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
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
18 ;-----------------------------------------------------------
19 ; In the dialplan (extensions.conf) you can use several
20 ; syntaxes for dialing SIP devices.
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
30 ; devicename is defined as a peer in a section below.
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.
48 ; SIP/sales:topsecret::account02@domain.com:5062
49 ; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
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
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
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
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.
103 ;-----------------------------------------------------------------------------
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.
115 context=default ; Default context for incoming calls
116 allowguest=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)
125 allowoverlap=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.
132 allowtransfer=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=192.0.2.1
153 ; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1
154 ; c) Listen on the IPv4 wildcard. Example: bindaddr=0.0.0.0
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
165 ; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061
167 ; IPv4 example: bindaddr=0.0.0.0:5062
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
178 udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
179 ; Optionally add a port number, 192.168.1.1:5062 (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.
208 tcpenable=no ; Enable server for incoming TCP connections (default is no)
209 tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
210 ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
212 ;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
213 ;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
214 ; Optionally add a port number, 192.168.1.1:5063 (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)
230 transport=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
235 srvlookup=yes ; Enable DNS SRV lookups on outbound calls
236 ; Note: Asterisk only uses the first host
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.
251 tos_sip=cs3 ; Sets TOS for SIP packets.
252 tos_audio=ef ; Sets TOS for RTP audio packets.
253 tos_video=af41 ; Sets TOS for RTP video packets.
254 tos_text=af41 ; Sets TOS for RTP text packets.
256 cos_sip=3 ; Sets 802.1p priority for SIP packets.
257 cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
258 cos_video=4 ; Sets 802.1p priority for RTP video packets.
259 cos_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
272 ;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
274 ;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
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
285 ;vmexten=voicemail ; dialplan extension to reach mailbox sets the
286 ; Message-Account in the MWI notify message
287 ; defaults to "asterisk"
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
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.
305 disallow=all ; First disallow all codecs
307 allow=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.
322 ;mohinterpret=default
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
333 language=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.
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
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
375 dtmfmode = auto ; Set default dtmfmode for sending DTMF. Default: rfc2833
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.
384 videosupport=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".
402 alwaysauthreject = 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=192.0.2.1 ; IPv4 address literal (default port is 5060)
423 ;outboundproxy=2001:db8::1 ; IPv6 address literal (default port is 5060)
424 ;outboundproxy=192.168.0.2.1:5062 ; 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=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to
439 ;contactpermit=172.16.0.0/255.255.0.0 ; 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.
455 ;regcontext=sipregistrations
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.
490 ;tlscafile=</path/to/certificate>
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.
495 ;tlscapath=</path/to/ca/dir>
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)
500 ;tlsdontverifyserver=[yes|no]
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.
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
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
534 rtptimeout=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.
539 rtpholdtimeout=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)
542 rtpkeepalive=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'.
560 ;session-timers=originate
563 ;session-refresher=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
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
592 allowsubscribe=yes ; Disable support for subscriptions. (Default is yes)
593 subscribecontext = AUF-local ; Set a specific context for SUBSCRIBE requests
594 ; Useful to limit subscriptions to local extensions
595 ; Settable per peer/user also
596 notifyringing = yes ; Control whether subscriptions already INUSE get sent
597 ; RINGING when another call is sent (default: yes)
598 notifyhold = 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
616 ;callcounter = yes ; Enable call counters on devices. This can be set per
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,
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]
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
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
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.
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
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
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.
760 ; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
761 ; Multiple entries are allowed, e.g. a reasonable set is the following:
763 ; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
764 ; localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
765 ; localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
766 ; localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
768 ; réseaux locaux avec lesquels il ne faut pas faire de NAT
770 localnet=172.16.0.0/12
771 localnet=192.168.0.0/16
772 localnet=169.254.0.0/16 ; 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 = 12.34.56.78 ; use this address.
789 ; externaddr = 12.34.56.78:9900 ; 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.
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
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
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 = 172.16.42.1
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
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.
876 directmedia=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
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
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=0.0.0.0/0 ; Use directmediapermit and directmediadeny to restrict
916 ;directmediapermit=172.16.0.0/16; 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
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
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:
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
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>]
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.
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
1002 ;domain=mydomain.tld,mydomain-incoming
1003 ; Add domain and configure incoming context
1004 ; for external calls to this domain
1005 ;domain=1.2.3.4 ; Add IP address as local domain
1006 ; You can have several "domain" settings
1007 allowexternaldomains=yes ; Disable INVITE and REFER to non-local domains
1009 autodomain=yes ; 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 --------------------------
1027 jbenable = 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".
1037 jbmaxsize = 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".
1067 ;-----------------------------------------------------------------------------------
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
1080 ; auth = <user>:<secret>@<realm>
1081 ; auth = <user>#<md5secret>@<realm>
1083 ;auth=mark:topsecret@digium.com
1085 ; You may also add auth= statements to [peer] definitions
1086 ; Peer auth= override all other authentication settings if we match on realm
1088 ;------------------------------------------------------------------------------
1089 ; DEVICE CONFIGURATION
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 ; --------------------
1166 ; t38pt_usertpsource
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
1192 ; unsolicited_mailbox
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)
1203 ;host=fwd.pulver.com
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!
1210 ;fromdomain=provider.sip.domain
1211 ;host=box.provider.com
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
1228 ;host=sip.provider1.com
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
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
1249 ; context=from-office
1252 ;[natted-phone](!,basic-options) ; another template inheriting basic-options
1256 ;[public-phone](!,basic-options) ; another template inheriting basic-options
1259 ;[my-codecs](!) ; a template for my preferred codecs
1267 ;[ulaw-phone](!) ; and another one for ulaw-only
1271 ; and finally instantiate a few phones
1273 ; [2133](natted-phone,my-codecs)
1275 ; [2134](natted-phone,ulaw-phone)
1276 ; secret = not_very_secret
1277 ; [2136](public-phone,ulaw-phone)
1278 ; secret = not_very_secret_either
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"
projets / asterisk-config-auf.git / blob