Mise à jour après premières vagues de gros tests
[asterisk-config-auf.git] / etc-asterisk / sip.conf
2; SIP Configuration example for Asterisk
4; Note: Please read the security documentation for Asterisk in order to
5; understand the risks of installing Asterisk with the sample
6; configuration. If your Asterisk is installed on a public
7; IP address connected to the Internet, you will want to learn
8; about the various security settings BEFORE you start
9; Asterisk.
11; Especially note the following settings:
12; - allowguest (default enabled)
13; - permit/deny - IP address filters
14; - contactpermit/contactdeny - IP address filters for registrations
15; - context - Which set of services you offer various users
17; SIP dial strings
19; In the dialplan (extensions.conf) you can use several
20; syntaxes for dialing SIP devices.
21; SIP/devicename
22; SIP/username@domain (SIP uri)
23; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
24; SIP/devicename/extension
25; SIP/devicename/extension/IPorHost
26; SIP/username@domain//IPorHost
29; Devicename
30; devicename is defined as a peer in a section below.
32; username@domain
33; Call any SIP user on the Internet
34; (Don't forget to enable DNS SRV records if you want to use this)
36; devicename/extension
37; If you define a SIP proxy as a peer below, you may call
38; SIP/proxyhostname/user or SIP/user@proxyhostname
39; where the proxyhostname is defined in a section below
40; This syntax also works with ATA's with FXO ports
42; SIP/username[:password[:md5secret[:authname]]]@host[:port]
43; This form allows you to specify password or md5secret and authname
44; without altering any authentication data in config.
45; Examples:
47; SIP/*98@mysipproxy
48; SIP/sales:topsecret::account02@domain.com:5062
49; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@
51; IPorHost
52; The next server for this call regardless of domain/peer
54; All of these dial strings specify the SIP request URI.
55; In addition, you can specify a specific To: header by adding an
56; exclamation mark after the dial string, like
58; SIP/sales@mysipproxy!sales@edvina.net
60; A new feature for 1.8 allows one to specify a host or IP address to use
61; when routing the call. This is typically used in tandem with func_srv if
62; multiple methods of reaching the same domain exist. The host or IP address
63; is specified after the third slash in the dialstring. Examples:
65; SIP/devicename/extension/IPorHost
66; SIP/username@domain//IPorHost
68; CLI Commands
69; -------------------------------------------------------------
70; Useful CLI commands to check peers/users:
71; sip show peers Show all SIP peers (including friends)
72; sip show registry Show status of hosts we register with
74; sip set debug on Show all SIP messages
76; sip reload Reload configuration file
77; sip show settings Show the current channel configuration
79;------- Naming devices ------------------------------------------------------
81; When naming devices, make sure you understand how Asterisk matches calls
82; that come in.
83; 1. Asterisk checks the SIP From: address username and matches against
84; names of devices with type=user
85; The name is the text between square brackets [name]
86; 2. Asterisk checks the From: addres and matches the list of devices
87; with a type=peer
88; 3. Asterisk checks the IP address (and port number) that the INVITE
89; was sent from and matches against any devices with type=peer
91; Don't mix extensions with the names of the devices. Devices need a unique
92; name. The device name is *not* used as phone numbers. Phone numbers are
93; anything you declare as an extension in the dialplan (extensions.conf).
95; When setting up trunks, make sure there's no risk that any From: username
96; (caller ID) will match any of your device names, because then Asterisk
97; might match the wrong device.
99; Note: The parameter "username" is not the username and in most cases is
100; not needed at all. Check below. In later releases, it's renamed
101; to "defaultuser" which is a better name, since it is used in
102; combination with the "defaultip" setting.
105; ** Old configuration options **
106; The "call-limit" configuation option is considered old is replaced
107; by new functionality. To enable callcounters, you use the new
108; "callcounter" setting (for extension states in queue and subscriptions)
109; You are encouraged to use the dialplan groupcount functionality
110; to enforce call limits instead of using this channel-specific method.
111; You can still set limits per device in sip.conf or in a database by using
112; "setvar" to set variables that can be used in the dialplan for various limits.
115context=default ; Default context for incoming calls
a52025b1 116allowguest=no ; Allow or reject guest calls (default is yes)
117 ; If your Asterisk is connected to the Internet
118 ; and you have allowguest=yes
119 ; you want to check which services you offer everyone
120 ; out there, by enabling them in the default context (see below).
121;match_auth_username=yes ; if available, match user entry using the
122 ; 'username' field from the authentication line
123 ; instead of the From: field.
124;allowoverlap=no ; Disable overlap dialing support. (Default is yes)
125allowoverlap=yes ; Enable RFC3578 overlap dialing support.
126 ; Can use the Incomplete application to collect the
127 ; needed digits from an ambiguous dialplan match.
128;allowoverlap=dtmf ; Enable overlap dialing support using DTMF delivery
129 ; methods (inband, RFC2833, SIP INFO) in the early
130 ; media phase. Uses the Incomplete application to
131 ; collect the needed digits.
a52025b1 132allowtransfer=yes ; Disable all transfers (unless enabled in peers or users)
133 ; Default is enabled. The Dial() options 't' and 'T' are not
134 ; related as to whether SIP transfers are allowed or not.
136; **** GESTION DES DOMAINES SIP : voir /etc/asterisk/auf/sip-general.local
137; **** pour y indiquer le nom du domaine SIP géré localement
139;realm=mydomain.tld ; Realm for digest authentication
140 ; defaults to "asterisk". If you set a system name in
141 ; asterisk.conf, it defaults to that system name
142 ; Realms MUST be globally unique according to RFC 3261
143 ; Set this to your host name or domain name
144;domainsasrealm=no ; Use domains list as realms
145 ; You can serve multiple Realms specifying several
146 ; 'domain=...' directives (see below).
147 ; In this case Realm will be based on request 'From'/'To' header
148 ; and should match one of domain names.
149 ; Otherwise default 'realm=...' will be used.
151; With the current situation, you can do one of four things:
152; a) Listen on a specific IPv4 address. Example: bindaddr=
153; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1
154; c) Listen on the IPv4 wildcard. Example: bindaddr=
155; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=::
156; (You can choose independently for UDP, TCP, and TLS, by specifying different values for
157; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".)
158; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat.
159; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)
161; Using bindaddr will only enable UDP support in order to be backwards compatible with those systems
162; that were upgraded prior to TCP support. Use udpbindaddr and tcpbindaddr to bind to UDP and TCP
163; independently.
165; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061
166; for TLS).
167; IPv4 example: bindaddr=
168; IPv6 example: bindaddr=[::]:5062
170; The address family of the bound UDP address is used to determine how Asterisk performs
171; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only
172; AAAA records are considered. In case d), both A and AAAA records are considered. Note,
173; however, that Asterisk ignores all records except the first one. In case d), when both A
174; and AAAA records are available, either an A or AAAA record will be first, and which one
175; depends on the operating system. On systems using glibc, AAAA records are given
176; priority.
178udpbindaddr= ; IP address to bind UDP listen socket to ( binds to all)
179 ; Optionally add a port number, (default is port 5060)
181; When a dialog is started with another SIP endpoint, the other endpoint
182; should include an Allow header telling us what SIP methods the endpoint
183; implements. However, some endpoints either do not include an Allow header
184; or lie about what methods they implement. In the former case, Asterisk
185; makes the assumption that the endpoint supports all known SIP methods.
186; If you know that your SIP endpoint does not provide support for a specific
187; method, then you may provide a comma-separated list of methods that your
188; endpoint does not implement in the disallowed_methods option. Note that
189; if your endpoint is truthful with its Allow header, then there is no need
190; to set this option. This option may be set in the general section or may
191; be set per endpoint. If this option is set both in the general section and
192; in a peer section, then the peer setting completely overrides the general
193; setting (i.e. the result is *not* the union of the two options).
195; Note also that while Asterisk currently will parse an Allow header to learn
196; what methods an endpoint supports, the only actual use for this currently
197; is for determining if Asterisk may send connected line UPDATE requests and
198; MESSAGE requests. Its use may be expanded in the future.
200; disallowed_methods = UPDATE
203; Note that the TCP and TLS support for chan_sip is currently considered
204; experimental. Since it is new, all of the related configuration options are
205; subject to change in any release. If they are changed, the changes will
206; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
208tcpenable=no ; Enable server for incoming TCP connections (default is no)
209tcpbindaddr= ; IP address for TCP server to bind to ( binds to all interfaces)
210 ; Optionally add a port number, (default is port 5060)
212;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
213;tlsbindaddr= ; IP address for TLS server to bind to ( binds to all interfaces)
214 ; Optionally add a port number, (default is port 5061)
215 ; Remember that the IP address must match the common name (hostname) in the
216 ; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
217 ; For details how to construct a certificate for SIP see
218 ; http://tools.ietf.org/html/draft-ietf-sip-domain-certs
220;tcpauthtimeout = 30 ; tcpauthtimeout specifies the maximum number
221 ; of seconds a client has to authenticate. If
222 ; the client does not authenticate beofre this
223 ; timeout expires, the client will be
224 ; disconnected. (default: 30 seconds)
226;tcpauthlimit = 100 ; tcpauthlimit specifies the maximum number of
227 ; unauthenticated sessions that will be allowed
228 ; to connect at any given time. (default: 100)
230transport=udp ; Set the default transports. The order determines the primary default transport.
231 ; If tcpenable=no and the transport set is tcp, we will fallback to UDP.
232 ; ******* pour aussi activer le TCP : transport = udp,tcp
233 ; ******* ne pas oublier de mettre tcpenable=yes, voir plus haut
235srvlookup=yes ; Enable DNS SRV lookups on outbound calls
236 ; Note: Asterisk only uses the first host
237 ; in SRV records
238 ; Disabling DNS SRV lookups disables the
239 ; ability to place SIP calls based on domain
240 ; names to some other SIP users on the Internet
241 ; Specifying a port in a SIP peer definition or
242 ; when dialing outbound calls will supress SRV
243 ; lookups for that peer or call.
245;pedantic=yes ; Enable checking of tags in headers,
246 ; international character conversions in URIs
247 ; and multiline formatted headers for strict
248 ; SIP compatibility (defaults to "yes")
250; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
251tos_sip=cs3 ; Sets TOS for SIP packets.
252tos_audio=ef ; Sets TOS for RTP audio packets.
253tos_video=af41 ; Sets TOS for RTP video packets.
254tos_text=af41 ; Sets TOS for RTP text packets.
3802e567 255
256cos_sip=3 ; Sets 802.1p priority for SIP packets.
257cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
258cos_video=4 ; Sets 802.1p priority for RTP video packets.
259cos_text=3 ; Sets 802.1p priority for RTP text packets.
261;maxexpiry=3600 ; Maximum allowed time of incoming registrations
262 ; and subscriptions (seconds)
263;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
264;defaultexpiry=120 ; Default length of incoming/outgoing registration
265;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
266;maxforwards=70 ; Setting for the SIP Max-Forwards: header (loop prevention)
267 ; Default value is 70
268;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds
269 ; and reported in milliseconds with sip show settings.
270 ; Set to low value if you use low timeout for NAT of UDP sessions
271 ; Default: 60
272;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
273 ; Default: 100
274;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
275 ; Default: 1
276;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
277;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
278 ; fully. Enable this option to not get error messages
279 ; when sending MWI to phones with this bug.
280;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in
281 ; the From: header as the "name" portion. Also fill the
282 ; "user" portion of the URI in the From: header with this
283 ; value if no fromuser is set
284 ; Default: empty
285;vmexten=voicemail ; dialplan extension to reach mailbox sets the
286 ; Message-Account in the MWI notify message
287 ; defaults to "asterisk"
289; Codec negotiation
291; When Asterisk is receiving a call, the codec will initially be set to the
292; first codec in the allowed codecs defined for the user receiving the call
293; that the caller also indicates that it supports. But, after the caller
294; starts sending RTP, Asterisk will switch to using whatever codec the caller
295; is sending.
297; When Asterisk is placing a call, the codec used will be the first codec in
298; the allowed codecs that the callee indicates that it supports. Asterisk will
299; *not* switch to whatever codec the callee is sending.
301;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
302 ; rather than advertising all joint codec capabilities. This
303 ; limits the other side's codec choice to exactly what we prefer.
305disallow=all ; First disallow all codecs
307allow=ulaw ; Allow codecs in order of preference
308;allow=ilbc ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
309 ; for framing options
315; This option specifies a preference for which music on hold class this channel
316; should listen to when put on hold if the music class has not been set on the
317; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
318; channel putting this one on hold did not suggest a music class.
320; This option may be specified globally, or on a per-user or per-peer basis.
324; This option specifies which music on hold class to suggest to the peer channel
325; when this channel places the peer on hold. It may be specified globally or on
326; a per-user or per-peer basis.
330;parkinglot=plaza ; Sets the default parking lot for call parking
331 ; This may also be set for individual users/peers
332 ; Parkinglots are configured in features.conf
a52025b1 333language=fr ; Default language setting for all users/peers
334 ; This may also be set for individual users/peers
335;relaxdtmf=yes ; Relax dtmf handling
336;trustrpid = no ; If Remote-Party-ID should be trusted
337;sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no)
338;sendrpid = rpid ; Use the "Remote-Party-ID" header
339 ; to send the identity of the remote party
340 ; This is identical to sendrpid=yes
341;sendrpid = pai ; Use the "P-Asserted-Identity" header
342 ; to send the identity of the remote party
343;rpid_update = no ; In certain cases, the only method by which a connected line
344 ; change may be immediately transmitted is with a SIP UPDATE request.
345 ; If communicating with another Asterisk server, and you wish to be able
346 ; transmit such UPDATE messages to it, then you must enable this option.
347 ; Otherwise, we will have to wait until we can send a reinvite to
348 ; transmit the information.
349;prematuremedia=no ; Some ISDN links send empty media frames before
350 ; the call is in ringing or progress state. The SIP
351 ; channel will then send 183 indicating early media
352 ; which will be empty - thus users get no ring signal.
353 ; Setting this to "yes" will stop any media before we have
354 ; call progress (meaning the SIP channel will not send 183 Session
355 ; Progress for early media). Default is "yes". Also make sure that
356 ; the SIP peer is configured with progressinband=never.
357 ;
358 ; In order for "noanswer" applications to work, you need to run
359 ; the progress() application in the priority before the app.
361;progressinband=never ; If we should generate in-band ringing always
362 ; use 'never' to never use in-band signalling, even in cases
363 ; where some buggy devices might not render it
364 ; Valid values: yes, no, never Default: never
365;useragent=Asterisk PBX ; Allows you to change the user agent string
366 ; The default user agent string also contains the Asterisk
367 ; version. If you don't want to expose this, change the
368 ; useragent string.
369;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
370 ; Note that promiscredir when redirects are made to the
371 ; local system will cause loops since Asterisk is incapable
372 ; of performing a "hairpin" call.
373;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
374 ; a valid phone number
a52025b1 375dtmfmode = auto ; Set default dtmfmode for sending DTMF. Default: rfc2833
376 ; Other options:
377 ; info : SIP INFO messages (application/dtmf-relay)
378 ; shortinfo : SIP INFO messages (application/dtmf)
379 ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
380 ; auto : Use rfc2833 if offered, inband otherwise
382;compactheaders = yes ; send compact sip headers.
a52025b1 384videosupport=yes ; Turn on support for SIP video. You need to turn this
385 ; on in this section to get any video support at all.
386 ; You can turn it off on a per peer basis if the general
387 ; video support is enabled, but you can't enable it for
388 ; one peer only without enabling in the general section.
389 ; If you set videosupport to "always", then RTP ports will
390 ; always be set up for video, even on clients that don't
391 ; support it. This assists callfile-derived calls and
392 ; certain transferred calls to use always use video when
393 ; available. [yes|NO|always]
395;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
396 ; Videosupport and maxcallbitrate is settable
397 ; for peers and users as well
398;callevents=no ; generate manager events when sip ua
399 ; performs events (e.g. hold)
400;authfailureevents=no ; generate manager "peerstatus" events when peer can't
401 ; authenticate with Asterisk. Peerstatus will be "rejected".
a52025b1 402alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
403 ; for any reason, always reject with an identical response
404 ; equivalent to valid username and invalid password/hash
405 ; instead of letting the requester know whether there was
406 ; a matching user or peer for their request. This reduces
407 ; the ability of an attacker to scan for valid SIP usernames.
408 ; This option is set to "yes" by default.
410;auth_options_requests = yes ; Enabling this option will authenticate OPTIONS requests just like
411 ; INVITE requests are. By default this option is disabled.
413;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
414 ; order instead of RFC3551 packing order (this is required
415 ; for Sipura and Grandstream ATAs, among others). This is
416 ; contrary to the RFC3551 specification, the peer _should_
417 ; be negotiating AAL2-G726-32 instead :-(
418;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
419;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
420;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
421;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls
422;outboundproxy= ; IPv4 address literal (default port is 5060)
423;outboundproxy=2001:db8::1 ; IPv6 address literal (default port is 5060)
424;outboundproxy= ; IPv4 address literal with explicit port
425;outboundproxy=[2001:db8::1]:5062 ; IPv6 address literal with explicit port
426; ; (could also be tcp,udp) - defining transports on the proxy line only
427; ; applies for the global proxy, otherwise use the transport= option
428;matchexternaddrlocally = yes ; Only substitute the externaddr or externhost setting if it matches
429 ; your localnet setting. Unless you have some sort of strange network
430 ; setup you will not need to enable this.
432;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering
433 ; as any IP address used for staticly defined
434 ; hosts. This helps avoid the configuration
435 ; error of allowing your users to register at
436 ; the same address as a SIP provider.
438;contactdeny= ; Use contactpermit and contactdeny to
439;contactpermit= ; restrict at what IPs your users may
440 ; register their phones.
442;engine=asterisk ; RTP engine to use when communicating with the device
445; If regcontext is specified, Asterisk will dynamically create and destroy a
446; NoOp priority 1 extension for a given peer who registers or unregisters with
447; us and have a "regexten=" configuration item.
448; Multiple contexts may be specified by separating them with '&'. The
449; actual extension is the 'regexten' parameter of the registering peer or its
450; name if 'regexten' is not provided. If more than one context is provided,
451; the context must be specified within regexten by appending the desired
452; context after '@'. More than one regexten may be supplied if they are
453; separated by '&'. Patterns may be used in regexten.
456;regextenonqualify=yes ; Default "no"
457 ; If you have qualify on and the peer becomes unreachable
458 ; this setting will enforce inactivation of the regexten
459 ; extension for the peer
460;legacy_useroption_parsing=yes ; Default "no" ; If you have this option enabled and there are semicolons
461 ; in the user field of a sip URI, the field be truncated
462 ; at the first semicolon seen. This effectively makes
463 ; semicolon a non-usable character for peer names, extensions,
464 ; and maybe other, less tested things. This can be useful
465 ; for improving compatability with devices that like to use
466 ; user options for whatever reason. The behavior is similar to
467 ; how SIP URI's were typically handled in 1.6.2, hence the name.
469; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
470; in square brackets. For example, the caller id value 555.5555 becomes 5555555
471; when this option is enabled. Disabling this option results in no modification
472; of the caller id value, which is necessary when the caller id represents something
473; that must be preserved. This option can only be used in the [general] section.
474; By default this option is on.
476;shrinkcallerid=yes ; on by default
479;use_q850_reason = no ; Default "no"
480 ; Set to yes add Reason header and use Reason header if it is available.
482;------------------------ TLS settings ------------------------------------------------------------
483;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem format only) to use for TLS connections
484 ; default is to look for "asterisk.pem" in current directory
486;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem format only) for TLS connections.
487 ; If no tlsprivatekey is specified, tlscertfile is searched for
488 ; for both public and private key.
491; If the server your connecting to uses a self signed certificate
492; you should have their certificate installed here so the code can
493; verify the authenticity of their certificate.
496; A directory full of CA certificates. The files must be named with
497; the CA subject name hash value.
498; (see man SSL_CTX_load_verify_locations for more info)
501; If set to yes, don't verify the servers certificate when acting as
502; a client. If you don't have the server's CA certificate you can
503; set this and it will connect without requiring tlscafile to be set.
504; Default is no.
506;tlscipher=<SSL cipher string>
507; A string specifying which SSL ciphers to use or not use
508; A list of valid SSL cipher strings can be found at:
509; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
511;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2.
512 ; Specify protocol for outbound client connections.
513 ; If left unspecified, the default is sslv2.
515;--------------------------- SIP timers ----------------------------------------------------
516; These timers are used primarily in INVITE transactions.
517; The default for Timer T1 is 500 ms or the measured run-trip time between
518; Asterisk and the device if you have qualify=yes for the device.
520;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
521 ; Defaults to 100 ms
522;timert1=500 ; Default T1 timer
523 ; Defaults to 500 ms or the measured round-trip
524 ; time to a peer (qualify=yes).
525;timerb=32000 ; Call setup timer. If a provisional response is not received
526 ; in this amount of time, the call will autocongest
527 ; Defaults to 64*timert1
529;--------------------------- RTP timers ----------------------------------------------------
530; These timers are currently used for both audio and video streams. The RTP timeouts
531; are only applied to the audio channel.
532; The settings are settable in the global section as well as per device
a52025b1 534rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
535 ; on the audio channel
536 ; when we're not on hold. This is to be able to hangup
537 ; a call in the case of a phone disappearing from the net,
538 ; like a powerloss or grandma tripping over a cable.
a52025b1 539rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
540 ; on the audio channel
541 ; when we're on hold (must be > rtptimeout)
a52025b1 542rtpkeepalive=60 ; Send keepalives in the RTP stream to keep NAT open
543 ; (default is off - zero)
545;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
546; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
547; This mechanism can detect and reclaim SIP channels that do not terminate through normal
548; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
549; The operation of Session-Timers is driven by the following configuration parameters:
551; * session-timers - Session-Timers feature operates in the following three modes:
552; originate : Request and run session-timers always
553; accept : Run session-timers only when requested by other UA
554; refuse : Do not run session timers in any case
555; The default mode of operation is 'accept'.
556; * session-expires - Maximum session refresh interval in seconds. Defaults to 1800 secs.
557; * session-minse - Minimum session refresh interval in seconds. Defualts to 90 secs.
558; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
565;--------------------------- SIP DEBUGGING ---------------------------------------------------
566;sipdebug = yes ; Turn on SIP debugging by default, from
567 ; the moment the channel loads this configuration
568;recordhistory=yes ; Record SIP history by default
569 ; (see sip history / sip no history)
570;dumphistory=yes ; Dump SIP history at end of SIP dialogue
571 ; SIP history is output to the DEBUG logging channel
574;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
575; You can subscribe to the status of extensions with a "hint" priority
576; (See extensions.conf.sample for examples)
577; chan_sip support two major formats for notifications: dialog-info and SIMPLE
579; You will get more detailed reports (busy etc) if you have a call counter enabled
580; for a device.
582; If you set the busylevel, we will indicate busy when we have a number of calls that
583; matches the busylevel treshold.
585; For queues, you will need this level of detail in status reporting, regardless
586; if you use SIP subscriptions. Queues and manager use the same internal interface
587; for reading status information.
589; Note: Subscriptions does not work if you have a realtime dialplan and use the
590; realtime switch.
592allowsubscribe=yes ; Disable support for subscriptions. (Default is yes)
593subscribecontext = AUF-local ; Set a specific context for SUBSCRIBE requests
594 ; Useful to limit subscriptions to local extensions
595 ; Settable per peer/user also
a52025b1 596notifyringing = yes ; Control whether subscriptions already INUSE get sent
3802e567 597 ; RINGING when another call is sent (default: yes)
a52025b1 598notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
599 ; Turning on notifyringing and notifyhold will add a lot
600 ; more database transactions if you are using realtime.
601;notifycid = yes ; Control whether caller ID information is sent along with
602 ; dialog-info+xml notifications (supported by snom phones).
603 ; Note that this feature will only work properly when the
604 ; incoming call is using the same extension and context that
605 ; is being used as the hint for the called extension. This means
606 ; that it won't work when using subscribecontext for your sip
607 ; user or peer (if subscribecontext is different than context).
608 ; This is also limited to a single caller, meaning that if an
609 ; extension is ringing because multiple calls are incoming,
610 ; only one will be used as the source of caller ID. Specify
611 ; 'ignore-context' to ignore the called context when looking
612 ; for the caller's channel. The default value is 'no.' Setting
613 ; notifycid to 'ignore-context' also causes call-pickups attempted
614 ; via SNOM's NOTIFY mechanism to set the context for the call pickup
615 ; to PICKUPMARK.
616;callcounter = yes ; Enable call counters on devices. This can be set per
617 ; device too.
619;----------------------------------------- T.38 FAX SUPPORT ----------------------------------
621; This setting is available in the [general] section as well as in device configurations.
622; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off.
624; t38pt_udptl = yes ; Enables T.38 with FEC error correction.
625; t38pt_udptl = yes,fec ; Enables T.38 with FEC error correction.
626; t38pt_udptl = yes,redundancy ; Enables T.38 with redundancy error correction.
627; t38pt_udptl = yes,none ; Enables T.38 with no error correction.
629; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that
630; is based on an incorrect interpretation of the T.38 recommendation, and results in failures
631; because Asterisk does not believe it can send T.38 packets of a reasonable size to that
632; endpoint (Cisco media gateways are one example of this situation). In these cases, during a
633; T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL
634; stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you
635; can set an override (globally, or on a per-device basis) to make Asterisk ignore the
636; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead.
637; This can be done by appending 'maxdatagram=<value>' to the t38pt_udptl configuration option,
638; like this:
640; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error correction and overrides
641; ; the other endpoint's provided value to assume we can
642; ; send 400 byte T.38 FAX packets to it.
644; FAX detection will cause the SIP channel to jump to the 'fax' extension (if it exists)
645; based one or more events being detected. The events that can be detected are an incoming
646; CNG tone or an incoming T.38 re-INVITE request.
648; faxdetect = yes ; Default 'no', 'yes' enables both CNG and T.38 detection
649; faxdetect = cng ; Enables only CNG detection
650; faxdetect = t38 ; Enables only T.38 detection
652;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
653; Asterisk can register as a SIP user agent to a SIP proxy (provider)
654; Format for the register statement is:
655; register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
659; domain is either
660; - domain in DNS
661; - host name in DNS
662; - the name of a peer defined below or in realtime
663; The domain is where you register your username, so your SIP uri you are registering to
664; is username@domain
666; If no extension is given, the 's' extension is used. The extension needs to
667; be defined in extensions.conf to be able to accept calls from this SIP proxy
668; (provider).
670; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
671; this is equivalent to having the following line in the general section:
673; register => username:secret@host/callbackextension
675; and more readable because you don't have to write the parameters in two places
676; (note that the "port" is ignored - this is a bug that should be fixed).
678; Note that a register= line doesn't mean that we will match the incoming call in any
679; other way than described above. If you want to control where the call enters your
680; dialplan, which context, you want to define a peer with the hostname of the provider's
681; server. If the provider has multiple servers to place calls to your system, you need
682; a peer for each server.
684; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may
685; contain a port number. Since the logical separator between a host and port number is a
686; ':' character, and this character is already used to separate between the optional "secret"
687; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish
688; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
689; they are blank. See the third example below for an illustration.
692; Examples:
694;register => 1234:password@mysipprovider.com
696; This will pass incoming calls to the 's' extension
699;register => 2345:password@sip_proxy/1234
701; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
702; connect to local extension 1234 in extensions.conf, default context,
703; unless you configure a [sip_proxy] section below, and configure a
704; context.
705; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
706; Tip 2: Use separate inbound and outbound sections for SIP providers
707; (instead of type=friend) if you have calls in both directions
709;register => 3456@mydomain:5082::@mysipprovider.com
711; Note that in this example, the optional authuser and secret portions have
712; been left blank because we have specified a port in the user section
714;register => tls://username:xxxxxx@sip-tls-proxy.example.org
716; The 'transport' part defaults to 'udp' but may also be 'tcp' or 'tls'.
717; Using 'udp://' explicitly is also useful in case the username part
718; contains a '/' ('user/name').
720;registertimeout=20 ; retry registration calls every 20 seconds (default)
721;registerattempts=10 ; Number of registration attempts before we give up
722 ; 0 = continue forever, hammering the other server
723 ; until it accepts the registration
724 ; Default is 0 tries, continue forever
726;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
727; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
728; by other phones. At this time, you can only subscribe using UDP as the transport.
729; Format for the mwi register statement is:
730; mwi => user[:secret[:authuser]]@host[:port]/mailbox
732; Examples:
733;mwi => 1234:password@mysipprovider.com/1234
734;mwi => 1234:password@myportprovider.com:6969/1234
735;mwi => 1234:password:authuser@myauthprovider.com/1234
736;mwi => 1234:password:authuser@myauthportprovider.com:6969/1234
738; MWI received will be stored in the 1234 mailbox of the SIP_Remote context. It can be used by other phones by following the below:
739; mailbox=1234@SIP_Remote
740;----------------------------------------- NAT SUPPORT ------------------------
741; si votre serveur Asterisk est derrière un système DNAT, il faut indiquer
742; ici son adresse IP publique.
744; ********** A FAIRE DANS /etc/asterisk/auf/sip-general.local
745; ********** sinon la modification sera effacée à la prochaine mise à jour...
748; WARNING: SIP operation behind a NAT is tricky and you really need
749; to read and understand well the following section.
751; When Asterisk is behind a NAT device, the "local" address (and port) that
752; a socket is bound to has different values when seen from the inside or
753; from the outside of the NATted network. Unfortunately this address must
754; be communicated to the outside (e.g. in SIP and SDP messages), and in
755; order to determine the correct value Asterisk needs to know:
757; + whether it is talking to someone "inside" or "outside" of the NATted network.
758; This is configured by assigning the "localnet" parameter with a list
759; of network addresses that are considered "inside" of the NATted network.
761; Multiple entries are allowed, e.g. a reasonable set is the following:
763; localnet= ; RFC 1918 addresses
764; localnet= ; Also RFC1918
765; localnet= ; Another RFC1918 with CIDR notation
766; localnet= ; Zero conf local network
768; réseaux locaux avec lesquels il ne faut pas faire de NAT
772localnet= ; ZeroConf
775; + the "externally visible" address and port number to be used when talking
776; to a host outside the NAT. This information is derived by one of the
777; following (mutually exclusive) config file parameters:
779; a. "externaddr = hostname[:port]" specifies a static address[:port] to
780; be used in SIP and SDP messages.
781; The hostname is looked up only once, when [re]loading sip.conf .
782; If a port number is not present, use the port specified in the "udpbindaddr"
783; (which is not guaranteed to work correctly, because a NAT box might remap the
784; port number as well as the address).
785; This approach can be useful if you have a NAT device where you can
786; configure the mapping statically. Examples:
788; externaddr = ; use this address.
789; externaddr = ; use this address and port.
790; externaddr = mynat.my.org:12600 ; Public address of my nat box.
791; externtcpport = 9900 ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT.
792; ; externtcpport will default to the externaddr or externhost port if either one is set.
793; externtlsport = 12600 ; The externally mapped tls port, when Asterisk is behind a static NAT or PAT.
794; ; externtlsport port will default to the RFC designated port of 5061.
796; b. "externhost = hostname[:port]" is similar to "externaddr" except
797; that the hostname is looked up every "externrefresh" seconds
798; (default 10s). This can be useful when your NAT device lets you choose
799; the port mapping, but the IP address is dynamic.
800; Beware, you might suffer from service disruption when the name server
801; resolution fails. Examples:
803; externhost=foo.dyndns.net ; refreshed periodically
804; externrefresh=180 ; change the refresh interval
806; Note that at the moment all these mechanism work only for the SIP socket.
807; The IP address discovered with externaddr/externhost is reused for
808; media sessions as well, but the port numbers are not remapped so you
809; may still experience problems.
811; NOTE 1: in some cases, NAT boxes will use different port numbers in
812; the internal<->external mapping. In these cases, the "externaddr" and
813; "externhost" might not help you configure addresses properly.
815; NOTE 2: when using "externaddr" or "externhost", the address part is
816; also used as the external address for media sessions. Thus, the port
817; information in the SDP may be wrong!
819; In addition to the above, Asterisk has an additional "nat" parameter to
820; address NAT-related issues in incoming SIP or media sessions.
821; In particular, depending on the 'nat= ' settings described below, Asterisk
822; may override the address/port information specified in the SIP/SDP messages,
823; and use the information (sender address) supplied by the network stack instead.
824; However, this is only useful if the external traffic can reach us.
825; The following settings are allowed (both globally and in individual sections):
827; nat = no ; Use rport if the remote side says to use it.
a52025b1 828nat=no
829; nat = force_rport ; Force rport to always be on. (default)
830; nat = yes ; Force rport to always be on and perform comedia RTP handling.
831; nat = comedia ; Use rport if the remote side says to use it and perform comedia RTP handling.
833; 'comedia RTP handling' refers to the technique of sending RTP to the port that the
834; the other endpoint's RTP arrived from, and means 'connection-oriented media'. This is
835; only partially related to RFC 4145 which was referred to as COMEDIA while it was in
836; draft form. This method is used to accomodate endpoints that may be located behind
837; NAT devices, and as such the port number they tell Asterisk to send RTP packets to
838; for their media streams is not actual port number that will be used on the nearer
839; side of the NAT.
841; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
842; the nat setting in a peer definition, then the peer username will be discoverable
843; by outside parties as Asterisk will respond to different ports for defined and
844; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
845; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the
846; other, then valid peers with settings differing from those in the general section will
847; be discoverable.
849; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by
850; RFC 4961; Asterisk will always send RTP packets from the same port number it expects
851; to receive them on.
853; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using
854; the media_address configuration option. This is only applicable to the general section and
855; can not be set per-user or per-peer.
857; media_address =
859; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the
860; perceived external network address has changed. When the stun_monitor is installed and
861; configured, chan_sip will renew all outbound registrations when the monitor detects any sort
862; of network change has occurred. By default this option is enabled, but only takes effect once
863; res_stun_monitor is configured. If res_stun_monitor is enabled and you wish to not
864; generate all outbound registrations on a network change, use the option below to disable
865; this feature.
867; subscribe_network_change_event = yes ; on by default
869;----------------------------------- MEDIA HANDLING --------------------------------
870; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
871; no reason for Asterisk to stay in the media path, the media will be redirected.
872; This does not really work well in the case where Asterisk is outside and the
873; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
876directmedia=no ; Asterisk reste sur le chemin du flux audio
878;directmedia=yes ; Asterisk by default tries to redirect the
879 ; RTP media stream to go directly from
880 ; the caller to the callee. Some devices do not
881 ; support this (especially if one of them is behind a NAT).
882 ; The default setting is YES. If you have all clients
883 ; behind a NAT, or for some other reason want Asterisk to
884 ; stay in the audio path, you may want to turn this off.
886 ; This setting also affect direct RTP
887 ; at call setup (a new feature in 1.4 - setting up the
888 ; call directly between the endpoints instead of sending
889 ; a re-INVITE).
891 ; Additionally this option does not disable all reINVITE operations.
892 ; It only controls Asterisk generating reINVITEs for the specific
893 ; purpose of setting up a direct media path. If a reINVITE is
894 ; needed to switch a media stream to inactive (when placed on
895 ; hold) or to T.38, it will still be done, regardless of this
896 ; setting. Note that direct T.38 is not supported.
898;directmedia=nonat ; An additional option is to allow media path redirection
899 ; (reinvite) but only when the peer where the media is being
900 ; sent is known to not be behind a NAT (as the RTP core can
901 ; determine it based on the apparent IP address the media
902 ; arrives from).
904;directmedia=update ; Yet a third option... use UPDATE for media path redirection,
905 ; instead of INVITE. This can be combined with 'nonat', as
906 ; 'directmedia=update,nonat'. It implies 'yes'.
908;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
909 ; the call directly with media peer-2-peer without re-invites.
910 ; Will not work for video and cases where the callee sends
911 ; RTP payloads and fmtp headers in the 200 OK that does not match the
912 ; callers INVITE. This will also fail if directmedia is enabled when
913 ; the device is actually behind NAT.
915;directmediadeny= ; Use directmediapermit and directmediadeny to restrict
916;directmediapermit=; which peers should be able to pass directmedia to each other
917 ; (There is no default setting, this is just an example)
918 ; Use this if some of your phones are on IP addresses that
919 ; can not reach each other directly. This way you can force
920 ; RTP to always flow through asterisk in such cases.
922;ignoresdpversion=yes ; By default, Asterisk will honor the session version
923 ; number in SDP packets and will only modify the SDP
924 ; session if the version number changes. This option will
925 ; force asterisk to ignore the SDP session version number
926 ; and treat all SDP data as new data. This is required
927 ; for devices that send us non standard SDP packets
928 ; (observed with Microsoft OCS). By default this option is
929 ; off.
931;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
932 ; Like the useragent parameter, the default user agent string
933 ; also contains the Asterisk version.
934;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
935 ; This field MUST NOT contain spaces
936;encryption=no ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
937 ; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
938 ; the peer does not support SRTP. Defaults to no.
940;----------------------------------------- REALTIME SUPPORT ------------------------
941; For additional information on ARA, the Asterisk Realtime Architecture,
942; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
944;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
945 ; just like friends added from the config file only on a
946 ; as-needed basis? (yes|no)
948;rtsavesysname=yes ; Save systemname in realtime database at registration
949 ; Default= no
951;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
952 ; If set to yes, when a SIP UA registers successfully, the ip address,
953 ; the origination port, the registration period, and the username of
954 ; the UA will be set to database via realtime.
955 ; If not present, defaults to 'yes'. Note: realtime peers will
956 ; probably not function across reloads in the way that you expect, if
957 ; you turn this option off.
958;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
959 ; as if it had just registered? (yes|no|<seconds>)
960 ; If set to yes, when the registration expires, the friend will
961 ; vanish from the configuration until requested again. If set
962 ; to an integer, friends expire within this number of seconds
963 ; instead of the registration interval.
965;ignoreregexpire=yes ; Enabling this setting has two functions:
966 ;
967 ; For non-realtime peers, when their registration expires, the
968 ; information will _not_ be removed from memory or the Asterisk database
969 ; if you attempt to place a call to the peer, the existing information
970 ; will be used in spite of it having expired
971 ;
972 ; For realtime peers, when the peer is retrieved from realtime storage,
973 ; the registration information will be used regardless of whether
974 ; it has expired or not; if it expires while the realtime peer
975 ; is still in memory (due to caching or other reasons), the
976 ; information will not be removed from realtime storage
978;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
979; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
980; domains, each of which can direct the call to a specific context if desired.
981; By default, all domains are accepted and sent to the default context or the
982; context associated with the user/peer placing the call.
983; REGISTER to non-local domains will be automatically denied if a domain
984; list is configured.
986; Domains can be specified using:
987; domain=<domain>[,<context>]
988; Examples:
989; domain=myasterisk.dom
990; domain=customer.com,customer-context
992; In addition, all the 'default' domains associated with a server should be
993; added if incoming request filtering is desired.
994; autodomain=yes
996; To disallow requests for domains not serviced by this server:
997; allowexternaldomains=no
999; **** GESTION DES DOMAINES SIP : voir /etc/asterisk/auf/sip-general.local
1000; **** pour y indiquer le nom du domaine SIP géré localement
1003 ; Add domain and configure incoming context
1004 ; for external calls to this domain
1005;domain= ; Add IP address as local domain
1006 ; You can have several "domain" settings
a52025b1 1007allowexternaldomains=yes ; Disable INVITE and REFER to non-local domains
3802e567 1008 ; Default is yes
5bc84f52 1009autodomain=no ; Turn this on to have Asterisk add local host
1010 ; name and local IP to domain list.
1012; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
1013 ; non-peers, use your primary domain "identity"
1014 ; for From: headers instead of just your IP
1015 ; address. This is to be polite and
1016 ; it may be a mandatory requirement for some
1017 ; destinations which do not have a prior
1018 ; account relationship with your server.
1020;------------------------------ Advice of Charge CONFIGURATION --------------------------
1021; snom_aoc_enabled = yes; ; This options turns on and off support for sending AOC-D and
1022 ; AOC-E to snom endpoints. This option can be used both in the
1023 ; peer and global scope. The default for this option is off.
1026;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
a52025b1 1027jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
1028 ; SIP channel. Defaults to "no". An enabled jitterbuffer will
1029 ; be used only if the sending side can create and the receiving
1030 ; side can not accept jitter. The SIP channel can accept jitter,
1031 ; thus a jitterbuffer on the receive SIP side will be used only
1032 ; if it is forced and enabled.
1034; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
1035 ; channel. Defaults to "no".
a52025b1 1037jbmaxsize = 500 ; Max length of the jitterbuffer in milliseconds.
1039; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
1040 ; resynchronized. Useful to improve the quality of the voice, with
1041 ; big jumps in/broken timestamps, usually sent from exotic devices
1042 ; and programs. Defaults to 1000.
1044; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
1045 ; channel. Two implementations are currently available - "fixed"
1046 ; (with size always equals to jbmaxsize) and "adaptive" (with
1047 ; variable size, actually the new jb of IAX2). Defaults to fixed.
1049; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
1050 ; The option represents the number of milliseconds by which the new jitter buffer
1051 ; will pad its size. the default is 40, so without modification, the new
1052 ; jitter buffer will set its size to the jitter value plus 40 milliseconds.
1053 ; increasing this value may help if your network normally has low jitter,
1054 ; but occasionally has spikes.
1056; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
1058;----------------------------- SIP_CAUSE reporting ---------------------------------
1059; storesipcause = no ; This option causes chan_sip to set the
1060 ; HASH(SIP_CAUSE,<channel name>) channel variable
1061 ; to the value of the last sip response.
1062 ; WARNING: enabling this option carries a
1063 ; significant performance burden. It should only
1064 ; be used in low call volume situations. This
1065 ; option defaults to "no".
1069#include "auf/sip-general.local"
1072; Global credentials for outbound calls, i.e. when a proxy challenges your
1073; Asterisk server for authentication. These credentials override
1074; any credentials in peer/register definition if realm is matched.
1076; This way, Asterisk can authenticate for outbound calls to other
1077; realms. We match realm on the proxy challenge and pick an set of
1078; credentials from this list
1079; Syntax:
1080; auth = <user>:<secret>@<realm>
1081; auth = <user>#<md5secret>@<realm>
1082; Example:
1085; You may also add auth= statements to [peer] definitions
1086; Peer auth= override all other authentication settings if we match on realm
1091; SIP entities have a 'type' which determines their roles within Asterisk.
1092; * For entities with 'type=peer':
1093; Peers handle both inbound and outbound calls and are matched by ip/port, so for
1094; The case of incoming calls from the peer, the IP address must match in order for
1095; The invitation to work. This means calls made from either direction won't work if
1096; The peer is unregistered while host=dynamic or if the host is otherise not set to
1097; the correct IP of the sender.
1098; * For entities with 'type=user':
1099; Asterisk users handle inbound calls only (meaning they call Asterisk, Asterisk can't
1100; call them) and are matched by their authorization information (authname and secret).
1101; Asterisk doesn't rely on their IP and will accept calls regardless of the host setting
1102; as long as the incoming SIP invite authorizes successfully.
1103; * For entities with 'type=friend':
1104; Asterisk will create the entity as both a friend and a peer. Asterisk will accept
1105; calls from friends like it would for users, requiring only that the authorization
1106; matches rather than the IP address. Since it is also a peer, a friend entity can
1107; be called as long as its IP is known to Asterisk. In the case of host=dynamic,
1108; this means it is necessary for the entity to register before Asterisk can call it.
1110; Use remotesecret for outbound authentication, and secret for authenticating
1111; inbound requests. For historical reasons, if no remotesecret is supplied for an
1112; outbound registration or call, the secret will be used.
1114; For device names, we recommend using only a-z, numerics (0-9) and underscore
1116; For local phones, type=friend works most of the time
1118; If you have one-way audio, you probably have NAT problems.
1119; If Asterisk is on a public IP, and the phone is inside of a NAT device
1120; you will need to configure nat option for those phones.
1121; Also, turn on qualify=yes to keep the nat session open
1123; Configuration options available
1124; --------------------
1125; context
1126; callingpres
1127; permit
1128; deny
1129; secret
1130; md5secret
1131; remotesecret
1132; transport
1133; dtmfmode
1134; directmedia
1135; nat
1136; callgroup
1137; pickupgroup
1138; language
1139; allow
1140; disallow
1141; insecure
1142; trustrpid
1143; progressinband
1144; promiscredir
1145; useclientcode
1146; accountcode
1147; setvar
1148; callerid
1149; amaflags
1150; callcounter
1151; busylevel
1152; allowoverlap
1153; allowsubscribe
1154; allowtransfer
1155; ignoresdpversion
1156; subscribecontext
1157; template
1158; videosupport
1159; maxcallbitrate
1160; rfc2833compensate
1161; mailbox
1162; session-timers
1163; session-expires
1164; session-minse
1165; session-refresher
1166; t38pt_usertpsource
1167; regexten
1168; fromdomain
1169; fromuser
1170; host
1171; port
1172; qualify
1173; defaultip
1174; defaultuser
1175; rtptimeout
1176; rtpholdtimeout
1177; sendrpid
1178; outboundproxy
1179; rfc2833compensate
1180; callbackextension
1181; registertrying
1182; timert1
1183; timerb
1184; qualifyfreq
1185; t38pt_usertpsource
1186; contactpermit ; Limit what a host may register as (a neat trick
1187; contactdeny ; is to register at the same IP as a SIP provider,
1188; ; then call oneself, and get redirected to that
1189; ; same location).
1190; directmediapermit
1191; directmediadeny
1192; unsolicited_mailbox
1193; use_q850_reason
1194; maxforwards
1195; encryption
1198; For incoming calls only. Example: FWD (Free World Dialup)
1199; We match on IP address of the proxy for incoming calls
1200; since we can not match on username (caller id)
1206;type=peer ; we only want to call out, not be called
1207;remotesecret=guessit ; Our password to their service
1208;defaultuser=yourusername ; Authentication user for outbound proxies
1209;fromuser=yourusername ; Many SIP providers require this!
1212;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will
1213; ; accept both tcp and udp. The default transport type is only used for
1214; ; outbound messages until a Registration takes place. During the
1215; ; peer Registration the transport type may change to another supported
1216; ; type if the peer requests so.
1218;usereqphone=yes ; This provider requires ";user=phone" on URI
1219;callcounter=yes ; Enable call counter
1220;busylevel=2 ; Signal busy at 2 or more calls
1221;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
1222;port=80 ; The port number we want to connect to on the remote side
1223 ; Also used as "defaultport" in combination with "defaultip" settings
1225;--- sample definition for a provider
1229;fromuser=4015552299 ; how your provider knows you
1230;remotesecret=youwillneverguessit ; The password we use to authenticate to them
1231;secret=gissadetdu ; The password they use to contact us
1232;callbackextension=123 ; Register with this server and require calls coming back to this extension
1233;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
1234; ; accept both tcp and udp. Default is udp. The first transport
1235; ; listed will always be used for outgoing connections.
1236;unsolicited_mailbox=4015552299 ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old
1237; ; message count will be stored in the configured virtual mailbox. It can be used
1238; ; by any device supporting MWI by specifying <configured value>@SIP_Remote as the
1239; ; mailbox.
1242; Because you might have a large number of similar sections, it is generally
1243; convenient to use templates for the common parameters, and add them
1244; the the various sections. Examples are below, and we can even leave
1245; the templates uncommented as they will not harm:
1247; [basic-options](!) ; a template
1248; dtmfmode=rfc2833
1249; context=from-office
1250; type=friend
3802e567 1251
1252;[natted-phone](!,basic-options) ; another template inheriting basic-options
1253; directmedia=no
1254; host=dynamic
3802e567 1255
1256;[public-phone](!,basic-options) ; another template inheriting basic-options
1257; directmedia=yes
3802e567 1258
1259;[my-codecs](!) ; a template for my preferred codecs
1260; disallow=all
1261; allow=ilbc
1262; allow=g729
1263; allow=gsm
1264; allow=g723
1265; allow=ulaw
3802e567 1266
1267;[ulaw-phone](!) ; and another one for ulaw-only
1268; disallow=all
1269; allow=ulaw
1271; and finally instantiate a few phones
1273; [2133](natted-phone,my-codecs)
1274; secret = peekaboo
1275; [2134](natted-phone,ulaw-phone)
1276; secret = not_very_secret
1277; [2136](public-phone,ulaw-phone)
1278; secret = not_very_secret_either
1279; ...
1282; Comptes pour postes clients locaux SIP
1283#include "auf/sip.local"
1285; Comptes pour fournisseurs de service SIP
1286#include "auf/sip-peers.local"