2 ; SIP Configuration example for Asterisk
4 ; Syntax for specifying a SIP device in extensions.conf is
5 ; SIP/devicename where devicename is defined in a section below.
8 ; SIP/username@domain to call any SIP user on the Internet
9 ; (Don't forget to enable DNS SRV records if you want to use this)
11 ; If you define a SIP proxy as a peer below, you may call
12 ; SIP/proxyhostname/user or SIP/user@proxyhostname
13 ; where the proxyhostname is defined in a section below
15 ; Useful CLI commands to check peers/users:
16 ; sip show peers Show all SIP peers (including friends)
17 ; sip show users Show all SIP users (including friends)
18 ; sip show registry Show status of hosts we register with
20 ; sip set debug Show all SIP messages
22 ; sip reload Reload configuration file
23 ; Active SIP peers will not be reconfigured
27 context=default ; Default context for incoming calls
28 ;allowguest=no ; Allow or reject guest calls (default is yes)
29 ;match_auth_username=yes ; if available, match user entry using the
30 ; 'username' field from the authentication line
31 ; instead of the From: field.
33 allowoverlap=no ; Disable overlap dialing support. (Default is yes)
34 ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
36 ;realm=mydomain.tld ; Realm for digest authentication
37 ; defaults to "asterisk". If you set a system name in
38 ; asterisk.conf, it defaults to that system name
39 ; Realms MUST be globally unique according to RFC 3261
40 ; Set this to your host name or domain name
41 bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
42 ; bindport is the local UDP port that Asterisk will listen on
43 bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
44 srvlookup=yes ; Enable DNS SRV lookups on outbound calls
45 ; Note: Asterisk only uses the first host
47 ; Disabling DNS SRV lookups disables the
48 ; ability to place SIP calls based on domain
49 ; names to some other SIP users on the Internet
51 ;domain=mydomain.tld ; Set default domain for this host
52 ; If configured, Asterisk will only allow
53 ; INVITE and REFER to non-local domains
54 ; Use "sip show domains" to list local domains
55 ;pedantic=yes ; Enable checking of tags in headers,
56 ; international character conversions in URIs
57 ; and multiline formatted headers for strict
58 ; SIP compatibility (defaults to "no")
60 ; See doc/qos.tex for a description of these parameters.
61 ;tos_sip=cs3 ; Sets TOS for SIP packets.
62 ;tos_audio=ef ; Sets TOS for RTP audio packets.
63 ;tos_video=af41 ; Sets TOS for RTP video packets.
64 ;tos_text=af41 ; Sets TOS for RTP text packets.
66 ;cos_sip=4 ; Sets CoS for SIP packets.
67 ;cos_audio=6 ; Sets CoS for RTP audio packets.
68 ;cos_video=5 ; Sets CoS for RTP video packets.
69 ;cos_text=0 ; Sets CoS for RTP text packets.
71 ;maxexpiry=3600 ; Maximum allowed time of incoming registrations
72 ; and subscriptions (seconds)
73 ;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
74 ;defaultexpiry=120 ; Default length of incoming/outgoing registration
75 ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
77 ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
78 ;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
79 ; fully. Enable this option to not get error messages
80 ; when sending MWI to phones with this bug.
81 ;vmexten=voicemail ; dialplan extension to reach mailbox sets the
82 ; Message-Account in the MWI notify message
83 ; defaults to "asterisk"
84 ;disallow=all ; First disallow all codecs
85 ;allow=ulaw ; Allow codecs in order of preference
86 ;allow=ilbc ; see doc/rtp-packetization for framing options
88 ; This option specifies a preference for which music on hold class this channel
89 ; should listen to when put on hold if the music class has not been set on the
90 ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
91 ; channel putting this one on hold did not suggest a music class.
93 ; This option may be specified globally, or on a per-user or per-peer basis.
97 ; This option specifies which music on hold class to suggest to the peer channel
98 ; when this channel places the peer on hold. It may be specified globally or on
99 ; a per-user or per-peer basis.
103 ;language=en ; Default language setting for all users/peers
104 ; This may also be set for individual users/peers
105 ;relaxdtmf=yes ; Relax dtmf handling
106 ;trustrpid = no ; If Remote-Party-ID should be trusted
107 ;sendrpid = yes ; If Remote-Party-ID should be sent
108 ;progressinband=never ; If we should generate in-band ringing always
109 ; use 'never' to never use in-band signalling, even in cases
110 ; where some buggy devices might not render it
111 ; Valid values: yes, no, never Default: never
112 ;useragent=Asterisk PBX ; Allows you to change the user agent string
113 ; The default user agent string also contains the Asterisk
114 ; version. If you don't want to expose this, change the
116 ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
117 ; Note that promiscredir when redirects are made to the
118 ; local system will cause loops since Asterisk is incapable
119 ; of performing a "hairpin" call.
120 ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
121 ; a valid phone number
122 ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
124 ; info : SIP INFO messages
125 ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
126 ; auto : Use rfc2833 if offered, inband otherwise
128 ;compactheaders = yes ; send compact sip headers.
130 ;videosupport=yes ; Turn on support for SIP video. You need to turn this on
131 ; in the this section to get any video support at all.
132 ; You can turn it off on a per peer basis if the general
133 ; video support is enabled, but you can't enable it for
134 ; one peer only without enabling in the general section.
135 ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
136 ; Videosupport and maxcallbitrate is settable
137 ; for peers and users as well
138 ;callevents=no ; generate manager events when sip ua
139 ; performs events (e.g. hold)
140 ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
141 ; for any reason, always reject with '401 Unauthorized'
142 ; instead of letting the requester know whether there was
143 ; a matching user or peer for their request
145 ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
146 ; order instead of RFC3551 packing order (this is required
147 ; for Sipura and Grandstream ATAs, among others). This is
148 ; contrary to the RFC3551 specification, the peer _should_
149 ; be negotiating AAL2-G726-32 instead :-(
150 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
151 ;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
152 ;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
153 ;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
154 ; your localnet setting. Unless you have some sort of strange network
155 ; setup you will not need to enable this.
158 ; If regcontext is specified, Asterisk will dynamically create and destroy a
159 ; NoOp priority 1 extension for a given peer who registers or unregisters with
160 ; us and have a "regexten=" configuration item.
161 ; Multiple contexts may be specified by separating them with '&'. The
162 ; actual extension is the 'regexten' parameter of the registering peer or its
163 ; name if 'regexten' is not provided. If more than one context is provided,
164 ; the context must be specified within regexten by appending the desired
165 ; context after '@'. More than one regexten may be supplied if they are
166 ; separated by '&'. Patterns may be used in regexten.
168 ;regcontext=sipregistrations
169 ;regextenonqualify=yes ; Default "no"
170 ; If you have qualify on and the peer becomes unreachable
171 ; this setting will enforce inactivation of the regexten
172 ; extension for the peer
174 ;--------------------------- RTP timers ----------------------------------------------------
175 ; These timers are currently used for both audio and video streams. The RTP timeouts
176 ; are only applied to the audio channel.
177 ; The settings are settable in the global section as well as per device
179 ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
180 ; on the audio channel
181 ; when we're not on hold. This is to be able to hangup
182 ; a call in the case of a phone disappearing from the net,
183 ; like a powerloss or grandma tripping over a cable.
184 ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
185 ; on the audio channel
186 ; when we're on hold (must be > rtptimeout)
187 ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
188 ; (default is off - zero)
189 ;--------------------------- SIP DEBUGGING ---------------------------------------------------
190 ;sipdebug = yes ; Turn on SIP debugging by default, from
191 ; the moment the channel loads this configuration
192 ;recordhistory=yes ; Record SIP history by default
193 ; (see sip history / sip no history)
194 ;dumphistory=yes ; Dump SIP history at end of SIP dialogue
195 ; SIP history is output to the DEBUG logging channel
198 ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
199 ; You can subscribe to the status of extensions with a "hint" priority
200 ; (See extensions.conf.sample for examples)
201 ; chan_sip support two major formats for notifications: dialog-info and SIMPLE
203 ; You will get more detailed reports (busy etc) if you have a call limit set
204 ; for a device. When the call limit is filled, we will indicate busy. Note that
205 ; you need at least 2 in order to be able to do attended transfers.
207 ; If you set the busy-level in addition to the call limit, we will indicate busy
208 ; when we have a number of calls that matches busy-level, but still allow calls
209 ; up to the call-limit. This allows for transfers while still having blinking
210 ; lamps and queues understanding that a device is busy.
212 ; For queues, you will need this level of detail in status reporting, regardless
213 ; if you use SIP subscriptions. Queues and manager use the same internal interface
214 ; for reading status information.
216 ; Note: Subscriptions does not work if you have a realtime dialplan and use the
219 ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
220 ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
221 ; Useful to limit subscriptions to local extensions
222 ; Settable per peer/user also
223 ;notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
224 ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
225 ; Turning on notifyringing and notifyhold will add a lot
226 ; more database transactions if you are using realtime.
227 ;limitonpeer = yes ; Apply call limits on peers only. This will improve
228 ; status notification when you are using type=friend
229 ; Inbound calls, that really apply to the user part
230 ; of a friend will now be added to and compared with
231 ; the peer limit instead of applying two call limits,
232 ; one for the peer and one for the user.
234 ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
236 ; This setting is available in the [general] section as well as in device configurations.
237 ; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
238 ; both parties have T38 support enabled in their Asterisk configuration
239 ; This has to be enabled in the general section for all devices to work. You can then
240 ; disable it on a per device basis.
242 ; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.
244 ; t38pt_udptl = yes ; Default false
246 ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
247 ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
248 ; Format for the register statement is:
249 ; register => user[:secret[:authuser]]@host[:port][/extension]
251 ; If no extension is given, the 's' extension is used. The extension needs to
252 ; be defined in extensions.conf to be able to accept calls from this SIP proxy
255 ; host is either a host name defined in DNS or the name of a section defined
258 ; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
259 ; this is equivalent to having the following line in the general section:
261 ; register => username:secret@host/callbackextension
263 ; and more readable because you don't have to write the parameters in two places
264 ; (note that the "port" is ignored - this is a bug that should be fixed).
268 ;register => 1234:password@mysipprovider.com
270 ; This will pass incoming calls to the 's' extension
273 ;register => 2345:password@sip_proxy/1234
275 ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
276 ; connect to local extension 1234 in extensions.conf, default context,
277 ; unless you configure a [sip_proxy] section below, and configure a
279 ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
280 ; Tip 2: Use separate type=peer and type=user sections for SIP providers
281 ; (instead of type=friend) if you have calls in both directions
283 ;registertimeout=20 ; retry registration calls every 20 seconds (default)
284 ;registerattempts=10 ; Number of registration attempts before we give up
285 ; 0 = continue forever, hammering the other server
286 ; until it accepts the registration
287 ; Default is 0 tries, continue forever
289 ;----------------------------------------- NAT SUPPORT ------------------------
291 ; WARNING: SIP operation behind a NAT is tricky and you really need
292 ; to read and understand well the following section.
294 ; When Asterisk is behind a NAT device, the "local" address (and port) that
295 ; a socket is bound to has different values when seen from the inside or
296 ; from the outside of the NATted network. Unfortunately this address must
297 ; be communicated to the outside (e.g. in SIP and SDP messages), and in
298 ; order to determine the correct value Asterisk needs to know:
300 ; + whether it is talking to someone "inside" or "outside" of the NATted network.
301 ; This is configured by assigning the "localnet" parameter with a list
302 ; of network addresses that are considered "inside" of the NATted network.
303 ; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
304 ; Multiple entries are allowed, e.g. a reasonable set is the following:
306 ; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
307 ; localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
308 ; localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
309 ; localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
311 ; + the "externally visible" address and port number to be used when talking
312 ; to a host outside the NAT. This information is derived by one of the
313 ; following (mutually exclusive) config file parameters:
315 ; a. "externip = hostname[:port]" specifies a static address[:port] to
316 ; be used in SIP and SDP messages.
317 ; The hostname is looked up only once, when [re]loading sip.conf .
318 ; If a port number is not present, use the "bindport" value (which is
319 ; not guaranteed to work correctly, because a NAT box might remap the
320 ; port number as well as the address).
321 ; This approach can be useful if you have a NAT device where you can
322 ; configure the mapping statically. Examples:
324 ; externip = 12.34.56.78 ; use this address.
325 ; externip = 12.34.56.78:9900 ; use this address and port.
326 ; externip = mynat.my.org:12600 ; Public address of my nat box.
328 ; b. "externhost = hostname[:port]" is similar to "externip" except
329 ; that the hostname is looked up every "externrefresh" seconds
330 ; (default 10s). This can be useful when your NAT device lets you choose
331 ; the port mapping, but the IP address is dynamic.
332 ; Beware, you might suffer from service disruption when the name server
333 ; resolution fails. Examples:
335 ; externhost=foo.dyndns.net ; refreshed periodically
336 ; externrefresh=180 ; change the refresh interval
338 ; c. "stunaddr = stun.server[:port]" queries the STUN server specified
339 ; as an argument to obtain the external address/port.
340 ; Queries are also sent periodically every "externrefresh" seconds
341 ; (as a side effect, sending the query also acts as a keepalive for
342 ; the state entry on the nat box):
344 ; stunaddr = foo.stun.com:3478
347 ; Note that at the moment all these mechanism work only for the SIP socket.
348 ; The IP address discovered with externip/externhost/STUN is reused for
349 ; media sessions as well, but the port numbers are not remapped so you
350 ; may still experience problems.
352 ; NOTE 1: in some cases, NAT boxes will use different port numbers in
353 ; the internal<->external mapping. In these cases, the "externip" and
354 ; "externhost" might not help you configure addresses properly, and you
355 ; really need to use STUN.
357 ; NOTE 2: when using "externip" or "externhost", the address part is
358 ; also used as the external address for media sessions.
359 ; If you use "stunaddr", STUN queries will be sent to the same server
360 ; also from media sockets, and this should permit a correct mapping of
361 ; the port numbers as well.
363 ; In addition to the above, Asterisk has an additional "nat" parameter to
364 ; address NAT-related issues in incoming SIP or media sessions.
365 ; In particular, depending on the 'nat= ' settings described below, Asterisk
366 ; may override the address/port information specified in the SIP/SDP messages,
367 ; and use the information (sender address) supplied by the network stack instead.
368 ; However, this is only useful if the external traffic can reach us.
369 ; The following settings are allowed (both globally and in individual sections):
371 ; nat = no ; default. Use NAT mode only according to RFC3581 (;rport)
372 ; nat = yes ; Always ignore info and assume NAT
373 ; nat = never ; Never attempt NAT mode or RFC3581 support
374 ; nat = route ; route = Assume NAT, don't send rport
375 ; ; (work around more UNIDEN bugs)
377 ;----------------------------------- MEDIA HANDLING --------------------------------
378 ; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
379 ; no reason for Asterisk to stay in the media path, the media will be redirected.
380 ; This does not really work with in the case where Asterisk is outside and have
381 ; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
383 ;canreinvite=yes ; Asterisk by default tries to redirect the
384 ; RTP media stream (audio) to go directly from
385 ; the caller to the callee. Some devices do not
386 ; support this (especially if one of them is behind a NAT).
387 ; The default setting is YES. If you have all clients
388 ; behind a NAT, or for some other reason wants Asterisk to
389 ; stay in the audio path, you may want to turn this off.
391 ; This setting also affect direct RTP
392 ; at call setup (a new feature in 1.4 - setting up the
393 ; call directly between the endpoints instead of sending
396 ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
397 ; the call directly with media peer-2-peer without re-invites.
398 ; Will not work for video and cases where the callee sends
399 ; RTP payloads and fmtp headers in the 200 OK that does not match the
400 ; callers INVITE. This will also fail if canreinvite is enabled when
401 ; the device is actually behind NAT.
403 ;canreinvite=nonat ; An additional option is to allow media path redirection
404 ; (reinvite) but only when the peer where the media is being
405 ; sent is known to not be behind a NAT (as the RTP core can
406 ; determine it based on the apparent IP address the media
409 ;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
410 ; instead of INVITE. This can be combined with 'nonat', as
411 ; 'canreinvite=update,nonat'. It implies 'yes'.
413 ;----------------------------------------- REALTIME SUPPORT ------------------------
414 ; For additional information on ARA, the Asterisk Realtime Architecture,
415 ; please read realtime.txt and extconfig.txt in the /doc directory of the
418 ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
419 ; just like friends added from the config file only on a
420 ; as-needed basis? (yes|no)
422 ;rtsavesysname=yes ; Save systemname in realtime database at registration
425 ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
426 ; If set to yes, when a SIP UA registers successfully, the ip address,
427 ; the origination port, the registration period, and the username of
428 ; the UA will be set to database via realtime.
429 ; If not present, defaults to 'yes'.
430 ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
431 ; as if it had just registered? (yes|no|<seconds>)
432 ; If set to yes, when the registration expires, the friend will
433 ; vanish from the configuration until requested again. If set
434 ; to an integer, friends expire within this number of seconds
435 ; instead of the registration interval.
437 ;ignoreregexpire=yes ; Enabling this setting has two functions:
439 ; For non-realtime peers, when their registration expires, the
440 ; information will _not_ be removed from memory or the Asterisk database
441 ; if you attempt to place a call to the peer, the existing information
442 ; will be used in spite of it having expired
444 ; For realtime peers, when the peer is retrieved from realtime storage,
445 ; the registration information will be used regardless of whether
446 ; it has expired or not; if it expires while the realtime peer
447 ; is still in memory (due to caching or other reasons), the
448 ; information will not be removed from realtime storage
450 ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
451 ; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
452 ; domains, each of which can direct the call to a specific context if desired.
453 ; By default, all domains are accepted and sent to the default context or the
454 ; context associated with the user/peer placing the call.
455 ; Domains can be specified using:
456 ; domain=<domain>[,<context>]
458 ; domain=myasterisk.dom
459 ; domain=customer.com,customer-context
461 ; In addition, all the 'default' domains associated with a server should be
462 ; added if incoming request filtering is desired.
465 ; To disallow requests for domains not serviced by this server:
466 ; allowexternaldomains=no
468 ;domain=mydomain.tld,mydomain-incoming
469 ; Add domain and configure incoming context
470 ; for external calls to this domain
471 ;domain=1.2.3.4 ; Add IP address as local domain
472 ; You can have several "domain" settings
473 ;allowexternalinvites=no ; Disable INVITE and REFER to non-local domains
475 ;autodomain=yes ; Turn this on to have Asterisk add local host
476 ; name and local IP to domain list.
478 ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
479 ; non-peers, use your primary domain "identity"
480 ; for From: headers instead of just your IP
481 ; address. This is to be polite and
482 ; it may be a mandatory requirement for some
483 ; destinations which do not have a prior
484 ; account relationship with your server.
486 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
487 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
488 ; SIP channel. Defaults to "no". An enabled jitterbuffer will
489 ; be used only if the sending side can create and the receiving
490 ; side can not accept jitter. The SIP channel can accept jitter,
491 ; thus a jitterbuffer on the receive SIP side will be used only
492 ; if it is forced and enabled.
494 ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
495 ; channel. Defaults to "no".
497 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
499 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
500 ; resynchronized. Useful to improve the quality of the voice, with
501 ; big jumps in/broken timestamps, usually sent from exotic devices
502 ; and programs. Defaults to 1000.
504 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
505 ; channel. Two implementations are currently available - "fixed"
506 ; (with size always equals to jbmaxsize) and "adaptive" (with
507 ; variable size, actually the new jb of IAX2). Defaults to fixed.
509 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
510 ;-----------------------------------------------------------------------------------
513 ; Global credentials for outbound calls, i.e. when a proxy challenges your
514 ; Asterisk server for authentication. These credentials override
515 ; any credentials in peer/register definition if realm is matched.
517 ; This way, Asterisk can authenticate for outbound calls to other
518 ; realms. We match realm on the proxy challenge and pick an set of
519 ; credentials from this list
521 ; auth = <user>:<secret>@<realm>
522 ; auth = <user>#<md5secret>@<realm>
524 ;auth=mark:topsecret@digium.com
526 ; You may also add auth= statements to [peer] definitions
527 ; Peer auth= override all other authentication settings if we match on realm
529 ;------------------------------------------------------------------------------
530 ; Users and peers have different settings available. Friends have all settings,
531 ; since a friend is both a peer and a user
533 ; User config options: Peer configuration:
534 ; -------------------- -------------------
536 ; callingpres callingpres
540 ; md5secret md5secret
542 ; canreinvite canreinvite
544 ; callgroup callgroup
545 ; pickupgroup pickupgroup
550 ; trustrpid trustrpid
551 ; progressinband progressinband
552 ; promiscredir promiscredir
553 ; useclientcode useclientcode
554 ; accountcode accountcode
558 ; call-limit call-limit
559 ; allowoverlap allowoverlap
560 ; allowsubscribe allowsubscribe
561 ; allowtransfer allowtransfer
562 ; subscribecontext subscribecontext
563 ; videosupport videosupport
564 ; maxcallbitrate maxcallbitrate
565 ; rfc2833compensate mailbox
585 ; For incoming calls only. Example: FWD (Free World Dialup)
586 ; We match on IP address of the proxy for incoming calls
587 ; since we can not match on username (caller id)
593 ;type=peer ; we only want to call out, not be called
595 ;username=yourusername ; Authentication user for outbound proxies
596 ;fromuser=yourusername ; Many SIP providers require this!
597 ;fromdomain=provider.sip.domain
598 ;host=box.provider.com
599 ;usereqphone=yes ; This provider requires ";user=phone" on URI
600 ;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer
601 ; Call-limits will not be enforced on real-time peers,
602 ; since they are not stored in-memory
603 ;busy-level=2 ; Signal busy at 2 or more calls
604 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
605 ;port=80 ; The port number we want to connect to on the remote side
606 ; Also used as "defaultport" in combination with "defaultip" settings
608 ;--- sample definition for a provider
611 ;host=sip.provider1.com
612 ;username=4015552299 ; how your provider knows you
613 ;secret=youwillneverguessit
614 ;callbackextension=123 ; Register with this server and require calls coming back to this extension
616 ;------------------------------------------------------------------------------
617 ; Definitions of locally connected SIP devices
619 ; type = user a device that authenticates to us by "from" field to place calls
620 ; type = peer a device we place calls to or that calls us and we match by host
621 ; type = friend two configurations (peer+user) in one
623 ; For device names, we recommend using only a-z, numerics (0-9) and underscore
625 ; For local phones, type=friend works most of the time
627 ; If you have one-way audio, you probably have NAT problems.
628 ; If Asterisk is on a public IP, and the phone is inside of a NAT device
629 ; you will need to configure nat option for those phones.
630 ; Also, turn on qualify=yes to keep the nat session open
632 ; Because you might have a large number of similar sections, it is generally
633 ; convenient to use templates for the common parameters, and add them
634 ; the the various sections. Examples are below, and we can even leave
635 ; the templates uncommented as they will not harm:
637 [basic-options](!) ; a template
642 [natted-phone](!,basic-options) ; another template inheriting basic-options
647 [public-phone](!,basic-options) ; another template inheriting basic-options
651 [my-codecs](!) ; a template for my preferred codecs
659 [ulaw-phone](!) ; and another one for ulaw-only
663 ; and finally instantiate a few phones
665 ; [2133](natted-phone,my-codecs)
667 ; [2134](natted-phone,ulaw-phone)
668 ; secret = not_very_secret
669 ; [2136](public-phone,ulaw-phone)
670 ; secret = not_very_secret_either
674 ; Standard configurations not using templates look like this:
678 ;context=from-sip ; Where to start in the dialplan when this phone calls
679 ;callerid=John Doe <1234> ; Full caller ID, to override the phones config
680 ; on incoming calls to Asterisk
681 ;host=192.168.0.23 ; we have a static but private IP address
682 ; No registration allowed
683 ;nat=no ; there is not NAT between phone and Asterisk
684 ;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
685 ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
686 ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
687 ; from the phone to asterisk
688 ; 1 for the explicit peer, 1 for the explicit user,
689 ; remember that a friend equals 1 peer and 1 user in
691 ; This will affect your subscriptions as well.
692 ; There is no combined call counter for a "friend"
693 ; so there's currently no way in sip.conf to limit
694 ; to one inbound or outbound call per phone. Use
695 ; the group counters in the dial plan for that.
697 ;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
698 ;disallow=all ; need to disallow=all before we can use allow=
699 ;allow=ulaw ; Note: In user sections the order of codecs
700 ; listed with allow= does NOT matter!
702 ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
703 ;allow=g729 ; Pass-thru only unless g729 license obtained
704 ;callingpres=allowed_passed_screen ; Set caller ID presentation
705 ; See README.callingpres for more information
708 ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
709 ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
711 ;regexten=1234 ; When they register, create extension 1234
712 ;callerid="Jane Smith" <5678>
713 ;host=dynamic ; This device needs to register
714 ;nat=yes ; X-Lite is behind a NAT router
715 ;canreinvite=no ; Typically set to NO if behind NAT
717 ;allow=gsm ; GSM consumes far less bandwidth than ulaw
720 ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
721 ;registertrying=yes ; Send a 100 Trying when the device registers.
724 ;type=friend ; Friends place calls and receive calls
725 ;context=from-sip ; Context for incoming calls from this user
727 ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
728 ;language=de ; Use German prompts for this user
729 ;host=dynamic ; This peer register with us
730 ;dtmfmode=inband ; Choices are inband, rfc2833, or info
731 ;defaultip=192.168.0.59 ; IP used until peer registers
732 ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
733 ;subscribemwi=yes ; Only send notifications if this phone
734 ; subscribes for mailbox notification
735 ;vmexten=voicemail ; dialplan extension to reach mailbox
736 ; sets the Message-Account in the MWI notify message
737 ; defaults to global vmexten which defaults to "asterisk"
739 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
743 ;type=friend ; Friends place calls and receive calls
744 ;context=from-sip ; Context for incoming calls from this user
746 ;host=dynamic ; This peer register with us
747 ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
748 ;username=polly ; Username to use in INVITE until peer registers
749 ; Normally you do NOT need to set this parameter
751 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
752 ;progressinband=no ; Polycom phones don't work properly with "never"
759 ;insecure=port ; Allow matching of peer by IP address without
760 ; matching port number
761 ;insecure=invite ; Do not require authentication of incoming INVITEs
762 ;insecure=port,invite ; (both)
763 ;qualify=1000 ; Consider it down if it's 1 second to reply
764 ; Helps with NAT session
765 ; qualify=yes uses default value
767 ; Call group and Pickup group should be in the range from 0 to 63
769 ;callgroup=1,3-4 ; We are in caller groups 1,3,4
770 ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
771 ;defaultip=192.168.0.60 ; IP address to use if peer has not registered
772 ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
773 ;permit=192.168.0.60/255.255.255.0
778 ;qualify=200 ; Qualify peer is no more than 200ms away
779 ;nat=yes ; This phone may be natted
780 ; Send SIP and RTP to the IP address that packet is
781 ; received from instead of trusting SIP headers
782 ;host=dynamic ; This device registers with us
783 ;canreinvite=no ; Asterisk by default tries to redirect the
784 ; RTP media stream (audio) to go directly from
785 ; the caller to the callee. Some devices do not
786 ; support this (especially if one of them is
788 ;defaultip=192.168.0.4 ; IP address to use until registration
789 ;username=goran ; Username to use when calling this device before registration
790 ; Normally you do NOT need to set this parameter
791 ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
797 ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
798 ; You must have this turned on or DTMF reception will work improperly.
projects / asterisk/asterisk.git / blob