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 list peers Show all SIP peers (including friends)
17 ; sip list users Show all SIP users (including friends)
18 ; sip list registry Show status of hosts we register with
20 ; sip 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/README.tos 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.
65 ;maxexpiry=3600 ; Maximum allowed time of incoming registrations
66 ; and subscriptions (seconds)
67 ;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
68 ;defaultexpiry=120 ; Default length of incoming/outgoing registration
69 ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
71 ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
72 ;checkmwi=10 ; Default time between mailbox checks for peers
73 ;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
74 ; fully. Enable this option to not get error messages
75 ; when sending MWI to phones with this bug.
76 ;vmexten=voicemail ; dialplan extension to reach mailbox sets the
77 ; Message-Account in the MWI notify message
78 ; defaults to "asterisk"
79 ;disallow=all ; First disallow all codecs
80 ;allow=ulaw ; Allow codecs in order of preference
81 ;allow=ilbc ; see doc/rtp-packetization for framing options
83 ; This option specifies a preference for which music on hold class this channel
84 ; should listen to when put on hold if the music class has not been set on the
85 ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
86 ; channel putting this one on hold did not suggest a music class.
88 ; This option may be specified globally, or on a per-user or per-peer basis.
92 ; This option specifies which music on hold class to suggest to the peer channel
93 ; when this channel places the peer on hold. It may be specified globally or on
94 ; a per-user or per-peer basis.
98 ;language=en ; Default language setting for all users/peers
99 ; This may also be set for individual users/peers
100 ;relaxdtmf=yes ; Relax dtmf handling
101 ;trustrpid = no ; If Remote-Party-ID should be trusted
102 ;sendrpid = yes ; If Remote-Party-ID should be sent
103 ;progressinband=never ; If we should generate in-band ringing always
104 ; use 'never' to never use in-band signalling, even in cases
105 ; where some buggy devices might not render it
106 ; Valid values: yes, no, never Default: never
107 ;useragent=Asterisk PBX ; Allows you to change the user agent string
108 ; The default user agent string also contains the Asterisk
109 ; version. If you don't want to expose this, change the
111 ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
112 ; Note that promiscredir when redirects are made to the
113 ; local system will cause loops since Asterisk is incapable
114 ; of performing a "hairpin" call.
115 ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
116 ; a valid phone number
117 ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
119 ; info : SIP INFO messages
120 ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
121 ; auto : Use rfc2833 if offered, inband otherwise
123 ;compactheaders = yes ; send compact sip headers.
125 ;videosupport=yes ; Turn on support for SIP video. You need to turn this on
126 ; in the this section to get any video support at all.
127 ; You can turn it off on a per peer basis if the general
128 ; video support is enabled, but you can't enable it for
129 ; one peer only without enabling in the general section.
130 ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
131 ; Videosupport and maxcallbitrate is settable
132 ; for peers and users as well
133 ;callevents=no ; generate manager events when sip ua
134 ; performs events (e.g. hold)
135 ;limitpeersonly=no ; Apply all call limits ("limit=") only to peers, never
136 ; to users. This improves handling of call limits
137 ; and device states in certain situations. The user part
138 ; of a type=friend will still be affected by the call
139 ; limit, but Asterisk will only use one object for
140 ; counting the simultaneous calls.
141 ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
142 ; for any reason, always reject with '401 Unauthorized'
143 ; instead of letting the requester know whether there was
144 ; a matching user or peer for their request
146 ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
147 ; order instead of RFC3551 packing order (this is required
148 ; for Sipura and Grandstream ATAs, among others). This is
149 ; contrary to the RFC3551 specification, the peer _should_
150 ; be negotiating AAL2-G726-32 instead :-(
153 ; If regcontext is specified, Asterisk will dynamically create and destroy a
154 ; NoOp priority 1 extension for a given peer who registers or unregisters with
155 ; us and have a "regexten=" configuration item.
156 ; Multiple contexts may be specified by separating them with '&'. The
157 ; actual extension is the 'regexten' parameter of the registering peer or its
158 ; name if 'regexten' is not provided. If more than one context is provided,
159 ; the context must be specified within regexten by appending the desired
160 ; context after '@'. More than one regexten may be supplied if they are
161 ; separated by '&'. Patterns may be used in regexten.
163 ;regcontext=sipregistrations
165 ;--------------------------- RTP timers ----------------------------------------------------
166 ; These timers are currently used for both audio and video streams. The RTP timeouts
167 ; are only applied to the audio channel.
168 ; The settings are settable in the global section as well as per device
170 ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
171 ; on the audio channel
172 ; when we're not on hold. This is to be able to hangup
173 ; a call in the case of a phone disappearing from the net,
174 ; like a powerloss or grandma tripping over a cable.
175 ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
176 ; on the audio channel
177 ; when we're on hold (must be > rtptimeout)
178 ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
179 ; (default is off - zero)
180 ;--------------------------- SIP DEBUGGING ---------------------------------------------------
181 ;sipdebug = yes ; Turn on SIP debugging by default, from
182 ; the moment the channel loads this configuration
183 ;recordhistory=yes ; Record SIP history by default
184 ; (see sip history / sip no history)
185 ;dumphistory=yes ; Dump SIP history at end of SIP dialogue
186 ; SIP history is output to the DEBUG logging channel
189 ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
190 ; You can subscribe to the status of extensions with a "hint" priority
191 ; (See extensions.conf.sample for examples)
192 ; chan_sip support two major formats for notifications: dialog-info and SIMPLE
194 ; You will get more detailed reports (busy etc) if you have a call limit set
195 ; for a device. When the call limit is filled, we will indicate busy. Note that
196 ; you need at least 2 in order to be able to do attended transfers.
198 ; If you set the busy-limit in addition to the call limit, we will indicate busy
199 ; when we have a number of calls that matches busy-limit, but still allow calls
200 ; up to the call-limit. This allows for transfers while still having blinking
201 ; lamps and queues understanding that a device is busy.
203 ; For queues, you will need this level of detail in status reporting, regardless
204 ; if you use SIP subscriptions. Queues and manager use the same internal interface
205 ; for reading status information.
207 ; Note: Subscriptions does not work if you have a realtime dialplan and use the
210 ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
211 ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
212 ; Useful to limit subscriptions to local extensions
213 ; Settable per peer/user also
214 ;notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
215 ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
216 ; Turning on notifyringing and notifyhold will add a lot
217 ; more database transactions if you are using realtime.
218 ;limitonpeer = yes ; Apply call limits on peers only. This will improve
219 ; status notification when you are using type=friend
220 ; Inbound calls, that really apply to the user part
221 ; of a friend will now be added to and compared with
222 ; the peer limit instead of applying two call limits,
223 ; one for the peer and one for the user.
225 ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
227 ; This setting is available in the [general] section as well as in device configurations.
228 ; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
229 ; both parties have T38 support enabled in their Asterisk configuration
230 ; This has to be enabled in the general section for all devices to work. You can then
231 ; disable it on a per device basis.
233 ; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.
235 ; t38pt_udptl = yes ; Default false
237 ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
238 ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
239 ; Format for the register statement is:
240 ; register => user[:secret[:authuser]]@host[:port][/extension]
242 ; If no extension is given, the 's' extension is used. The extension needs to
243 ; be defined in extensions.conf to be able to accept calls from this SIP proxy
246 ; host is either a host name defined in DNS or the name of a section defined
249 ; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
250 ; this is equivalent to having the following line in the general section:
252 ; register => username:secret@host/callbackextension
254 ; and more readable because you don't have to write the parameters in two places
255 ; (note that the "port" is ignored - this is a bug that should be fixed).
259 ;register => 1234:password@mysipprovider.com
261 ; This will pass incoming calls to the 's' extension
264 ;register => 2345:password@sip_proxy/1234
266 ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
267 ; connect to local extension 1234 in extensions.conf, default context,
268 ; unless you configure a [sip_proxy] section below, and configure a
270 ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
271 ; Tip 2: Use separate type=peer and type=user sections for SIP providers
272 ; (instead of type=friend) if you have calls in both directions
274 ;registertimeout=20 ; retry registration calls every 20 seconds (default)
275 ;registerattempts=10 ; Number of registration attempts before we give up
276 ; 0 = continue forever, hammering the other server
277 ; until it accepts the registration
278 ; Default is 0 tries, continue forever
280 ;----------------------------------------- NAT SUPPORT ------------------------
281 ; The externip, externhost and localnet settings are used if you use Asterisk
282 ; behind a NAT device to communicate with services on the outside.
284 ;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP
285 ; messages if we're behind a NAT
287 ; The externip and localnet is used
288 ; when registering and communicating with other proxies
289 ; that we're registered with
290 ;externhost=foo.dyndns.net ; Alternatively you can specify an
291 ; external host, and Asterisk will
292 ; perform DNS queries periodically. Not
293 ; recommended for production
294 ; environments! Use externip instead
295 ;externrefresh=10 ; How often to refresh externhost if
297 ; You may add multiple local networks. A reasonable
298 ; set of defaults are:
299 ;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
300 ;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
301 ;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
302 ;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
304 ; The nat= setting is used when Asterisk is on a public IP, communicating with
305 ; devices hidden behind a NAT device (broadband router). If you have one-way
306 ; audio problems, you usually have problems with your NAT configuration or your
307 ; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
308 ; ports for incoming audio in rtp.conf
310 ;nat=no ; Global NAT settings (Affects all peers and users)
311 ; yes = Always ignore info and assume NAT
312 ; no = Use NAT mode only according to RFC3581 (;rport)
313 ; never = Never attempt NAT mode or RFC3581 support
314 ; route = Assume NAT, don't send rport
315 ; (work around more UNIDEN bugs)
317 ;----------------------------------- MEDIA HANDLING --------------------------------
318 ; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
319 ; no reason for Asterisk to stay in the media path, the media will be redirected.
320 ; This does not really work with in the case where Asterisk is outside and have
321 ; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
323 ;canreinvite=yes ; Asterisk by default tries to redirect the
324 ; RTP media stream (audio) to go directly from
325 ; the caller to the callee. Some devices do not
326 ; support this (especially if one of them is behind a NAT).
327 ; The default setting is YES. If you have all clients
328 ; behind a NAT, or for some other reason wants Asterisk to
329 ; stay in the audio path, you may want to turn this off.
331 ; This setting also affect direct RTP
332 ; at call setup (a new feature in 1.4 - setting up the
333 ; call directly between the endpoints instead of sending
336 ;canreinvite=nonat ; An additional option is to allow media path redirection
337 ; (reinvite) but only when the peer where the media is being
338 ; sent is known to not be behind a NAT (as the RTP core can
339 ; determine it based on the apparent IP address the media
342 ;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
343 ; instead of INVITE. This can be combined with 'nonat', as
344 ; 'canreinvite=update,nonat'. It implies 'yes'.
346 ;----------------------------------------- REALTIME SUPPORT ------------------------
347 ; For additional information on ARA, the Asterisk Realtime Architecture,
348 ; please read realtime.txt and extconfig.txt in the /doc directory of the
351 ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
352 ; just like friends added from the config file only on a
353 ; as-needed basis? (yes|no)
355 ;rtsavesysname=yes ; Save systemname in realtime database at registration
358 ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
359 ; If set to yes, when a SIP UA registers successfully, the ip address,
360 ; the origination port, the registration period, and the username of
361 ; the UA will be set to database via realtime.
362 ; If not present, defaults to 'yes'.
363 ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
364 ; as if it had just registered? (yes|no|<seconds>)
365 ; If set to yes, when the registration expires, the friend will
366 ; vanish from the configuration until requested again. If set
367 ; to an integer, friends expire within this number of seconds
368 ; instead of the registration interval.
370 ;ignoreregexpire=yes ; Enabling this setting has two functions:
372 ; For non-realtime peers, when their registration expires, the
373 ; information will _not_ be removed from memory or the Asterisk database
374 ; if you attempt to place a call to the peer, the existing information
375 ; will be used in spite of it having expired
377 ; For realtime peers, when the peer is retrieved from realtime storage,
378 ; the registration information will be used regardless of whether
379 ; it has expired or not; if it expires while the realtime peer
380 ; is still in memory (due to caching or other reasons), the
381 ; information will not be removed from realtime storage
383 ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
384 ; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
385 ; domains, each of which can direct the call to a specific context if desired.
386 ; By default, all domains are accepted and sent to the default context or the
387 ; context associated with the user/peer placing the call.
388 ; Domains can be specified using:
389 ; domain=<domain>[,<context>]
391 ; domain=myasterisk.dom
392 ; domain=customer.com,customer-context
394 ; In addition, all the 'default' domains associated with a server should be
395 ; added if incoming request filtering is desired.
398 ; To disallow requests for domains not serviced by this server:
399 ; allowexternaldomains=no
401 ;domain=mydomain.tld,mydomain-incoming
402 ; Add domain and configure incoming context
403 ; for external calls to this domain
404 ;domain=1.2.3.4 ; Add IP address as local domain
405 ; You can have several "domain" settings
406 ;allowexternalinvites=no ; Disable INVITE and REFER to non-local domains
408 ;autodomain=yes ; Turn this on to have Asterisk add local host
409 ; name and local IP to domain list.
411 ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
412 ; non-peers, use your primary domain "identity"
413 ; for From: headers instead of just your IP
414 ; address. This is to be polite and
415 ; it may be a mandatory requirement for some
416 ; destinations which do not have a prior
417 ; account relationship with your server.
419 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
420 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
421 ; SIP channel. Defaults to "no". An enabled jitterbuffer will
422 ; be used only if the sending side can create and the receiving
423 ; side can not accept jitter. The SIP channel can accept jitter,
424 ; thus a jitterbuffer on the receive SIP side will be used only
425 ; if it is forced and enabled.
427 ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
428 ; channel. Defaults to "no".
430 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
432 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
433 ; resynchronized. Useful to improve the quality of the voice, with
434 ; big jumps in/broken timestamps, usually sent from exotic devices
435 ; and programs. Defaults to 1000.
437 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
438 ; channel. Two implementations are currently available - "fixed"
439 ; (with size always equals to jbmaxsize) and "adaptive" (with
440 ; variable size, actually the new jb of IAX2). Defaults to fixed.
442 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
443 ;-----------------------------------------------------------------------------------
446 ; Global credentials for outbound calls, i.e. when a proxy challenges your
447 ; Asterisk server for authentication. These credentials override
448 ; any credentials in peer/register definition if realm is matched.
450 ; This way, Asterisk can authenticate for outbound calls to other
451 ; realms. We match realm on the proxy challenge and pick an set of
452 ; credentials from this list
454 ; auth = <user>:<secret>@<realm>
455 ; auth = <user>#<md5secret>@<realm>
457 ;auth=mark:topsecret@digium.com
459 ; You may also add auth= statements to [peer] definitions
460 ; Peer auth= override all other authentication settings if we match on realm
462 ;------------------------------------------------------------------------------
463 ; Users and peers have different settings available. Friends have all settings,
464 ; since a friend is both a peer and a user
466 ; User config options: Peer configuration:
467 ; -------------------- -------------------
469 ; callingpres callingpres
473 ; md5secret md5secret
475 ; canreinvite canreinvite
477 ; callgroup callgroup
478 ; pickupgroup pickupgroup
483 ; trustrpid trustrpid
484 ; progressinband progressinband
485 ; promiscredir promiscredir
486 ; useclientcode useclientcode
487 ; accountcode accountcode
491 ; call-limit call-limit
492 ; allowoverlap allowoverlap
493 ; allowsubscribe allowsubscribe
494 ; allowtransfer allowtransfer
495 ; subscribecontext subscribecontext
496 ; videosupport videosupport
497 ; maxcallbitrate maxcallbitrate
498 ; rfc2833compensate mailbox
517 ; For incoming calls only. Example: FWD (Free World Dialup)
518 ; We match on IP address of the proxy for incoming calls
519 ; since we can not match on username (caller id)
525 ;type=peer ; we only want to call out, not be called
527 ;username=yourusername ; Authentication user for outbound proxies
528 ;fromuser=yourusername ; Many SIP providers require this!
529 ;fromdomain=provider.sip.domain
530 ;host=box.provider.com
531 ;usereqphone=yes ; This provider requires ";user=phone" on URI
532 ;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer
533 ;busy-limit=2 ; Signal busy at 2 or more calls
534 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
535 ; Call-limits will not be enforced on real-time peers,
536 ; since they are not stored in-memory
537 ;port=80 ; The port number we want to connect to on the remote side
538 ; Also used as "defaultport" in combination with "defaultip" settings
540 ;--- sample definition for a provider
543 ;host=sip.provider1.com
544 ;username=4015552299 ; how your provider knows you
545 ;secret=youwillneverguessit
546 ;callbackextension=123 ; Register with this server and require calls coming back to this extension
548 ;------------------------------------------------------------------------------
549 ; Definitions of locally connected SIP devices
551 ; type = user a device that authenticates to us by "from" field to place calls
552 ; type = peer a device we place calls to or that calls us and we match by host
553 ; type = friend two configurations (peer+user) in one
555 ; For device names, we recommend using only a-z, numerics (0-9) and underscore
557 ; For local phones, type=friend works most of the time
559 ; If you have one-way audio, you probably have NAT problems.
560 ; If Asterisk is on a public IP, and the phone is inside of a NAT device
561 ; you will need to configure nat option for those phones.
562 ; Also, turn on qualify=yes to keep the nat session open
564 ; Because you might have a large number of similar sections, it is generally
565 ; convenient to use templates for the common parameters, and add them
566 ; the the various sections. Examples are below, and we can even leave
567 ; the templates uncommented as they will not harm:
569 [basic-options](!) ; a template
574 [natted-phone](!,basic-options) ; another template inheriting basic-options
579 [public-phone](!,basic-options) ; another template inheriting basic-options
583 [my-codecs](!) ; a template for my preferred codecs
591 [ulaw-phone](!) ; and another one for ulaw-only
595 ; and finally instantiate a few phones
597 ; [2133](natted-phone,my-codecs)
599 ; [2134](natted-phone,ulaw-hone)
600 ; secret = not_very_secret
601 ; [2136](public-phone,ulaw-hone)
602 ; secret = not_very_secret_either
606 ; Standard configurations not using templates look like this:
610 ;context=from-sip ; Where to start in the dialplan when this phone calls
611 ;callerid=John Doe <1234> ; Full caller ID, to override the phones config
612 ; on incoming calls to Asterisk
613 ;host=192.168.0.23 ; we have a static but private IP address
614 ; No registration allowed
615 ;nat=no ; there is not NAT between phone and Asterisk
616 ;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
617 ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
618 ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
619 ; from the phone to asterisk
620 ; 1 for the explicit peer, 1 for the explicit user,
621 ; remember that a friend equals 1 peer and 1 user in
623 ; This will affect your subscriptions as well.
624 ; There is no combined call counter for a "friend"
625 ; so there's currently no way in sip.conf to limit
626 ; to one inbound or outbound call per phone. Use
627 ; the group counters in the dial plan for that.
629 ;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
630 ;disallow=all ; need to disallow=all before we can use allow=
631 ;allow=ulaw ; Note: In user sections the order of codecs
632 ; listed with allow= does NOT matter!
634 ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
635 ;allow=g729 ; Pass-thru only unless g729 license obtained
636 ;callingpres=allowed_passed_screen ; Set caller ID presentation
637 ; See README.callingpres for more information
641 ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
642 ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
644 ;regexten=1234 ; When they register, create extension 1234
645 ;callerid="Jane Smith" <5678>
646 ;host=dynamic ; This device needs to register
647 ;nat=yes ; X-Lite is behind a NAT router
648 ;canreinvite=no ; Typically set to NO if behind NAT
650 ;allow=gsm ; GSM consumes far less bandwidth than ulaw
653 ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
657 ;type=friend ; Friends place calls and receive calls
658 ;context=from-sip ; Context for incoming calls from this user
660 ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
661 ;language=de ; Use German prompts for this user
662 ;host=dynamic ; This peer register with us
663 ;dtmfmode=inband ; Choices are inband, rfc2833, or info
664 ;defaultip=192.168.0.59 ; IP used until peer registers
665 ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
666 ;subscribemwi=yes ; Only send notifications if this phone
667 ; subscribes for mailbox notification
668 ;vmexten=voicemail ; dialplan extension to reach mailbox
669 ; sets the Message-Account in the MWI notify message
670 ; defaults to global vmexten which defaults to "asterisk"
672 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
676 ;type=friend ; Friends place calls and receive calls
677 ;context=from-sip ; Context for incoming calls from this user
679 ;host=dynamic ; This peer register with us
680 ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
681 ;username=polly ; Username to use in INVITE until peer registers
682 ; Normally you do NOT need to set this parameter
684 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
685 ;progressinband=no ; Polycom phones don't work properly with "never"
692 ;insecure=port ; Allow matching of peer by IP address without
693 ; matching port number
694 ;insecure=invite ; Do not require authentication of incoming INVITEs
695 ;insecure=port,invite ; (both)
696 ;qualify=1000 ; Consider it down if it's 1 second to reply
697 ; Helps with NAT session
698 ; qualify=yes uses default value
700 ; Call group and Pickup group should be in the range from 0 to 63
702 ;callgroup=1,3-4 ; We are in caller groups 1,3,4
703 ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
704 ;defaultip=192.168.0.60 ; IP address to use if peer has not registered
705 ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
706 ;permit=192.168.0.60/255.255.255.0
711 ;qualify=200 ; Qualify peer is no more than 200ms away
712 ;nat=yes ; This phone may be natted
713 ; Send SIP and RTP to the IP address that packet is
714 ; received from instead of trusting SIP headers
715 ;host=dynamic ; This device registers with us
716 ;canreinvite=no ; Asterisk by default tries to redirect the
717 ; RTP media stream (audio) to go directly from
718 ; the caller to the callee. Some devices do not
719 ; support this (especially if one of them is
721 ;defaultip=192.168.0.4 ; IP address to use until registration
722 ;username=goran ; Username to use when calling this device before registration
723 ; Normally you do NOT need to set this parameter
724 ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
730 ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
731 ; You must have this turned on or DTMF reception will work improperly.
projects / asterisk/asterisk.git / blob