dumphistory=yes ; Dump SIP history at end of SIP dialogue
187 ; SIP history is output to the DEBUG logging channel
190;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
191; You can subscribe to the status of extensions with a "hint" priority
192; (See extensions.conf.sample for examples)
193; chan_sip support two major formats for notifications: dialog-info and SIMPLE
195; You will get more detailed reports (busy etc) if you have a call limit set
196; for a device. When the call limit is filled, we will indicate busy. Note that
197; you need at least 2 in order to be able to do attended transfers.
199; For queues, you will need this level of detail in status reporting, regardless
200; if you use SIP subscriptions. Queues and manager use the same internal interface
201; for reading status information.
203; Note: Subscriptions does not work if you have a realtime dialplan and use the
204; realtime switch.
206;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
207;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
208 ; Useful to limit subscriptions to local extensions
209 ; Settable per peer/user also
210;notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
211;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
212 ; Turning on notifyringing and notifyhold will add a lot
213 ; more database transactions if you are using realtime.
214;limitonpeers = yes ; Apply call limits on peers only. This will improve
215 ; status notification when you are using type=friend
216 ; Inbound calls, that really apply to the user part
217 ; of a friend will now be added to and compared with
218 ; the peer limit instead of applying two call limits,
219 ; one for the peer and one for the user.
220 ; "sip show inuse" will only show active calls on
221 ; the peer side of a "type=friend" object if this
222 ; setting is turned on.
224;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
226; This setting is available in the [general] section as well as in device configurations.
227; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
228; both parties have T38 support enabled in their Asterisk configuration
229; This has to be enabled in the general section for all devices to work. You can then
230; disable it on a per device basis.
232; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.
234; t38pt_udptl = yes ; Default false
236;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
237; Asterisk can register as a SIP user agent to a SIP proxy (provider)
238; Format for the register statement is:
239; register => user[:secret[:authuser]]@host[:port][/extension]
241; If no extension is given, the 's' extension is used. The extension needs to
242; be defined in extensions.conf to be able to accept calls from this SIP proxy
243; (provider).
245; host is either a host name defined in DNS or the name of a section defined
246; below.
248; Examples:
250;register =>
252; This will pass incoming calls to the 's' extension
255;register => 2345:password@sip_proxy/1234
257; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
258; connect to local extension 1234 in extensions.conf, default context,
259; unless you configure a [sip_proxy] section below, and configure a
260; context.
261; Tip 1: Avoid assigning hostname to a sip.conf section like []
262; Tip 2: Use separate type=peer and type=user sections for SIP providers
263; (instead of type=friend) if you have calls in both directions
265;registertimeout=20 ; retry registration calls every 20 seconds (default)
266;registerattempts=10 ; Number of registration attempts before we give up
267 ; 0 = continue forever, hammering the other server
268 ; until it accepts the registration
269 ; Default is 0 tries, continue forever
271;----------------------------------------- NAT SUPPORT ------------------------
272; The externip, externhost and localnet settings are used if you use Asterisk
273; behind a NAT device to communicate with services on the outside.
275;externip = ; Address that we're going to put in outbound SIP
276 ; messages if we're behind a NAT
278 ; The externip and localnet is used
279 ; when registering and communicating with other proxies
280 ; that we're registered with
281; ; Alternatively you can specify an
282 ; external host, and Asterisk will
283 ; perform DNS queries periodically. Not
284 ; recommended for production
285 ; environments! Use externip instead
286;externrefresh=10 ; How often to refresh externhost if
287 ; used
288 ; You may add multiple local networks. A reasonable
289 ; set of defaults are:
290;localnet=; All RFC 1918 addresses are local networks
291;localnet= ; Also RFC1918
292;localnet= ; Another RFC1918 with CIDR notation
293;localnet= ;Zero conf local network
295; The nat= setting is used when Asterisk is on a public IP, communicating with
296; devices hidden behind a NAT device (broadband router). If you have one-way
297; audio problems, you usually have problems with your NAT configuration or your
298; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
299; ports for incoming audio in rtp.conf
301;nat=no ; Global NAT settings (Affects all peers and users)
302 ; yes = Always ignore info and assume NAT
303 ; no = Use NAT mode only according to RFC3581 (;rport)
304 ; never = Never attempt NAT mode or RFC3581 support
305 ; route = Assume NAT, don't send rport
306 ; (work around more UNIDEN bugs)
308;----------------------------------- MEDIA HANDLING --------------------------------
309; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
310; no reason for Asterisk to stay in the media path, the media will be redirected.
311; This does not really work with in the case where Asterisk is outside and have
312; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
314;canreinvite=yes ; Asterisk by default tries to redirect the
315 ; RTP media stream (audio) to go directly from
316 ; the caller to the callee. Some devices do not
317 ; support this (especially if one of them is behind a NAT).
318 ; The default setting is YES. If you have all clients
319 ; behind a NAT, or for some other reason wants Asterisk to
320 ; stay in the audio path, you may want to turn this off.
322 ; In Asterisk 1.4 this setting also affect direct RTP
323 ; at call setup (a new feature in 1.4 - setting up the
324 ; call directly between the endpoints instead of sending
325 ; a re-INVITE).
327;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
328 ; the call directly with media peer-2-peer without re-invites.
329 ; Will not work for video and cases where the callee sends
330 ; RTP payloads and fmtp headers in the 200 OK that does not match the
331 ; callers INVITE. This will also fail if canreinvite is enabled when
332 ; the device is actually behind NAT.
334;canreinvite=nonat ; An additional option is to allow media path redirection
335 ; (reinvite) but only when the peer where the media is being
336 ; sent is known to not be behind a NAT (as the RTP core can
337 ; determine it based on the apparent IP address the media
338 ; arrives from).
340;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
341 ; instead of INVITE. This can be combined with 'nonat', as
342 ; 'canreinvite=update,nonat'. It implies 'yes'.
344;----------------------------------------- REALTIME SUPPORT ------------------------
345; For additional information on ARA, the Asterisk Realtime Architecture,
346; please read realtime.txt and extconfig.txt in the /doc directory of the
347; source code.
349;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
350 ; just like friends added from the config file only on a
351 ; as-needed basis? (yes|no)
353;rtsavesysname=yes ; Save systemname in realtime database at registration
354 ; Default= no
356;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
357 ; If set to yes, when a SIP UA registers successfully, the ip address,
358 ; the origination port, the registration period, and the username of
359 ; the UA will be set to database via realtime.
360 ; If not present, defaults to 'yes'.
361;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
362 ; as if it had just registered? (yes|no|<seconds>)
363 ; If set to yes, when the registration expires, the friend will
364 ; vanish from the configuration until requested again. If set
365 ; to an integer, friends expire within this number of seconds
366 ; instead of the registration interval.
368;ignoreregexpire=yes ; Enabling this setting has two functions:
369 ;
370 ; For non-realtime peers, when their registration expires, the
371 ; information will _not_ be removed from memory or the Asterisk database
372 ; if you attempt to place a call to the peer, the existing information
373 ; will be used in spite of it having expired
374 ;
375 ; For realtime peers, when the peer is retrieved from realtime storage,
376 ; the registration information will be used regardless of whether
377 ; it has expired or not; if it expires while the realtime peer
378 ; is still in memory (due to caching or other reasons), the
379 ; information will not be removed from realtime storage
381;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
382; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
383; domains, each of which can direct the call to a specific context if desired.
384; By default, all domains are accepted and sent to the default context or the
385; context associated with the user/peer placing the call.
386; Domains can be specified using:
387; domain=<domain>[,<context>]
388; Examples:
389; domain=myasterisk.dom
392; In addition, all the 'default' domains associated with a server should be
393; added if incoming request filtering is desired.
394; autodomain=yes
396; To disallow requests for domains not serviced by this server:
397; allowexternaldomains=no
400 ; Add domain and configure incoming context
401 ; for external calls to this domain
402;domain= ; Add IP address as local domain
403 ; You can have several "domain" settings
404;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
405 ; Default is yes
406;autodomain=yes ; Turn this on to have Asterisk add local host
407 ; name and local IP to domain list.
409; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
410 ; non-peers, use your primary domain "identity"
411 ; for From: headers instead of just your IP
412 ; address. This is to be polite and
413 ; it may be a mandatory requirement for some
414 ; destinations which do not have a prior
415 ; account relationship with your server.
417;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
418jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
419 ; SIP channel. Defaults to "no". An enabled jitterbuffer will
420 ; be used only if the sending side can create and the receiving
421 ; side can not accept jitter. The SIP channel can accept jitter,
422 ; thus a jitterbuffer on the receive SIP side will be used only
423 ; if it is forced and enabled.
425; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
426 ; channel. Defaults to "no".
428jbmaxsize = 500 ; Max length of the jitterbuffer in milliseconds.
430; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
431 ; resynchronized. Useful to improve the quality of the voice, with
432 ; big jumps in/broken timestamps, usually sent from exotic devices
433 ; and programs. Defaults to 1000.
435; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
436 ; channel. Two implementations are currently available - "fixed"
437 ; (with size always equals to jbmaxsize) and "adaptive" (with
438 ; variable size, actually the new jb of IAX2). Defaults to fixed.
440; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
444; Global credentials for outbound calls, i.e. when a proxy challenges your
445; Asterisk server for authentication. These credentials override
446; any credentials in peer/register definition if realm is matched.
448; This way, Asterisk can authenticate for outbound calls to other
449; realms. We match realm on the proxy challenge and pick an set of
450; credentials from this list
451; Syntax:
452; auth = <user>:<secret>@<realm>
453; auth = <user>#<md5secret>@<realm>
454; Example:
457; You may also add auth= statements to [peer] definitions
458; Peer auth= override all other authentication settings if we match on realm
461; Users and peers have different settings available. Friends have all settings,
462; since a friend is both a peer and a user
464; User config options: Peer configuration:
465; -------------------- -------------------
466; context context
467; callingpres callingpres
468; permit permit
469; deny deny
470; secret secret
471; md5secret md5secret
472; dtmfmode dtmfmode
473; canreinvite canreinvite
474; nat nat
475; callgroup callgroup
476; pickupgroup pickupgroup
477; language language
478; allow allow
479; disallow disallow
480; insecure insecure
481; trustrpid trustrpid
482; progressinband progressinband
483; promiscredir promiscredir
484; useclientcode useclientcode
485; accountcode accountcode
486; setvar setvar
487; callerid callerid
488; amaflags amaflags
489; call-limit call-limit
490; allowoverlap allowoverlap
491; allowsubscribe allowsubscribe
492; allowtransfer allowtransfer
493; subscribecontext subscribecontext
494; videosupport videosupport
495; maxcallbitrate maxcallbitrate
496; rfc2833compensate mailbox
497; username
498; template
499; fromdomain
500; regexten
501; fromuser
502; host
503; port
504; qualify
505; defaultip
506; rtptimeout
507; rtpholdtimeout
508; sendrpid
509; outboundproxy
510; rfc2833compensate
513; For incoming calls only. Example: FWD (Free World Dialup)
514; We match on IP address of the proxy for incoming calls
515; since we can not match on username (caller id)
521;type=peer ; we only want to call out, not be called
523;username=yourusername ; Authentication user for outbound proxies
524;fromuser=yourusername ; Many SIP providers require this!
527;usereqphone=yes ; This provider requires ";user=phone" on URI
528;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer
529;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
530 ; Call-limits will not be enforced on real-time peers,
531 ; since they are not stored in-memory
532;port=80 ; The port number we want to connect to on the remote side
533 ; Also used as "defaultport" in combination with "defaultip" settings
536; Definitions of locally connected SIP devices
538; type = user a device that authenticates to us by "from" field to place calls
539; type = peer a device we place calls to or that calls us and we match by host
540; type = friend two configurations (peer+user) in one
542; For device names, we recommend using only a-z, numerics (0-9) and underscore
544; For local phones, type=friend works most of the time
546; If you have one-way audio, you probably have NAT problems.
547; If Asterisk is on a public IP, and the phone is inside of a NAT device
548; you will need to configure nat option for those phones.
549; Also, turn on qualify=yes to keep the nat session open
553; Pont "Codian" CERN/CNRS/IN2P3/Inserm/INRA pour extension *341
569; AUF
572; Comptes pour postes clients locaux SIP
573#include "auf/sip.local"