(changelog debian)
[asterisk-config-auf.git] / etc-asterisk / sip.conf
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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; reload chan_sip.so Reload configuration file
23; Active SIP peers will not be reconfigured
24;
25
26[general]
27context=default ; Default context for incoming calls
adb2451a 28allowguest=yes ; Allow or reject guest calls (default is yes)
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29allowoverlap=yes ; Disable overlap dialing support. (Default is yes)
30allowtransfer=yes ; Disable all transfers (unless enabled in peers or users)
31 ; Default is enabled
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32
33; **** GESTION DES DOMAINES SIP : voir /etc/asterisk/auf/sip-general.local
34; **** pour y indiquer le nom du domaine SIP géré localement
35
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36;realm=mydomain.tld ; Realm for digest authentication
37 ; defaults to "asterisk". If you set a system name in
38 ; asterisk.conf, it defaults to that system name
39 ; Realms MUST be globally unique according to RFC 3261
40 ; Set this to your host name or domain name
41bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
42 ; bindport is the local UDP port that Asterisk will listen on
43bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
44srvlookup=yes ; Enable DNS SRV lookups on outbound calls
45 ; Note: Asterisk only uses the first host
46 ; in SRV records
47 ; Disabling DNS SRV lookups disables the
48 ; ability to place SIP calls based on domain
49 ; names to some other SIP users on the Internet
50
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51; **** GESTION DES DOMAINES SIP : voir /etc/asterisk/auf/sip-general.local
52; **** pour y indiquer le nom du domaine SIP géré localement
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53;domain=mydomain.tld ; Set default domain for this host
54 ; If configured, Asterisk will only allow
55 ; INVITE and REFER to non-local domains
56 ; Use "sip show domains" to list local domains
57;pedantic=yes ; Enable checking of tags in headers,
58 ; international character conversions in URIs
59 ; and multiline formatted headers for strict
60 ; SIP compatibility (defaults to "no")
61
62; See doc/README.tos for a description of these parameters.
63tos_sip=cs3 ; Sets TOS for SIP packets.
64tos_audio=ef ; Sets TOS for RTP audio packets.
65tos_video=af41 ; Sets TOS for RTP video packets.
66
67;maxexpiry=3600 ; Maximum allowed time of incoming registrations
68 ; and subscriptions (seconds)
69;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
70;defaultexpiry=120 ; Default length of incoming/outgoing registration
71;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
72 ; Defaults to 100 ms
73;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
74;checkmwi=10 ; Default time between mailbox checks for peers
75;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
76 ; fully. Enable this option to not get error messages
77 ; when sending MWI to phones with this bug.
78;vmexten=voicemail ; dialplan extension to reach mailbox sets the
79 ; Message-Account in the MWI notify message
80 ; defaults to "asterisk"
81;disallow=all ; First disallow all codecs
82;allow=ulaw ; Allow codecs in order of preference
83;allow=ilbc ; see doc/rtp-packetization for framing options
84
85disallow=all
86allow=gsm
87allow=ulaw
88allow=h264
89allow=h263p
90allow=h263
91allow=h261
92
93;
94; This option specifies a preference for which music on hold class this channel
95; should listen to when put on hold if the music class has not been set on the
96; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
97; channel putting this one on hold did not suggest a music class.
98;
99; This option may be specified globally, or on a per-user or per-peer basis.
100;
101;mohinterpret=default
102;
103; This option specifies which music on hold class to suggest to the peer channel
104; when this channel places the peer on hold. It may be specified globally or on
105; a per-user or per-peer basis.
106;
107;mohsuggest=default
108;
109language=fr ; Default language setting for all users/peers
110 ; This may also be set for individual users/peers
111;relaxdtmf=yes ; Relax dtmf handling
112;trustrpid = no ; If Remote-Party-ID should be trusted
113;sendrpid = yes ; If Remote-Party-ID should be sent
114;progressinband=never ; If we should generate in-band ringing always
115 ; use 'never' to never use in-band signalling, even in cases
116 ; where some buggy devices might not render it
117 ; Valid values: yes, no, never Default: never
118;useragent=Asterisk PBX ; Allows you to change the user agent string
119;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
120 ; Note that promiscredir when redirects are made to the
121 ; local system will cause loops since Asterisk is incapable
122 ; of performing a "hairpin" call.
123;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
124 ; a valid phone number
125;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
126 ; Other options:
127 ; info : SIP INFO messages
128 ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
129 ; auto : Use rfc2833 if offered, inband otherwise
130dtmfmode=auto
131
132;compactheaders = yes ; send compact sip headers.
133;
134videosupport=yes ; Turn on support for SIP video. You need to turn this on
135 ; in the this section to get any video support at all.
136 ; You can turn it off on a per peer basis if the general
137 ; video support is enabled, but you can't enable it for
138 ; one peer only without enabling in the general section.
139;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
140 ; Videosupport and maxcallbitrate is settable
141 ; for peers and users as well
142;callevents=no ; generate manager events when sip ua
143 ; performs events (e.g. hold)
adb2451a 144alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
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145 ; for any reason, always reject with '401 Unauthorized'
146 ; instead of letting the requester know whether there was
147 ; a matching user or peer for their request
148
149;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
150 ; order instead of RFC3551 packing order (this is required
151 ; for Sipura and Grandstream ATAs, among others). This is
152 ; contrary to the RFC3551 specification, the peer _should_
153 ; be negotiating AAL2-G726-32 instead :-(
154
155;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
156 ; your localnet setting. Unless you have some sort of strange network
157 ; setup you will not need to enable this.
158
159;
160; If regcontext is specified, Asterisk will dynamically create and destroy a
161; NoOp priority 1 extension for a given peer who registers or unregisters with
162; us and have a "regexten=" configuration item.
163; Multiple contexts may be specified by separating them with '&'. The
164; actual extension is the 'regexten' parameter of the registering peer or its
165; name if 'regexten' is not provided. If more than one context is provided,
166; the context must be specified within regexten by appending the desired
167; context after '@'. More than one regexten may be supplied if they are
168; separated by '&'. Patterns may be used in regexten.
169;
170;regcontext=sipregistrations
171;
172;--------------------------- RTP timers ----------------------------------------------------
173; These timers are currently used for both audio and video streams. The RTP timeouts
174; are only applied to the audio channel.
175; The settings are settable in the global section as well as per device
176;
9b6412d4 177rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
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178 ; on the audio channel
179 ; when we're not on hold. This is to be able to hangup
180 ; a call in the case of a phone disappearing from the net,
181 ; like a powerloss or grandma tripping over a cable.
9b6412d4 182rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
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183 ; on the audio channel
184 ; when we're on hold (must be > rtptimeout)
9b6412d4 185rtpkeepalive=60 ; Send keepalives in the RTP stream to keep NAT open
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186 ; (default is off - zero)
187;--------------------------- SIP DEBUGGING ---------------------------------------------------
188;sipdebug = yes ; Turn on SIP debugging by default, from
189 ; the moment the channel loads this configuration
190;recordhistory=yes ; Record SIP history by default
191 ; (see sip history / sip no history)
192;dumphistory=yes ; Dump SIP history at end of SIP dialogue
193 ; SIP history is output to the DEBUG logging channel
194
195
196;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
197; You can subscribe to the status of extensions with a "hint" priority
198; (See extensions.conf.sample for examples)
199; chan_sip support two major formats for notifications: dialog-info and SIMPLE
200;
201; You will get more detailed reports (busy etc) if you have a call limit set
202; for a device. When the call limit is filled, we will indicate busy. Note that
203; you need at least 2 in order to be able to do attended transfers.
204;
205; For queues, you will need this level of detail in status reporting, regardless
206; if you use SIP subscriptions. Queues and manager use the same internal interface
207; for reading status information.
208;
209; Note: Subscriptions does not work if you have a realtime dialplan and use the
210; realtime switch.
211;
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212allowsubscribe = yes ; Disable support for subscriptions. (Default is yes)
213subscribecontext = AUF-local ; Set a specific context for SUBSCRIBE requests
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214 ; Useful to limit subscriptions to local extensions
215 ; Settable per peer/user also
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216notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
217notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
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218 ; Turning on notifyringing and notifyhold will add a lot
219 ; more database transactions if you are using realtime.
04c1f29b 220limitonpeers = yes ; Apply call limits on peers only. This will improve
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221 ; status notification when you are using type=friend
222 ; Inbound calls, that really apply to the user part
223 ; of a friend will now be added to and compared with
224 ; the peer limit instead of applying two call limits,
225 ; one for the peer and one for the user.
226 ; "sip show inuse" will only show active calls on
227 ; the peer side of a "type=friend" object if this
228 ; setting is turned on.
229
230;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
231;
232; This setting is available in the [general] section as well as in device configurations.
233; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
234; both parties have T38 support enabled in their Asterisk configuration
235; This has to be enabled in the general section for all devices to work. You can then
236; disable it on a per device basis.
237;
238; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.
239;
240; t38pt_udptl = yes ; Default false
241;
242;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
243; Asterisk can register as a SIP user agent to a SIP proxy (provider)
244; Format for the register statement is:
245; register => user[:secret[:authuser]]@host[:port][/extension]
246;
247; If no extension is given, the 's' extension is used. The extension needs to
248; be defined in extensions.conf to be able to accept calls from this SIP proxy
249; (provider).
250;
251; host is either a host name defined in DNS or the name of a section defined
252; below.
253;
254; Examples:
255;
256;register => 1234:password@mysipprovider.com
257;
258; This will pass incoming calls to the 's' extension
259;
260;
261;register => 2345:password@sip_proxy/1234
262;
263; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
264; connect to local extension 1234 in extensions.conf, default context,
265; unless you configure a [sip_proxy] section below, and configure a
266; context.
267; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
268; Tip 2: Use separate type=peer and type=user sections for SIP providers
269; (instead of type=friend) if you have calls in both directions
270
271;registertimeout=20 ; retry registration calls every 20 seconds (default)
272;registerattempts=10 ; Number of registration attempts before we give up
273 ; 0 = continue forever, hammering the other server
274 ; until it accepts the registration
275 ; Default is 0 tries, continue forever
276
277;----------------------------------------- NAT SUPPORT ------------------------
278; The externip, externhost and localnet settings are used if you use Asterisk
279; behind a NAT device to communicate with services on the outside.
280
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281; si votre serveur Asterisk est derrière un système DNAT, il faut indiquer
282; ici son adresse IP publique.
283
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284; ********** A FAIRE DANS /etc/asterisk/auf/sip-general.local
285; ********** sinon la modification sera effacée à la prochaine mise à jour...
286
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287;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP
288 ; messages if we're behind a NAT
289
290 ; The externip and localnet is used
291 ; when registering and communicating with other proxies
292 ; that we're registered with
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293
294; Si cette IP est dynamique, vous pouvez essayer d'utiliser externhost
295; et un système de type DynDNS
296
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297;externhost=foo.dyndns.net ; Alternatively you can specify an
298 ; external host, and Asterisk will
299 ; perform DNS queries periodically. Not
300 ; recommended for production
301 ; environments! Use externip instead
302;externrefresh=10 ; How often to refresh externhost if
303 ; used
304 ; You may add multiple local networks. A reasonable
305 ; set of defaults are:
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306
307; réseaux locaux avec lesquels il ne faut pas faire de NAT
308localnet=10.0.0.0/8
309localnet=172.16.0.0/12
310localnet=192.168.0.0/16
311localnet=169.254.0.0/16 ; ZeroConf
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312
313; The nat= setting is used when Asterisk is on a public IP, communicating with
314; devices hidden behind a NAT device (broadband router). If you have one-way
315; audio problems, you usually have problems with your NAT configuration or your
316; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
317; ports for incoming audio in rtp.conf
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318
319nat=no ; Global NAT settings (Affects all peers and users)
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320 ; yes = Always ignore info and assume NAT
321 ; no = Use NAT mode only according to RFC3581 (;rport)
322 ; never = Never attempt NAT mode or RFC3581 support
323 ; route = Assume NAT, don't send rport
324 ; (work around more UNIDEN bugs)
325
326;----------------------------------- MEDIA HANDLING --------------------------------
327; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
328; no reason for Asterisk to stay in the media path, the media will be redirected.
329; This does not really work with in the case where Asterisk is outside and have
330; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
331;
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332
333canreinvite=no ; Asterisk reste sur le chemin du flux audio
334
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335;canreinvite=yes ; Asterisk by default tries to redirect the
336 ; RTP media stream (audio) to go directly from
337 ; the caller to the callee. Some devices do not
338 ; support this (especially if one of them is behind a NAT).
339 ; The default setting is YES. If you have all clients
340 ; behind a NAT, or for some other reason wants Asterisk to
341 ; stay in the audio path, you may want to turn this off.
342
343 ; In Asterisk 1.4 this setting also affect direct RTP
344 ; at call setup (a new feature in 1.4 - setting up the
345 ; call directly between the endpoints instead of sending
346 ; a re-INVITE).
347
348;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
349 ; the call directly with media peer-2-peer without re-invites.
350 ; Will not work for video and cases where the callee sends
351 ; RTP payloads and fmtp headers in the 200 OK that does not match the
352 ; callers INVITE. This will also fail if canreinvite is enabled when
353 ; the device is actually behind NAT.
354
355;canreinvite=nonat ; An additional option is to allow media path redirection
356 ; (reinvite) but only when the peer where the media is being
357 ; sent is known to not be behind a NAT (as the RTP core can
358 ; determine it based on the apparent IP address the media
359 ; arrives from).
360
361;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
362 ; instead of INVITE. This can be combined with 'nonat', as
363 ; 'canreinvite=update,nonat'. It implies 'yes'.
364
365;----------------------------------------- REALTIME SUPPORT ------------------------
366; For additional information on ARA, the Asterisk Realtime Architecture,
367; please read realtime.txt and extconfig.txt in the /doc directory of the
368; source code.
369;
370;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
371 ; just like friends added from the config file only on a
372 ; as-needed basis? (yes|no)
373
374;rtsavesysname=yes ; Save systemname in realtime database at registration
375 ; Default= no
376
377;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
378 ; If set to yes, when a SIP UA registers successfully, the ip address,
379 ; the origination port, the registration period, and the username of
380 ; the UA will be set to database via realtime.
381 ; If not present, defaults to 'yes'.
382;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
383 ; as if it had just registered? (yes|no|<seconds>)
384 ; If set to yes, when the registration expires, the friend will
385 ; vanish from the configuration until requested again. If set
386 ; to an integer, friends expire within this number of seconds
387 ; instead of the registration interval.
388
389;ignoreregexpire=yes ; Enabling this setting has two functions:
390 ;
391 ; For non-realtime peers, when their registration expires, the
392 ; information will _not_ be removed from memory or the Asterisk database
393 ; if you attempt to place a call to the peer, the existing information
394 ; will be used in spite of it having expired
395 ;
396 ; For realtime peers, when the peer is retrieved from realtime storage,
397 ; the registration information will be used regardless of whether
398 ; it has expired or not; if it expires while the realtime peer
399 ; is still in memory (due to caching or other reasons), the
400 ; information will not be removed from realtime storage
401
402;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
403; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
404; domains, each of which can direct the call to a specific context if desired.
405; By default, all domains are accepted and sent to the default context or the
406; context associated with the user/peer placing the call.
407; Domains can be specified using:
408; domain=<domain>[,<context>]
409; Examples:
410; domain=myasterisk.dom
411; domain=customer.com,customer-context
412;
413; In addition, all the 'default' domains associated with a server should be
414; added if incoming request filtering is desired.
415; autodomain=yes
416;
417; To disallow requests for domains not serviced by this server:
418; allowexternaldomains=no
419
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420; **** GESTION DES DOMAINES SIP : voir /etc/asterisk/auf/sip-general.local
421; **** pour y indiquer le nom du domaine SIP géré localement
422
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423;domain=mydomain.tld,mydomain-incoming
424 ; Add domain and configure incoming context
425 ; for external calls to this domain
426;domain=1.2.3.4 ; Add IP address as local domain
427 ; You can have several "domain" settings
adb2451a 428allowexternaldomains=yes ; Disable INVITE and REFER to non-local domains
b2e905a6 429 ; Default is yes
adb2451a 430autodomain=no ; Turn this on to have Asterisk add local host
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431 ; name and local IP to domain list.
432
433; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
434 ; non-peers, use your primary domain "identity"
435 ; for From: headers instead of just your IP
436 ; address. This is to be polite and
437 ; it may be a mandatory requirement for some
438 ; destinations which do not have a prior
439 ; account relationship with your server.
440
441;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
442jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
443 ; SIP channel. Defaults to "no". An enabled jitterbuffer will
444 ; be used only if the sending side can create and the receiving
445 ; side can not accept jitter. The SIP channel can accept jitter,
446 ; thus a jitterbuffer on the receive SIP side will be used only
447 ; if it is forced and enabled.
448
449; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
450 ; channel. Defaults to "no".
451
452jbmaxsize = 500 ; Max length of the jitterbuffer in milliseconds.
453
454; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
455 ; resynchronized. Useful to improve the quality of the voice, with
456 ; big jumps in/broken timestamps, usually sent from exotic devices
457 ; and programs. Defaults to 1000.
458
459; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
460 ; channel. Two implementations are currently available - "fixed"
461 ; (with size always equals to jbmaxsize) and "adaptive" (with
462 ; variable size, actually the new jb of IAX2). Defaults to fixed.
463
464; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
465;-----------------------------------------------------------------------------------
466
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467#include "auf/sip-general.local"
468
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469[authentication]
470; Global credentials for outbound calls, i.e. when a proxy challenges your
471; Asterisk server for authentication. These credentials override
472; any credentials in peer/register definition if realm is matched.
473;
474; This way, Asterisk can authenticate for outbound calls to other
475; realms. We match realm on the proxy challenge and pick an set of
476; credentials from this list
477; Syntax:
478; auth = <user>:<secret>@<realm>
479; auth = <user>#<md5secret>@<realm>
480; Example:
481;auth=mark:topsecret@digium.com
482;
483; You may also add auth= statements to [peer] definitions
484; Peer auth= override all other authentication settings if we match on realm
485
486;------------------------------------------------------------------------------
487; Users and peers have different settings available. Friends have all settings,
488; since a friend is both a peer and a user
489;
490; User config options: Peer configuration:
491; -------------------- -------------------
492; context context
493; callingpres callingpres
494; permit permit
495; deny deny
496; secret secret
497; md5secret md5secret
498; dtmfmode dtmfmode
499; canreinvite canreinvite
500; nat nat
501; callgroup callgroup
502; pickupgroup pickupgroup
503; language language
504; allow allow
505; disallow disallow
506; insecure insecure
507; trustrpid trustrpid
508; progressinband progressinband
509; promiscredir promiscredir
510; useclientcode useclientcode
511; accountcode accountcode
512; setvar setvar
513; callerid callerid
514; amaflags amaflags
515; call-limit call-limit
516; allowoverlap allowoverlap
517; allowsubscribe allowsubscribe
518; allowtransfer allowtransfer
519; subscribecontext subscribecontext
520; videosupport videosupport
521; maxcallbitrate maxcallbitrate
522; rfc2833compensate mailbox
523; username
524; template
525; fromdomain
526; regexten
527; fromuser
528; host
529; port
530; qualify
531; defaultip
532; rtptimeout
533; rtpholdtimeout
534; sendrpid
535; outboundproxy
536; rfc2833compensate
537
538;[sip_proxy]
539; For incoming calls only. Example: FWD (Free World Dialup)
540; We match on IP address of the proxy for incoming calls
541; since we can not match on username (caller id)
542;type=peer
543;context=from-fwd
544;host=fwd.pulver.com
545
546;[sip_proxy-out]
547;type=peer ; we only want to call out, not be called
548;secret=guessit
549;username=yourusername ; Authentication user for outbound proxies
550;fromuser=yourusername ; Many SIP providers require this!
551;fromdomain=provider.sip.domain
552;host=box.provider.com
553;usereqphone=yes ; This provider requires ";user=phone" on URI
554;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer
555;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
556 ; Call-limits will not be enforced on real-time peers,
557 ; since they are not stored in-memory
558;port=80 ; The port number we want to connect to on the remote side
559 ; Also used as "defaultport" in combination with "defaultip" settings
560
561;------------------------------------------------------------------------------
562; Definitions of locally connected SIP devices
563;
564; type = user a device that authenticates to us by "from" field to place calls
565; type = peer a device we place calls to or that calls us and we match by host
566; type = friend two configurations (peer+user) in one
567;
568; For device names, we recommend using only a-z, numerics (0-9) and underscore
569;
570; For local phones, type=friend works most of the time
571;
572; If you have one-way audio, you probably have NAT problems.
573; If Asterisk is on a public IP, and the phone is inside of a NAT device
574; you will need to configure nat option for those phones.
575; Also, turn on qualify=yes to keep the nat session open
576
577
578;
579; Pont "Codian" CERN/CNRS/IN2P3/Inserm/INRA pour extension *341
580; http://vacs.in2p3.fr/
581;
582
583[ccmcu40-in2p3-fr]
584type=peer
585host=ccmcu40.in2p3.fr
586disallow=all
587allow=ulaw
588allow=h264
589allow=h263p
590allow=h263
591allow=h261
592dtmfmode=info
593
594;
fb03861a
TN
595; Sortie vers SIPBroker http://www.sipbroker.com
596;
597
598[sipbroker-out]
599type=peer
600fromuser=voip
601fromdomain=auf.org
602host=sipbroker.com
603port=5060
604canreinvite=yes
605
606;
607;
b2e905a6
TN
608;
609
610; Comptes pour postes clients locaux SIP
611#include "auf/sip.local"
612