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1 ;
2 ; SIP Configuration example for Asterisk
3 ;
4 ; SIP dial strings
5 ;-----------------------------------------------------------
6 ; In the dialplan (extensions.conf) you can use several
7 ; syntaxes for dialing SIP devices.
8 ; SIP/devicename
9 ; SIP/username@domain (SIP uri)
10 ; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
11 ; SIP/devicename/extension
12 ;
13 ;
14 ; Devicename
15 ; devicename is defined as a peer in a section below.
16 ;
17 ; username@domain
18 ; Call any SIP user on the Internet
19 ; (Don't forget to enable DNS SRV records if you want to use this)
20 ;
21 ; devicename/extension
22 ; If you define a SIP proxy as a peer below, you may call
23 ; SIP/proxyhostname/user or SIP/user@proxyhostname
24 ; where the proxyhostname is defined in a section below
25 ; This syntax also works with ATA's with FXO ports
26 ;
27 ; SIP/username[:password[:md5secret[:authname]]]@host[:port]
28 ; This form allows you to specify password or md5secret and authname
29 ; without altering any authentication data in config.
30 ; Examples:
31 ;
32 ; SIP/*98@mysipproxy
33 ; SIP/sales:topsecret::account02@domain.com:5062
34 ; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
35 ;
36 ; All of these dial strings specify the SIP request URI.
37 ; In addition, you can specify a specific To: header by adding an
38 ; exclamation mark after the dial string, like
39 ;
40 ; SIP/sales@mysipproxy!sales@edvina.net
41 ;
42 ; CLI Commands
43 ; -------------------------------------------------------------
44 ; Useful CLI commands to check peers/users:
45 ; sip show peers Show all SIP peers (including friends)
46 ; sip show users Show all SIP users (including friends)
47 ; sip show registry Show status of hosts we register with
48 ;
49 ; sip set debug Show all SIP messages
50 ;
51 ; module reload chan_sip.so Reload configuration file
52 ; Active SIP peers will not be reconfigured
53 ;
55 ; ** Deprecated configuration options **
56 ; The "call-limit" configuation option is deprecated. It still works in
57 ; this version of Asterisk, but will disappear in the next version.
58 ; You are encouraged to use the dialplan groupcount functionality
59 ; to enforce call limits instead of using this channel-specific method.
60 ;
61 ; You can still set limits per device in sip.conf or in a database by using
62 ; "setvar" to set variables that can be used in the dialplan for various limits.
64 [general]
65 context=default ; Default context for incoming calls
66 ;allowguest=no ; Allow or reject guest calls (default is yes)
67 ;match_auth_username=yes ; if available, match user entry using the
68 ; 'username' field from the authentication line
69 ; instead of the From: field.
70 allowoverlap=no ; Disable overlap dialing support. (Default is yes)
71 ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
72 ; Default is enabled
73 ;realm=mydomain.tld ; Realm for digest authentication
74 ; defaults to "asterisk". If you set a system name in
75 ; asterisk.conf, it defaults to that system name
76 ; Realms MUST be globally unique according to RFC 3261
77 ; Set this to your host name or domain name
78 udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
79 ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
81 ;
82 ; Note that the TCP and TLS support for chan_sip is currently considered
83 ; experimental. Since it is new, all of the related configuration options are
84 ; subject to change in any release. If they are changed, the changes will
85 ; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
86 ;
87 tcpenable=no ; Enable server for incoming TCP connections (default is no)
88 tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
89 ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
91 ;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
92 ;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
93 ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
94 ; Remember that the IP address must match the common name (hostname) in the
95 ; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
97 ;tlscertfile=asterisk.pem ; Certificate file (*.pem only) to use for TLS connections
98 ; default is to look for "asterisk.pem" in current directory
100 ;tlscafile=</path/to/certificate>
101 ; If the server your connecting to uses a self signed certificate
102 ; you should have their certificate installed here so the code can
103 ; verify the authenticity of their certificate.
105 ;tlscadir=</path/to/ca/dir>
106 ; A directory full of CA certificates. The files must be named with
107 ; the CA subject name hash value.
108 ; (see man SSL_CTX_load_verify_locations for more info)
110 ;tlsdontverifyserver=[yes|no]
111 ; If set to yes, don't verify the servers certificate when acting as
112 ; a client. If you don't have the server's CA certificate you can
113 ; set this and it will connect without requiring tlscafile to be set.
114 ; Default is no.
116 ;tlscipher=<SSL cipher string>
117 ; A string specifying which SSL ciphers to use or not use
118 ; A list of valid SSL cipher strings can be found at:
119 ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
121 srvlookup=yes ; Enable DNS SRV lookups on outbound calls
122 ; Note: Asterisk only uses the first host
123 ; in SRV records
124 ; Disabling DNS SRV lookups disables the
125 ; ability to place SIP calls based on domain
126 ; names to some other SIP users on the Internet
128 ;pedantic=yes ; Enable checking of tags in headers,
129 ; international character conversions in URIs
130 ; and multiline formatted headers for strict
131 ; SIP compatibility (defaults to "no")
133 ; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
134 ;tos_sip=cs3 ; Sets TOS for SIP packets.
135 ;tos_audio=ef ; Sets TOS for RTP audio packets.
136 ;tos_video=af41 ; Sets TOS for RTP video packets.
137 ;tos_text=af41 ; Sets TOS for RTP text packets.
139 ;cos_sip=3 ; Sets 802.1p priority for SIP packets.
140 ;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
141 ;cos_video=4 ; Sets 802.1p priority for RTP video packets.
142 ;cos_text=3 ; Sets 802.1p priority for RTP text packets.
144 ;maxexpiry=3600 ; Maximum allowed time of incoming registrations
145 ; and subscriptions (seconds)
146 ;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
147 ;defaultexpiry=120 ; Default length of incoming/outgoing registration
148 ;qualifyfreq=60 ; Qualification: How often to check for the
149 ; host to be up in seconds
150 ; Set to low value if you use low timeout for
151 ; NAT of UDP sessions
152 ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
153 ;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
154 ; fully. Enable this option to not get error messages
155 ; when sending MWI to phones with this bug.
156 ;vmexten=voicemail ; dialplan extension to reach mailbox sets the
157 ; Message-Account in the MWI notify message
158 ; defaults to "asterisk"
159 ;disallow=all ; First disallow all codecs
160 ;allow=ulaw ; Allow codecs in order of preference
161 ;allow=ilbc ; see doc/rtp-packetization for framing options
162 ;
163 ; This option specifies a preference for which music on hold class this channel
164 ; should listen to when put on hold if the music class has not been set on the
165 ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
166 ; channel putting this one on hold did not suggest a music class.
167 ;
168 ; This option may be specified globally, or on a per-user or per-peer basis.
169 ;
170 ;mohinterpret=default
171 ;
172 ; This option specifies which music on hold class to suggest to the peer channel
173 ; when this channel places the peer on hold. It may be specified globally or on
174 ; a per-user or per-peer basis.
175 ;
176 ;mohsuggest=default
177 ;
178 ;parkinglot=plaza ; Sets the default parking lot for call parking
179 ; This may also be set for individual users/peers
180 ; Parkinglots are configured in features.conf
181 ;language=en ; Default language setting for all users/peers
182 ; This may also be set for individual users/peers
183 ;relaxdtmf=yes ; Relax dtmf handling
184 ;trustrpid = no ; If Remote-Party-ID should be trusted
185 ;sendrpid = yes ; If Remote-Party-ID should be sent
186 ;progressinband=never ; If we should generate in-band ringing always
187 ; use 'never' to never use in-band signalling, even in cases
188 ; where some buggy devices might not render it
189 ; Valid values: yes, no, never Default: never
190 ;useragent=Asterisk PBX ; Allows you to change the user agent string
191 ; The default user agent string also contains the Asterisk
192 ; version. If you don't want to expose this, change the
193 ; useragent string.
194 ;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
195 ; Like the useragent parameter, the default user agent string
196 ; also contains the Asterisk version.
197 ;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
198 ; This field MUST NOT contain spaces
199 ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
200 ; Note that promiscredir when redirects are made to the
201 ; local system will cause loops since Asterisk is incapable
202 ; of performing a "hairpin" call.
203 ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
204 ; a valid phone number
205 ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
206 ; Other options:
207 ; info : SIP INFO messages (application/dtmf-relay)
208 ; shortinfo : SIP INFO messages (application/dtmf)
209 ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
210 ; auto : Use rfc2833 if offered, inband otherwise
212 ;compactheaders = yes ; send compact sip headers.
213 ;
214 ;videosupport=yes ; Turn on support for SIP video. You need to turn this
215 ; on in this section to get any video support at all.
216 ; You can turn it off on a per peer basis if the general
217 ; video support is enabled, but you can't enable it for
218 ; one peer only without enabling in the general section.
219 ; If you set videosupport to "always", then RTP ports will
220 ; always be set up for video, even on clients that don't
221 ; support it. This assists callfile-derived calls and
222 ; certain transferred calls to use always use video when
223 ; available. [yes|NO|always]
225 ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
226 ; Videosupport and maxcallbitrate is settable
227 ; for peers and users as well
228 ;callevents=no ; generate manager events when sip ua
229 ; performs events (e.g. hold)
230 ;authfailureevents=no ; generate manager "peerstatus" events when peer can't
231 ; authenticate with Asterisk. Peerstatus will be "rejected".
232 ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
233 ; for any reason, always reject with '401 Unauthorized'
234 ; instead of letting the requester know whether there was
235 ; a matching user or peer for their request
237 ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
238 ; order instead of RFC3551 packing order (this is required
239 ; for Sipura and Grandstream ATAs, among others). This is
240 ; contrary to the RFC3551 specification, the peer _should_
241 ; be negotiating AAL2-G726-32 instead :-(
242 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
243 ;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
244 ;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
245 ;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls
246 ; ; (could also be tcp,udp) - defining transports on the proxy line only
247 ; ; applies for the global proxy, otherwise use the transport= option
248 ;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
249 ; your localnet setting. Unless you have some sort of strange network
250 ; setup you will not need to enable this.
252 ;
253 ; If regcontext is specified, Asterisk will dynamically create and destroy a
254 ; NoOp priority 1 extension for a given peer who registers or unregisters with
255 ; us and have a "regexten=" configuration item.
256 ; Multiple contexts may be specified by separating them with '&'. The
257 ; actual extension is the 'regexten' parameter of the registering peer or its
258 ; name if 'regexten' is not provided. If more than one context is provided,
259 ; the context must be specified within regexten by appending the desired
260 ; context after '@'. More than one regexten may be supplied if they are
261 ; separated by '&'. Patterns may be used in regexten.
262 ;
263 ;regcontext=sipregistrations
264 ;regextenonqualify=yes ; Default "no"
265 ; If you have qualify on and the peer becomes unreachable
266 ; this setting will enforce inactivation of the regexten
267 ; extension for the peer
268 ;
269 ;--------------------------- SIP timers ----------------------------------------------------
270 ; These timers are used primarily in INVITE transactions.
271 ; The default for Timer T1 is 500 ms or the measured run-trip time between
272 ; Asterisk and the device if you have qualify=yes for the device.
273 ;
274 ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
275 ; Defaults to 100 ms
276 ;timert1=500 ; Default T1 timer
277 ; Defaults to 500 ms or the measured round-trip
278 ; time to a peer (qualify=yes).
279 ;timerb=32000 ; Call setup timer. If a provisional response is not received
280 ; in this amount of time, the call will autocongest
281 ; Defaults to 64*timert1
283 ;--------------------------- RTP timers ----------------------------------------------------
284 ; These timers are currently used for both audio and video streams. The RTP timeouts
285 ; are only applied to the audio channel.
286 ; The settings are settable in the global section as well as per device
287 ;
288 ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
289 ; on the audio channel
290 ; when we're not on hold. This is to be able to hangup
291 ; a call in the case of a phone disappearing from the net,
292 ; like a powerloss or grandma tripping over a cable.
293 ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
294 ; on the audio channel
295 ; when we're on hold (must be > rtptimeout)
296 ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
297 ; (default is off - zero)
299 ;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
300 ; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
301 ; This mechanism can detect and reclaim SIP channels that do not terminate through normal
302 ; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
303 ; The operation of Session-Timers is driven by the following configuration parameters:
304 ;
305 ; * session-timers - Session-Timers feature operates in the following three modes:
306 ; originate : Request and run session-timers always
307 ; accept : Run session-timers only when requested by other UA
308 ; refuse : Do not run session timers in any case
309 ; The default mode of operation is 'accept'.
310 ; * session-expires - Maximum session refresh interval in seconds. Defaults to 1800 secs.
311 ; * session-minse - Minimum session refresh interval in seconds. Defualts to 90 secs.
312 ; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
313 ;
314 ;session-timers=originate
315 ;session-expires=600
316 ;session-minse=90
317 ;session-refresher=uas
318 ;
319 ;--------------------------- HASH TABLE SIZES ------------------------------------------------
320 ; For maximum efficiency, adjust the following
321 ; values to be slightly larger than the maximum number of in-memory objects (devices).
322 ; Too large, and space is wasted. Too small, and things will run slower.
323 ; 563 is probably way too big for small (home) applications, but it
324 ; should cover most small/medium sites.
325 ; It is recommended to make the sizes be a prime number!
326 ; This was internally set to 17 for small-memory applications...
327 ; All tables default to 563, except when compiled in LOW_MEMORY mode,
328 ; in which case, they default to 17. You can override this by uncommenting
329 ; the following, and changing the values.
330 ;hash_users=563
331 ;hash_peers=563
332 ;hash_dialogs=563
334 ;--------------------------- SIP DEBUGGING ---------------------------------------------------
335 ;sipdebug = yes ; Turn on SIP debugging by default, from
336 ; the moment the channel loads this configuration
337 ;recordhistory=yes ; Record SIP history by default
338 ; (see sip history / sip no history)
339 ;dumphistory=yes ; Dump SIP history at end of SIP dialogue
340 ; SIP history is output to the DEBUG logging channel
343 ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
344 ; You can subscribe to the status of extensions with a "hint" priority
345 ; (See extensions.conf.sample for examples)
346 ; chan_sip support two major formats for notifications: dialog-info and SIMPLE
347 ;
348 ; You will get more detailed reports (busy etc) if you have a call counter enabled
349 ; for a device.
350 ;
351 ; If you set the busylevel, we will indicate busy when we have a number of calls that
352 ; matches the busylevel treshold.
353 ;
354 ; For queues, you will need this level of detail in status reporting, regardless
355 ; if you use SIP subscriptions. Queues and manager use the same internal interface
356 ; for reading status information.
357 ;
358 ; Note: Subscriptions does not work if you have a realtime dialplan and use the
359 ; realtime switch.
360 ;
361 ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
362 ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
363 ; Useful to limit subscriptions to local extensions
364 ; Settable per peer/user also
365 ;notifyringing = yes ; Control whether subscriptions already INUSE get sent
366 ; RINGING when another call is sent (default: no)
367 ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
368 ; Turning on notifyringing and notifyhold will add a lot
369 ; more database transactions if you are using realtime.
370 ;callcounter = yes ; Enable call counters on devices. This can be set per
371 ; device too.
372 ;counteronpeer = yes ; Apply call counting on peers only. This will improve
373 ; status notification when you are using type=friend
374 ; Inbound calls, that really apply to the user part
375 ; of a friend will now be added to and compared with
376 ; the peer counter instead of applying two call counters,
377 ; one for the peer and one for the user.
378 ; "sip show inuse" will only show active calls on
379 ; the peer side of a "type=friend" object if this
380 ; setting is turned on.
382 ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
383 ;
384 ; This setting is available in the [general] section as well as in device configurations.
385 ; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
386 ; both parties have T38 support enabled in their Asterisk configuration
387 ; This has to be enabled in the general section for all devices to work. You can then
388 ; disable it on a per device basis.
389 ;
390 ; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.
391 ;
392 ; t38pt_udptl = yes ; Default false
393 ;
394 ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
395 ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
396 ; Format for the register statement is:
397 ; register => [transport://]user[:secret[:authuser]]@host[:port][/extension][~expiry]
398 ;
399 ;
400 ;
401 ; If no extension is given, the 's' extension is used. The extension needs to
402 ; be defined in extensions.conf to be able to accept calls from this SIP proxy
403 ; (provider).
404 ;
405 ; host is either a host name defined in DNS or the name of a section defined
406 ; below.
407 ;
408 ; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
409 ; this is equivalent to having the following line in the general section:
410 ;
411 ; register => username:secret@host/callbackextension
412 ;
413 ; and more readable because you don't have to write the parameters in two places
414 ; (note that the "port" is ignored - this is a bug that should be fixed).
415 ;
416 ; Examples:
417 ;
418 ;register => 1234:password@mysipprovider.com
419 ;
420 ; This will pass incoming calls to the 's' extension
421 ;
422 ;
423 ;register => 2345:password@sip_proxy/1234
424 ;
425 ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
426 ; connect to local extension 1234 in extensions.conf, default context,
427 ; unless you configure a [sip_proxy] section below, and configure a
428 ; context.
429 ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
430 ; Tip 2: Use separate type=peer and type=user sections for SIP providers
431 ; (instead of type=friend) if you have calls in both directions
433 ;registertimeout=20 ; retry registration calls every 20 seconds (default)
434 ;registerattempts=10 ; Number of registration attempts before we give up
435 ; 0 = continue forever, hammering the other server
436 ; until it accepts the registration
437 ; Default is 0 tries, continue forever
439 ;----------------------------------------- NAT SUPPORT ------------------------
440 ;
441 ; WARNING: SIP operation behind a NAT is tricky and you really need
442 ; to read and understand well the following section.
443 ;
444 ; When Asterisk is behind a NAT device, the "local" address (and port) that
445 ; a socket is bound to has different values when seen from the inside or
446 ; from the outside of the NATted network. Unfortunately this address must
447 ; be communicated to the outside (e.g. in SIP and SDP messages), and in
448 ; order to determine the correct value Asterisk needs to know:
449 ;
450 ; + whether it is talking to someone "inside" or "outside" of the NATted network.
451 ; This is configured by assigning the "localnet" parameter with a list
452 ; of network addresses that are considered "inside" of the NATted network.
453 ; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
454 ; Multiple entries are allowed, e.g. a reasonable set is the following:
455 ;
456 ; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
457 ; localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
458 ; localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
459 ; localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
460 ;
461 ; + the "externally visible" address and port number to be used when talking
462 ; to a host outside the NAT. This information is derived by one of the
463 ; following (mutually exclusive) config file parameters:
464 ;
465 ; a. "externip = hostname[:port]" specifies a static address[:port] to
466 ; be used in SIP and SDP messages.
467 ; The hostname is looked up only once, when [re]loading sip.conf .
468 ; If a port number is not present, use the "bindport" value (which is
469 ; not guaranteed to work correctly, because a NAT box might remap the
470 ; port number as well as the address).
471 ; This approach can be useful if you have a NAT device where you can
472 ; configure the mapping statically. Examples:
473 ;
474 ; externip = 12.34.56.78 ; use this address.
475 ; externip = 12.34.56.78:9900 ; use this address and port.
476 ; externip = mynat.my.org:12600 ; Public address of my nat box.
477 ;
478 ; b. "externhost = hostname[:port]" is similar to "externip" except
479 ; that the hostname is looked up every "externrefresh" seconds
480 ; (default 10s). This can be useful when your NAT device lets you choose
481 ; the port mapping, but the IP address is dynamic.
482 ; Beware, you might suffer from service disruption when the name server
483 ; resolution fails. Examples:
484 ;
485 ; externhost=foo.dyndns.net ; refreshed periodically
486 ; externrefresh=180 ; change the refresh interval
487 ;
488 ; c. "stunaddr = stun.server[:port]" queries the STUN server specified
489 ; as an argument to obtain the external address/port.
490 ; Queries are also sent periodically every "externrefresh" seconds
491 ; (as a side effect, sending the query also acts as a keepalive for
492 ; the state entry on the nat box):
493 ;
494 ; stunaddr = foo.stun.com:3478
495 ; externrefresh = 15
496 ;
497 ; Note that at the moment all these mechanism work only for the SIP socket.
498 ; The IP address discovered with externip/externhost/STUN is reused for
499 ; media sessions as well, but the port numbers are not remapped so you
500 ; may still experience problems.
501 ;
502 ; NOTE 1: in some cases, NAT boxes will use different port numbers in
503 ; the internal<->external mapping. In these cases, the "externip" and
504 ; "externhost" might not help you configure addresses properly, and you
505 ; really need to use STUN.
506 ;
507 ; NOTE 2: when using "externip" or "externhost", the address part is
508 ; also used as the external address for media sessions.
509 ; If you use "stunaddr", STUN queries will be sent to the same server
510 ; also from media sockets, and this should permit a correct mapping of
511 ; the port numbers as well.
512 ;
513 ; In addition to the above, Asterisk has an additional "nat" parameter to
514 ; address NAT-related issues in incoming SIP or media sessions.
515 ; In particular, depending on the 'nat= ' settings described below, Asterisk
516 ; may override the address/port information specified in the SIP/SDP messages,
517 ; and use the information (sender address) supplied by the network stack instead.
518 ; However, this is only useful if the external traffic can reach us.
519 ; The following settings are allowed (both globally and in individual sections):
520 ;
521 ; nat = no ; default. Use NAT mode only according to RFC3581 (;rport)
522 ; nat = yes ; Always ignore info and assume NAT
523 ; nat = never ; Never attempt NAT mode or RFC3581 support
524 ; nat = route ; route = Assume NAT, don't send rport
525 ; ; (work around more UNIDEN bugs)
527 ;----------------------------------- MEDIA HANDLING --------------------------------
528 ; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
529 ; no reason for Asterisk to stay in the media path, the media will be redirected.
530 ; This does not really work with in the case where Asterisk is outside and have
531 ; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
532 ;
533 ;canreinvite=yes ; Asterisk by default tries to redirect the
534 ; RTP media stream (audio) to go directly from
535 ; the caller to the callee. Some devices do not
536 ; support this (especially if one of them is behind a NAT).
537 ; The default setting is YES. If you have all clients
538 ; behind a NAT, or for some other reason wants Asterisk to
539 ; stay in the audio path, you may want to turn this off.
541 ; This setting also affect direct RTP
542 ; at call setup (a new feature in 1.4 - setting up the
543 ; call directly between the endpoints instead of sending
544 ; a re-INVITE).
546 ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
547 ; the call directly with media peer-2-peer without re-invites.
548 ; Will not work for video and cases where the callee sends
549 ; RTP payloads and fmtp headers in the 200 OK that does not match the
550 ; callers INVITE. This will also fail if canreinvite is enabled when
551 ; the device is actually behind NAT.
553 ;canreinvite=nonat ; An additional option is to allow media path redirection
554 ; (reinvite) but only when the peer where the media is being
555 ; sent is known to not be behind a NAT (as the RTP core can
556 ; determine it based on the apparent IP address the media
557 ; arrives from).
559 ;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
560 ; instead of INVITE. This can be combined with 'nonat', as
561 ; 'canreinvite=update,nonat'. It implies 'yes'.
563 ;----------------------------------------- REALTIME SUPPORT ------------------------
564 ; For additional information on ARA, the Asterisk Realtime Architecture,
565 ; please read realtime.txt and extconfig.txt in the /doc directory of the
566 ; source code.
567 ;
568 ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
569 ; just like friends added from the config file only on a
570 ; as-needed basis? (yes|no)
572 ;rtsavesysname=yes ; Save systemname in realtime database at registration
573 ; Default= no
575 ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
576 ; If set to yes, when a SIP UA registers successfully, the ip address,
577 ; the origination port, the registration period, and the username of
578 ; the UA will be set to database via realtime.
579 ; If not present, defaults to 'yes'. Note: realtime peers will
580 ; probably not function across reloads in the way that you expect, if
581 ; you turn this option off.
582 ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
583 ; as if it had just registered? (yes|no|<seconds>)
584 ; If set to yes, when the registration expires, the friend will
585 ; vanish from the configuration until requested again. If set
586 ; to an integer, friends expire within this number of seconds
587 ; instead of the registration interval.
589 ;ignoreregexpire=yes ; Enabling this setting has two functions:
590 ;
591 ; For non-realtime peers, when their registration expires, the
592 ; information will _not_ be removed from memory or the Asterisk database
593 ; if you attempt to place a call to the peer, the existing information
594 ; will be used in spite of it having expired
595 ;
596 ; For realtime peers, when the peer is retrieved from realtime storage,
597 ; the registration information will be used regardless of whether
598 ; it has expired or not; if it expires while the realtime peer
599 ; is still in memory (due to caching or other reasons), the
600 ; information will not be removed from realtime storage
602 ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
603 ; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
604 ; domains, each of which can direct the call to a specific context if desired.
605 ; By default, all domains are accepted and sent to the default context or the
606 ; context associated with the user/peer placing the call.
607 ; REGISTER to non-local domains will be automatically denied if a domain
608 ; list is configured.
609 ;
610 ; Domains can be specified using:
611 ; domain=<domain>[,<context>]
612 ; Examples:
613 ; domain=myasterisk.dom
614 ; domain=customer.com,customer-context
615 ;
616 ; In addition, all the 'default' domains associated with a server should be
617 ; added if incoming request filtering is desired.
618 ; autodomain=yes
619 ;
620 ; To disallow requests for domains not serviced by this server:
621 ; allowexternaldomains=no
623 ;domain=mydomain.tld,mydomain-incoming
624 ; Add domain and configure incoming context
625 ; for external calls to this domain
626 ;domain=1.2.3.4 ; Add IP address as local domain
627 ; You can have several "domain" settings
628 ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
629 ; Default is yes
630 ;autodomain=yes ; Turn this on to have Asterisk add local host
631 ; name and local IP to domain list.
633 ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
634 ; non-peers, use your primary domain "identity"
635 ; for From: headers instead of just your IP
636 ; address. This is to be polite and
637 ; it may be a mandatory requirement for some
638 ; destinations which do not have a prior
639 ; account relationship with your server.
641 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
642 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
643 ; SIP channel. Defaults to "no". An enabled jitterbuffer will
644 ; be used only if the sending side can create and the receiving
645 ; side can not accept jitter. The SIP channel can accept jitter,
646 ; thus a jitterbuffer on the receive SIP side will be used only
647 ; if it is forced and enabled.
649 ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
650 ; channel. Defaults to "no".
652 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
654 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
655 ; resynchronized. Useful to improve the quality of the voice, with
656 ; big jumps in/broken timestamps, usually sent from exotic devices
657 ; and programs. Defaults to 1000.
659 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
660 ; channel. Two implementations are currently available - "fixed"
661 ; (with size always equals to jbmaxsize) and "adaptive" (with
662 ; variable size, actually the new jb of IAX2). Defaults to fixed.
664 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
665 ;-----------------------------------------------------------------------------------
667 [authentication]
668 ; Global credentials for outbound calls, i.e. when a proxy challenges your
669 ; Asterisk server for authentication. These credentials override
670 ; any credentials in peer/register definition if realm is matched.
671 ;
672 ; This way, Asterisk can authenticate for outbound calls to other
673 ; realms. We match realm on the proxy challenge and pick an set of
674 ; credentials from this list
675 ; Syntax:
676 ; auth = <user>:<secret>@<realm>
677 ; auth = <user>#<md5secret>@<realm>
678 ; Example:
679 ;auth=mark:topsecret@digium.com
680 ;
681 ; You may also add auth= statements to [peer] definitions
682 ; Peer auth= override all other authentication settings if we match on realm
684 ;------------------------------------------------------------------------------
685 ; Users and peers have different settings available. Friends have all settings,
686 ; since a friend is both a peer and a user
687 ;
688 ; User config options: Peer configuration:
689 ; -------------------- -------------------
690 ; context context
691 ; callingpres callingpres
692 ; permit permit
693 ; deny deny
694 ; secret secret
695 ; md5secret md5secret
696 ; transport transport
697 ; dtmfmode dtmfmode
698 ; canreinvite canreinvite
699 ; nat nat
700 ; callgroup callgroup
701 ; pickupgroup pickupgroup
702 ; language language
703 ; allow allow
704 ; disallow disallow
705 ; insecure insecure
706 ; trustrpid trustrpid
707 ; progressinband progressinband
708 ; promiscredir promiscredir
709 ; useclientcode useclientcode
710 ; accountcode accountcode
711 ; setvar setvar
712 ; callerid callerid
713 ; amaflags amaflags
714 ; call-limit call-limit (deprecated)
715 ; callcounter callcounter
716 ; allowoverlap allowoverlap
717 ; allowsubscribe allowsubscribe
718 ; allowtransfer allowtransfer
719 ; subscribecontext subscribecontext
720 ; videosupport videosupport
721 ; maxcallbitrate maxcallbitrate
722 ; rfc2833compensate mailbox
723 ; session-timers busylevel
724 ; session-expires
725 ; session-minse template
726 ; session-refresher fromdomain
727 ; t38pt_usertpsource regexten
728 ; fromuser
729 ; host
730 ; port
731 ; qualify
732 ; defaultip
733 ; defaultuser
734 ; rtptimeout
735 ; rtpholdtimeout
736 ; sendrpid
737 ; outboundproxy
738 ; rfc2833compensate
739 ; callbackextension
740 ; registertrying
741 ; session-timers
742 ; session-expires
743 ; session-minse
744 ; session-refresher
745 ; timert1
746 ; timerb
747 ; qualifyfreq
748 ; t38pt_usertpsource
750 ;[sip_proxy]
751 ; For incoming calls only. Example: FWD (Free World Dialup)
752 ; We match on IP address of the proxy for incoming calls
753 ; since we can not match on username (caller id)
754 ;type=peer
755 ;context=from-fwd
756 ;host=fwd.pulver.com
758 ;[sip_proxy-out]
759 ;type=peer ; we only want to call out, not be called
760 ;secret=guessit
761 ;defaultuser=yourusername ; Authentication user for outbound proxies
762 ;fromuser=yourusername ; Many SIP providers require this!
763 ;fromdomain=provider.sip.domain
764 ;host=box.provider.com
765 ;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
766 ; ; accept both tcp and udp. Default is udp. The first transport
767 ; ; listed will always be used for outgoing connections.
768 ;usereqphone=yes ; This provider requires ";user=phone" on URI
769 ;callcounter=yes ; Enable call counter
770 ;busylevel=2 ; Signal busy at 2 or more calls
771 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
772 ;port=80 ; The port number we want to connect to on the remote side
773 ; Also used as "defaultport" in combination with "defaultip" settings
775 ;--- sample definition for a provider
776 ;[provider1]
777 ;type=peer
778 ;host=sip.provider1.com
779 ;fromuser=4015552299 ; how your provider knows you
780 ;secret=youwillneverguessit
781 ;callbackextension=123 ; Register with this server and require calls coming back to this extension
782 ;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
783 ; ; accept both tcp and udp. Default is udp. The first transport
784 ; ; listed will always be used for outgoing connections.
786 ;------------------------------------------------------------------------------
787 ; Definitions of locally connected SIP devices
788 ;
789 ; type = user a device that authenticates to us by "from" field to place calls
790 ; type = peer a device we place calls to or that calls us and we match by host
791 ; type = friend two configurations (peer+user) in one
792 ;
793 ; For device names, we recommend using only a-z, numerics (0-9) and underscore
794 ;
795 ; For local phones, type=friend works most of the time
796 ;
797 ; If you have one-way audio, you probably have NAT problems.
798 ; If Asterisk is on a public IP, and the phone is inside of a NAT device
799 ; you will need to configure nat option for those phones.
800 ; Also, turn on qualify=yes to keep the nat session open
801 ;
802 ; Because you might have a large number of similar sections, it is generally
803 ; convenient to use templates for the common parameters, and add them
804 ; the the various sections. Examples are below, and we can even leave
805 ; the templates uncommented as they will not harm:
807 [basic-options](!) ; a template
808 dtmfmode=rfc2833
809 context=from-office
810 type=friend
812 [natted-phone](!,basic-options) ; another template inheriting basic-options
813 nat=yes
814 canreinvite=no
815 host=dynamic
817 [public-phone](!,basic-options) ; another template inheriting basic-options
818 nat=no
819 canreinvite=yes
821 [my-codecs](!) ; a template for my preferred codecs
822 disallow=all
823 allow=ilbc
824 allow=g729
825 allow=gsm
826 allow=g723
827 allow=ulaw
829 [ulaw-phone](!) ; and another one for ulaw-only
830 disallow=all
831 allow=ulaw
833 ; and finally instantiate a few phones
834 ;
835 ; [2133](natted-phone,my-codecs)
836 ; secret = peekaboo
837 ; [2134](natted-phone,ulaw-phone)
838 ; secret = not_very_secret
839 ; [2136](public-phone,ulaw-phone)
840 ; secret = not_very_secret_either
841 ; ...
842 ;
844 ; Standard configurations not using templates look like this:
845 ;
846 ;[grandstream1]
847 ;type=friend
848 ;context=from-sip ; Where to start in the dialplan when this phone calls
849 ;callerid=John Doe <1234> ; Full caller ID, to override the phones config
850 ; on incoming calls to Asterisk
851 ;host=192.168.0.23 ; we have a static but private IP address
852 ; No registration allowed
853 ;nat=no ; there is not NAT between phone and Asterisk
854 ;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
855 ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
856 ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
857 ; from the phone to asterisk (deprecated)
858 ; 1 for the explicit peer, 1 for the explicit user,
859 ; remember that a friend equals 1 peer and 1 user in
860 ; memory
861 ; There is no combined call counter for a "friend"
862 ; so there's currently no way in sip.conf to limit
863 ; to one inbound or outbound call per phone. Use
864 ; the group counters in the dial plan for that.
865 ;
866 ;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
867 ;disallow=all ; need to disallow=all before we can use allow=
868 ;allow=ulaw ; Note: In user sections the order of codecs
869 ; listed with allow= does NOT matter!
870 ;allow=alaw
871 ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
872 ;allow=g729 ; Pass-thru only unless g729 license obtained
873 ;callingpres=allowed_passed_screen ; Set caller ID presentation
874 ; See README.callingpres for more information
876 ;[xlite1]
877 ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
878 ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
879 ;type=friend
880 ;regexten=1234 ; When they register, create extension 1234
881 ;callerid="Jane Smith" <5678>
882 ;host=dynamic ; This device needs to register
883 ;nat=yes ; X-Lite is behind a NAT router
884 ;canreinvite=no ; Typically set to NO if behind NAT
885 ;disallow=all
886 ;allow=gsm ; GSM consumes far less bandwidth than ulaw
887 ;allow=ulaw
888 ;allow=alaw
889 ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
890 ;registertrying=yes ; Send a 100 Trying when the device registers.
892 ;[snom]
893 ;type=friend ; Friends place calls and receive calls
894 ;context=from-sip ; Context for incoming calls from this user
895 ;secret=blah
896 ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
897 ;language=de ; Use German prompts for this user
898 ;host=dynamic ; This peer register with us
899 ;dtmfmode=inband ; Choices are inband, rfc2833, or info
900 ;defaultip=192.168.0.59 ; IP used until peer registers
901 ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
902 ;subscribemwi=yes ; Only send notifications if this phone
903 ; subscribes for mailbox notification
904 ;vmexten=voicemail ; dialplan extension to reach mailbox
905 ; sets the Message-Account in the MWI notify message
906 ; defaults to global vmexten which defaults to "asterisk"
907 ;disallow=all
908 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
911 ;[polycom]
912 ;type=friend ; Friends place calls and receive calls
913 ;context=from-sip ; Context for incoming calls from this user
914 ;secret=blahpoly
915 ;host=dynamic ; This peer register with us
916 ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
917 ;defaultuser=polly ; Username to use in INVITE until peer registers
918 ;defaultip=192.168.40.123
919 ; Normally you do NOT need to set this parameter
920 ;disallow=all
921 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
922 ;progressinband=no ; Polycom phones don't work properly with "never"
925 ;[pingtel]
926 ;type=friend
927 ;secret=blah
928 ;host=dynamic
929 ;insecure=port ; Allow matching of peer by IP address without
930 ; matching port number
931 ;insecure=invite ; Do not require authentication of incoming INVITEs
932 ;insecure=port,invite ; (both)
933 ;qualify=1000 ; Consider it down if it's 1 second to reply
934 ; Helps with NAT session
935 ; qualify=yes uses default value
936 ;qualifyfreq=60 ; Qualification: How often to check for the
937 ; host to be up in seconds
938 ; Set to low value if you use low timeout for
939 ; NAT of UDP sessions
940 ;
941 ; Call group and Pickup group should be in the range from 0 to 63
942 ;
943 ;callgroup=1,3-4 ; We are in caller groups 1,3,4
944 ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
945 ;defaultip=192.168.0.60 ; IP address to use if peer has not registered
946 ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
947 ;permit=192.168.0.60/255.255.255.0
949 ;[cisco1]
950 ;type=friend
951 ;secret=blah
952 ;qualify=200 ; Qualify peer is no more than 200ms away
953 ;nat=yes ; This phone may be natted
954 ; Send SIP and RTP to the IP address that packet is
955 ; received from instead of trusting SIP headers
956 ;host=dynamic ; This device registers with us
957 ;canreinvite=no ; Asterisk by default tries to redirect the
958 ; RTP media stream (audio) to go directly from
959 ; the caller to the callee. Some devices do not
960 ; support this (especially if one of them is
961 ; behind a NAT).
962 ;defaultip=192.168.0.4 ; IP address to use until registration
963 ;defaultuser=goran ; Username to use when calling this device before registration
964 ; Normally you do NOT need to set this parameter
965 ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
966 ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
967 ; cause the given audio file to
968 ; be played upon completion of
969 ; an attended transfer.
971 ;[pre14-asterisk]
972 ;type=friend
973 ;secret=digium
974 ;host=dynamic
975 ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
976 ; You must have this turned on or DTMF reception will work improperly.
977 ;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
978 ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
979 ; external IP address of the remote device. If port forwarding is done at the client side
980 ; then UDPTL will flow to the remote device.