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1 ;
2 ; SIP Configuration example for Asterisk
3 ;
4 ; Syntax for specifying a SIP device in extensions.conf is
5 ; SIP/devicename where devicename is defined in a section below.
6 ;
7 ; You may also use
8 ; SIP/username@domain to call any SIP user on the Internet
9 ; (Don't forget to enable DNS SRV records if you want to use this)
10 ;
11 ; If you define a SIP proxy as a peer below, you may call
12 ; SIP/proxyhostname/user or SIP/user@proxyhostname
13 ; where the proxyhostname is defined in a section below
14 ;
15 ; Useful CLI commands to check peers/users:
16 ; sip show peers Show all SIP peers (including friends)
17 ; sip show users Show all SIP users (including friends)
18 ; sip show registry Show status of hosts we register with
19 ;
20 ; sip debug Show all SIP messages
21 ;
22 ; reload chan_sip.so Reload configuration file
23 ; Active SIP peers will not be reconfigured
24 ;
26 [general]
27 context=default ; Default context for incoming calls
28 ;allowguest=no ; Allow or reject guest calls (default is yes)
29 allowoverlap=no ; Disable overlap dialing support. (Default is yes)
30 ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
31 ; Default is enabled
32 ;realm=mydomain.tld ; Realm for digest authentication
33 ; defaults to "asterisk". If you set a system name in
34 ; asterisk.conf, it defaults to that system name
35 ; Realms MUST be globally unique according to RFC 3261
36 ; Set this to your host name or domain name
37 bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
38 ; bindport is the local UDP port that Asterisk will listen on
39 bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
40 srvlookup=yes ; Enable DNS SRV lookups on outbound calls
41 ; Note: Asterisk only uses the first host
42 ; in SRV records
43 ; Disabling DNS SRV lookups disables the
44 ; ability to place SIP calls based on domain
45 ; names to some other SIP users on the Internet
47 ;domain=mydomain.tld ; Set default domain for this host
48 ; If configured, Asterisk will only allow
49 ; INVITE and REFER to non-local domains
50 ; Use "sip show domains" to list local domains
51 ;pedantic=yes ; Enable checking of tags in headers,
52 ; international character conversions in URIs
53 ; and multiline formatted headers for strict
54 ; SIP compatibility (defaults to "no")
56 ; See doc/ip-tos.txt for a description of these parameters.
57 ;tos_sip=cs3 ; Sets TOS for SIP packets.
58 ;tos_audio=ef ; Sets TOS for RTP audio packets.
59 ;tos_video=af41 ; Sets TOS for RTP video packets.
61 ;maxexpiry=3600 ; Maximum allowed time of incoming registrations
62 ; and subscriptions (seconds)
63 ;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
64 ;defaultexpiry=120 ; Default length of incoming/outgoing registration
65 ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
66 ; Defaults to 100 ms
67 ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
68 ;checkmwi=10 ; Default time between mailbox checks for peers
69 ;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
70 ; fully. Enable this option to not get error messages
71 ; when sending MWI to phones with this bug.
72 ;vmexten=voicemail ; dialplan extension to reach mailbox sets the
73 ; Message-Account in the MWI notify message
74 ; defaults to "asterisk"
75 ;disallow=all ; First disallow all codecs
76 ;allow=ulaw ; Allow codecs in order of preference
77 ;allow=ilbc ; see doc/rtp-packetization for framing options
78 ;
79 ; This option specifies a preference for which music on hold class this channel
80 ; should listen to when put on hold if the music class has not been set on the
81 ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
82 ; channel putting this one on hold did not suggest a music class.
83 ;
84 ; This option may be specified globally, or on a per-user or per-peer basis.
85 ;
86 ;mohinterpret=default
87 ;
88 ; This option specifies which music on hold class to suggest to the peer channel
89 ; when this channel places the peer on hold. It may be specified globally or on
90 ; a per-user or per-peer basis.
91 ;
92 ;mohsuggest=default
93 ;
94 ;language=en ; Default language setting for all users/peers
95 ; This may also be set for individual users/peers
96 ;relaxdtmf=yes ; Relax dtmf handling
97 ;trustrpid = no ; If Remote-Party-ID should be trusted
98 ;sendrpid = yes ; If Remote-Party-ID should be sent
99 ;progressinband=never ; If we should generate in-band ringing always
100 ; use 'never' to never use in-band signalling, even in cases
101 ; where some buggy devices might not render it
102 ; Valid values: yes, no, never Default: never
103 ;useragent=Asterisk PBX ; Allows you to change the user agent string
104 ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
105 ; Note that promiscredir when redirects are made to the
106 ; local system will cause loops since Asterisk is incapable
107 ; of performing a "hairpin" call.
108 ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
109 ; a valid phone number
110 ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
111 ; Other options:
112 ; info : SIP INFO messages
113 ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
114 ; auto : Use rfc2833 if offered, inband otherwise
116 ;compactheaders = yes ; send compact sip headers.
117 ;
118 ;videosupport=yes ; Turn on support for SIP video. You need to turn this on
119 ; in the this section to get any video support at all.
120 ; You can turn it off on a per peer basis if the general
121 ; video support is enabled, but you can't enable it for
122 ; one peer only without enabling in the general section.
123 ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
124 ; Videosupport and maxcallbitrate is settable
125 ; for peers and users as well
126 ;callevents=no ; generate manager events when sip ua
127 ; performs events (e.g. hold)
128 ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
129 ; for any reason, always reject with '401 Unauthorized'
130 ; instead of letting the requester know whether there was
131 ; a matching user or peer for their request
133 ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
134 ; order instead of RFC3551 packing order (this is required
135 ; for Sipura and Grandstream ATAs, among others). This is
136 ; contrary to the RFC3551 specification, the peer _should_
137 ; be negotiating AAL2-G726-32 instead :-(
139 ;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
140 ; your localnet setting. Unless you have some sort of strange network
141 ; setup you will not need to enable this.
143 ;
144 ; If regcontext is specified, Asterisk will dynamically create and destroy a
145 ; NoOp priority 1 extension for a given peer who registers or unregisters with
146 ; us and have a "regexten=" configuration item.
147 ; Multiple contexts may be specified by separating them with '&'. The
148 ; actual extension is the 'regexten' parameter of the registering peer or its
149 ; name if 'regexten' is not provided. If more than one context is provided,
150 ; the context must be specified within regexten by appending the desired
151 ; context after '@'. More than one regexten may be supplied if they are
152 ; separated by '&'. Patterns may be used in regexten.
153 ;
154 ;regcontext=sipregistrations
155 ;
156 ;--------------------------- RTP timers ----------------------------------------------------
157 ; These timers are currently used for both audio and video streams. The RTP timeouts
158 ; are only applied to the audio channel.
159 ; The settings are settable in the global section as well as per device
160 ;
161 ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
162 ; on the audio channel
163 ; when we're not on hold. This is to be able to hangup
164 ; a call in the case of a phone disappearing from the net,
165 ; like a powerloss or grandma tripping over a cable.
166 ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
167 ; on the audio channel
168 ; when we're on hold (must be > rtptimeout)
169 ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
170 ; (default is off - zero)
171 ;--------------------------- SIP DEBUGGING ---------------------------------------------------
172 ;sipdebug = yes ; Turn on SIP debugging by default, from
173 ; the moment the channel loads this configuration
174 ;recordhistory=yes ; Record SIP history by default
175 ; (see sip history / sip no history)
176 ;dumphistory=yes ; Dump SIP history at end of SIP dialogue
177 ; SIP history is output to the DEBUG logging channel
180 ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
181 ; You can subscribe to the status of extensions with a "hint" priority
182 ; (See extensions.conf.sample for examples)
183 ; chan_sip support two major formats for notifications: dialog-info and SIMPLE
184 ;
185 ; You will get more detailed reports (busy etc) if you have a call limit set
186 ; for a device. When the call limit is filled, we will indicate busy. Note that
187 ; you need at least 2 in order to be able to do attended transfers.
188 ;
189 ; For queues, you will need this level of detail in status reporting, regardless
190 ; if you use SIP subscriptions. Queues and manager use the same internal interface
191 ; for reading status information.
192 ;
193 ; Note: Subscriptions does not work if you have a realtime dialplan and use the
194 ; realtime switch.
195 ;
196 ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
197 ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
198 ; Useful to limit subscriptions to local extensions
199 ; Settable per peer/user also
200 ;notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
201 ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
202 ; Turning on notifyringing and notifyhold will add a lot
203 ; more database transactions if you are using realtime.
204 ;limitonpeers = yes ; Apply call limits on peers only. This will improve
205 ; status notification when you are using type=friend
206 ; Inbound calls, that really apply to the user part
207 ; of a friend will now be added to and compared with
208 ; the peer limit instead of applying two call limits,
209 ; one for the peer and one for the user.
210 ; "sip show inuse" will only show active calls on
211 ; the peer side of a "type=friend" object if this
212 ; setting is turned on.
214 ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
215 ;
216 ; This setting is available in the [general] section as well as in device configurations.
217 ; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
218 ; both parties have T38 support enabled in their Asterisk configuration
219 ; This has to be enabled in the general section for all devices to work. You can then
220 ; disable it on a per device basis.
221 ;
222 ; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.
223 ;
224 ; t38pt_udptl = yes ; Default false
225 ;
226 ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
227 ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
228 ; Format for the register statement is:
229 ; register => user[:secret[:authuser]]@host[:port][/extension]
230 ;
231 ; If no extension is given, the 's' extension is used. The extension needs to
232 ; be defined in extensions.conf to be able to accept calls from this SIP proxy
233 ; (provider).
234 ;
235 ; host is either a host name defined in DNS or the name of a section defined
236 ; below.
237 ;
238 ; Examples:
239 ;
240 ;register => 1234:password@mysipprovider.com
241 ;
242 ; This will pass incoming calls to the 's' extension
243 ;
244 ;
245 ;register => 2345:password@sip_proxy/1234
246 ;
247 ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
248 ; connect to local extension 1234 in extensions.conf, default context,
249 ; unless you configure a [sip_proxy] section below, and configure a
250 ; context.
251 ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
252 ; Tip 2: Use separate type=peer and type=user sections for SIP providers
253 ; (instead of type=friend) if you have calls in both directions
255 ;registertimeout=20 ; retry registration calls every 20 seconds (default)
256 ;registerattempts=10 ; Number of registration attempts before we give up
257 ; 0 = continue forever, hammering the other server
258 ; until it accepts the registration
259 ; Default is 0 tries, continue forever
261 ;----------------------------------------- NAT SUPPORT ------------------------
262 ; The externip, externhost and localnet settings are used if you use Asterisk
263 ; behind a NAT device to communicate with services on the outside.
265 ;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP
266 ; messages if we're behind a NAT
268 ; The externip and localnet is used
269 ; when registering and communicating with other proxies
270 ; that we're registered with
271 ;externhost=foo.dyndns.net ; Alternatively you can specify an
272 ; external host, and Asterisk will
273 ; perform DNS queries periodically. Not
274 ; recommended for production
275 ; environments! Use externip instead
276 ;externrefresh=10 ; How often to refresh externhost if
277 ; used
278 ; You may add multiple local networks. A reasonable
279 ; set of defaults are:
280 ;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
281 ;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
282 ;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
283 ;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
285 ; The nat= setting is used when Asterisk is on a public IP, communicating with
286 ; devices hidden behind a NAT device (broadband router). If you have one-way
287 ; audio problems, you usually have problems with your NAT configuration or your
288 ; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
289 ; ports for incoming audio in rtp.conf
290 ;
291 ;nat=no ; Global NAT settings (Affects all peers and users)
292 ; yes = Always ignore info and assume NAT
293 ; no = Use NAT mode only according to RFC3581 (;rport)
294 ; never = Never attempt NAT mode or RFC3581 support
295 ; route = Assume NAT, don't send rport
296 ; (work around more UNIDEN bugs)
298 ;----------------------------------- MEDIA HANDLING --------------------------------
299 ; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
300 ; no reason for Asterisk to stay in the media path, the media will be redirected.
301 ; This does not really work with in the case where Asterisk is outside and have
302 ; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
303 ;
304 ;canreinvite=yes ; Asterisk by default tries to redirect the
305 ; RTP media stream (audio) to go directly from
306 ; the caller to the callee. Some devices do not
307 ; support this (especially if one of them is behind a NAT).
308 ; The default setting is YES. If you have all clients
309 ; behind a NAT, or for some other reason wants Asterisk to
310 ; stay in the audio path, you may want to turn this off.
312 ; In Asterisk 1.4 this setting also affect direct RTP
313 ; at call setup (a new feature in 1.4 - setting up the
314 ; call directly between the endpoints instead of sending
315 ; a re-INVITE).
317 ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
318 ; the call directly with media peer-2-peer without re-invites.
319 ; Will not work for video and cases where the callee sends
320 ; RTP payloads and fmtp headers in the 200 OK that does not match the
321 ; callers INVITE. This will also fail if canreinvite is enabled when
322 ; the device is actually behind NAT.
324 ;canreinvite=nonat ; An additional option is to allow media path redirection
325 ; (reinvite) but only when the peer where the media is being
326 ; sent is known to not be behind a NAT (as the RTP core can
327 ; determine it based on the apparent IP address the media
328 ; arrives from).
330 ;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
331 ; instead of INVITE. This can be combined with 'nonat', as
332 ; 'canreinvite=update,nonat'. It implies 'yes'.
334 ;----------------------------------------- REALTIME SUPPORT ------------------------
335 ; For additional information on ARA, the Asterisk Realtime Architecture,
336 ; please read realtime.txt and extconfig.txt in the /doc directory of the
337 ; source code.
338 ;
339 ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
340 ; just like friends added from the config file only on a
341 ; as-needed basis? (yes|no)
343 ;rtsavesysname=yes ; Save systemname in realtime database at registration
344 ; Default= no
346 ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
347 ; If set to yes, when a SIP UA registers successfully, the ip address,
348 ; the origination port, the registration period, and the username of
349 ; the UA will be set to database via realtime.
350 ; If not present, defaults to 'yes'.
351 ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
352 ; as if it had just registered? (yes|no|<seconds>)
353 ; If set to yes, when the registration expires, the friend will
354 ; vanish from the configuration until requested again. If set
355 ; to an integer, friends expire within this number of seconds
356 ; instead of the registration interval.
358 ;ignoreregexpire=yes ; Enabling this setting has two functions:
359 ;
360 ; For non-realtime peers, when their registration expires, the
361 ; information will _not_ be removed from memory or the Asterisk database
362 ; if you attempt to place a call to the peer, the existing information
363 ; will be used in spite of it having expired
364 ;
365 ; For realtime peers, when the peer is retrieved from realtime storage,
366 ; the registration information will be used regardless of whether
367 ; it has expired or not; if it expires while the realtime peer
368 ; is still in memory (due to caching or other reasons), the
369 ; information will not be removed from realtime storage
371 ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
372 ; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
373 ; domains, each of which can direct the call to a specific context if desired.
374 ; By default, all domains are accepted and sent to the default context or the
375 ; context associated with the user/peer placing the call.
376 ; Domains can be specified using:
377 ; domain=<domain>[,<context>]
378 ; Examples:
379 ; domain=myasterisk.dom
380 ; domain=customer.com,customer-context
381 ;
382 ; In addition, all the 'default' domains associated with a server should be
383 ; added if incoming request filtering is desired.
384 ; autodomain=yes
385 ;
386 ; To disallow requests for domains not serviced by this server:
387 ; allowexternaldomains=no
389 ;domain=mydomain.tld,mydomain-incoming
390 ; Add domain and configure incoming context
391 ; for external calls to this domain
392 ;domain=1.2.3.4 ; Add IP address as local domain
393 ; You can have several "domain" settings
394 ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
395 ; Default is yes
396 ;autodomain=yes ; Turn this on to have Asterisk add local host
397 ; name and local IP to domain list.
399 ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
400 ; non-peers, use your primary domain "identity"
401 ; for From: headers instead of just your IP
402 ; address. This is to be polite and
403 ; it may be a mandatory requirement for some
404 ; destinations which do not have a prior
405 ; account relationship with your server.
407 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
408 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
409 ; SIP channel. Defaults to "no". An enabled jitterbuffer will
410 ; be used only if the sending side can create and the receiving
411 ; side can not accept jitter. The SIP channel can accept jitter,
412 ; thus a jitterbuffer on the receive SIP side will be used only
413 ; if it is forced and enabled.
415 ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
416 ; channel. Defaults to "no".
418 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
420 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
421 ; resynchronized. Useful to improve the quality of the voice, with
422 ; big jumps in/broken timestamps, usually sent from exotic devices
423 ; and programs. Defaults to 1000.
425 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
426 ; channel. Two implementations are currently available - "fixed"
427 ; (with size always equals to jbmaxsize) and "adaptive" (with
428 ; variable size, actually the new jb of IAX2). Defaults to fixed.
430 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
431 ;-----------------------------------------------------------------------------------
433 [authentication]
434 ; Global credentials for outbound calls, i.e. when a proxy challenges your
435 ; Asterisk server for authentication. These credentials override
436 ; any credentials in peer/register definition if realm is matched.
437 ;
438 ; This way, Asterisk can authenticate for outbound calls to other
439 ; realms. We match realm on the proxy challenge and pick an set of
440 ; credentials from this list
441 ; Syntax:
442 ; auth = <user>:<secret>@<realm>
443 ; auth = <user>#<md5secret>@<realm>
444 ; Example:
445 ;auth=mark:topsecret@digium.com
446 ;
447 ; You may also add auth= statements to [peer] definitions
448 ; Peer auth= override all other authentication settings if we match on realm
450 ;------------------------------------------------------------------------------
451 ; Users and peers have different settings available. Friends have all settings,
452 ; since a friend is both a peer and a user
453 ;
454 ; User config options: Peer configuration:
455 ; -------------------- -------------------
456 ; context context
457 ; callingpres callingpres
458 ; permit permit
459 ; deny deny
460 ; secret secret
461 ; md5secret md5secret
462 ; dtmfmode dtmfmode
463 ; canreinvite canreinvite
464 ; nat nat
465 ; callgroup callgroup
466 ; pickupgroup pickupgroup
467 ; language language
468 ; allow allow
469 ; disallow disallow
470 ; insecure insecure
471 ; trustrpid trustrpid
472 ; progressinband progressinband
473 ; promiscredir promiscredir
474 ; useclientcode useclientcode
475 ; accountcode accountcode
476 ; setvar setvar
477 ; callerid callerid
478 ; amaflags amaflags
479 ; call-limit call-limit
480 ; allowoverlap allowoverlap
481 ; allowsubscribe allowsubscribe
482 ; allowtransfer allowtransfer
483 ; subscribecontext subscribecontext
484 ; videosupport videosupport
485 ; maxcallbitrate maxcallbitrate
486 ; rfc2833compensate mailbox
487 ; t38pt_usertpsource username
488 ; template
489 ; fromdomain
490 ; regexten
491 ; fromuser
492 ; host
493 ; port
494 ; qualify
495 ; defaultip
496 ; rtptimeout
497 ; rtpholdtimeout
498 ; sendrpid
499 ; outboundproxy
500 ; rfc2833compensate
501 ; t38pt_usertpsource
503 ;[sip_proxy]
504 ; For incoming calls only. Example: FWD (Free World Dialup)
505 ; We match on IP address of the proxy for incoming calls
506 ; since we can not match on username (caller id)
507 ;type=peer
508 ;context=from-fwd
509 ;host=fwd.pulver.com
511 ;[sip_proxy-out]
512 ;type=peer ; we only want to call out, not be called
513 ;secret=guessit
514 ;username=yourusername ; Authentication user for outbound proxies
515 ;fromuser=yourusername ; Many SIP providers require this!
516 ;fromdomain=provider.sip.domain
517 ;host=box.provider.com
518 ;usereqphone=yes ; This provider requires ";user=phone" on URI
519 ;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer
520 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
521 ; Call-limits will not be enforced on real-time peers,
522 ; since they are not stored in-memory
523 ;port=80 ; The port number we want to connect to on the remote side
524 ; Also used as "defaultport" in combination with "defaultip" settings
526 ;------------------------------------------------------------------------------
527 ; Definitions of locally connected SIP devices
528 ;
529 ; type = user a device that authenticates to us by "from" field to place calls
530 ; type = peer a device we place calls to or that calls us and we match by host
531 ; type = friend two configurations (peer+user) in one
532 ;
533 ; For device names, we recommend using only a-z, numerics (0-9) and underscore
534 ;
535 ; For local phones, type=friend works most of the time
536 ;
537 ; If you have one-way audio, you probably have NAT problems.
538 ; If Asterisk is on a public IP, and the phone is inside of a NAT device
539 ; you will need to configure nat option for those phones.
540 ; Also, turn on qualify=yes to keep the nat session open
542 ;[grandstream1]
543 ;type=friend
544 ;context=from-sip ; Where to start in the dialplan when this phone calls
545 ;callerid=John Doe <1234> ; Full caller ID, to override the phones config
546 ; on incoming calls to Asterisk
547 ;host=192.168.0.23 ; we have a static but private IP address
548 ; No registration allowed
549 ;nat=no ; there is not NAT between phone and Asterisk
550 ;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
551 ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
552 ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
553 ; from the phone to asterisk
554 ; 1 for the explicit peer, 1 for the explicit user,
555 ; remember that a friend equals 1 peer and 1 user in
556 ; memory
557 ; This will affect your subscriptions as well.
558 ; There is no combined call counter for a "friend"
559 ; so there's currently no way in sip.conf to limit
560 ; to one inbound or outbound call per phone. Use
561 ; the group counters in the dial plan for that.
562 ;
563 ;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
564 ;disallow=all ; need to disallow=all before we can use allow=
565 ;allow=ulaw ; Note: In user sections the order of codecs
566 ; listed with allow= does NOT matter!
567 ;allow=alaw
568 ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
569 ;allow=g729 ; Pass-thru only unless g729 license obtained
570 ;callingpres=allowed_passed_screen ; Set caller ID presentation
571 ; See doc/callingpres.txt for more information
574 ;[xlite1]
575 ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
576 ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
577 ;type=friend
578 ;regexten=1234 ; When they register, create extension 1234
579 ;callerid="Jane Smith" <5678>
580 ;host=dynamic ; This device needs to register
581 ;nat=yes ; X-Lite is behind a NAT router
582 ;canreinvite=no ; Typically set to NO if behind NAT
583 ;disallow=all
584 ;allow=gsm ; GSM consumes far less bandwidth than ulaw
585 ;allow=ulaw
586 ;allow=alaw
587 ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
590 ;[snom]
591 ;type=friend ; Friends place calls and receive calls
592 ;context=from-sip ; Context for incoming calls from this user
593 ;secret=blah
594 ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
595 ;language=de ; Use German prompts for this user
596 ;host=dynamic ; This peer register with us
597 ;dtmfmode=inband ; Choices are inband, rfc2833, or info
598 ;defaultip=192.168.0.59 ; IP used until peer registers
599 ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
600 ;subscribemwi=yes ; Only send notifications if this phone
601 ; subscribes for mailbox notification
602 ;vmexten=voicemail ; dialplan extension to reach mailbox
603 ; sets the Message-Account in the MWI notify message
604 ; defaults to global vmexten which defaults to "asterisk"
605 ;disallow=all
606 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
609 ;[polycom]
610 ;type=friend ; Friends place calls and receive calls
611 ;context=from-sip ; Context for incoming calls from this user
612 ;secret=blahpoly
613 ;host=dynamic ; This peer register with us
614 ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
615 ;username=polly ; Username to use in INVITE until peer registers
616 ; Normally you do NOT need to set this parameter
617 ;disallow=all
618 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
619 ;progressinband=no ; Polycom phones don't work properly with "never"
622 ;[pingtel]
623 ;type=friend
624 ;secret=blah
625 ;host=dynamic
626 ;insecure=port ; Allow matching of peer by IP address without
627 ; matching port number
628 ;insecure=invite ; Do not require authentication of incoming INVITEs
629 ;insecure=port,invite ; (both)
630 ;qualify=1000 ; Consider it down if it's 1 second to reply
631 ; Helps with NAT session
632 ; qualify=yes uses default value
633 ;
634 ; Call group and Pickup group should be in the range from 0 to 63
635 ;
636 ;callgroup=1,3-4 ; We are in caller groups 1,3,4
637 ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
638 ;defaultip=192.168.0.60 ; IP address to use if peer has not registered
639 ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
640 ;permit=192.168.0.60/255.255.255.0
642 ;[cisco1]
643 ;type=friend
644 ;secret=blah
645 ;qualify=200 ; Qualify peer is no more than 200ms away
646 ;nat=yes ; This phone may be natted
647 ; Send SIP and RTP to the IP address that packet is
648 ; received from instead of trusting SIP headers
649 ;host=dynamic ; This device registers with us
650 ;canreinvite=no ; Asterisk by default tries to redirect the
651 ; RTP media stream (audio) to go directly from
652 ; the caller to the callee. Some devices do not
653 ; support this (especially if one of them is
654 ; behind a NAT).
655 ;defaultip=192.168.0.4 ; IP address to use until registration
656 ;username=goran ; Username to use when calling this device before registration
657 ; Normally you do NOT need to set this parameter
658 ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
660 ;[pre14-asterisk]
661 ;type=friend
662 ;secret=digium
663 ;host=dynamic
664 ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
665 ; You must have this turned on or DTMF reception will work improperly.
666 ;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
667 ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
668 ; external IP address of the remote device. If port forwarding is done at the client side
669 ; then UDPTL will flow to the remote device.