2 ; SIP Configuration example for Asterisk
5 ;-----------------------------------------------------------
6 ; In the dialplan (extensions.conf) you can use several
7 ; syntaxes for dialing SIP devices.
9 ; SIP/username@domain (SIP uri)
10 ; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
11 ; SIP/devicename/extension
15 ; devicename is defined as a peer in a section below.
18 ; Call any SIP user on the Internet
19 ; (Don't forget to enable DNS SRV records if you want to use this)
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
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.
33 ; SIP/sales:topsecret::account02@domain.com:5062
34 ; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
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
40 ; SIP/sales@mysipproxy!sales@edvina.net
43 ; -------------------------------------------------------------
44 ; Useful CLI commands to check peers/users:
45 ; sip show peers Show all SIP peers (including friends)
46 ; sip show registry Show status of hosts we register with
48 ; sip set debug on Show all SIP messages
50 ; module reload chan_sip.so Reload configuration file
52 ;------- Naming devices ------------------------------------------------------
54 ; When naming devices, make sure you understand how Asterisk matches calls
56 ; 1. Asterisk checks the SIP From: address username and matches against
57 ; names of devices with type=user
58 ; The name is the text between square brackets [name]
59 ; 2. Asterisk checks the From: addres and matches the list of devices
61 ; 3. Asterisk checks the IP address (and port number) that the INVITE
62 ; was sent from and matches against any devices with type=peer
64 ; Don't mix extensions with the names of the devices. Devices need a unique
65 ; name. The device name is *not* used as phone numbers. Phone numbers are
66 ; anything you declare as an extension in the dialplan (extensions.conf).
68 ; When setting up trunks, make sure there's no risk that any From: username
69 ; (caller ID) will match any of your device names, because then Asterisk
70 ; might match the wrong device.
72 ; Note: The parameter "username" is not the username and in most cases is
73 ; not needed at all. Check below. In later releases, it's renamed
74 ; to "defaultuser" which is a better name, since it is used in
75 ; combination with the "defaultip" setting.
76 ;-----------------------------------------------------------------------------
78 ; ** Deprecated configuration options **
79 ; The "call-limit" configuation option is deprecated. It still works in
80 ; this version of Asterisk, but will disappear in the next version.
81 ; You are encouraged to use the dialplan groupcount functionality
82 ; to enforce call limits instead of using this channel-specific method.
84 ; You can still set limits per device in sip.conf or in a database by using
85 ; "setvar" to set variables that can be used in the dialplan for various limits.
88 context=default ; Default context for incoming calls
89 ;allowguest=no ; Allow or reject guest calls (default is yes)
90 ;match_auth_username=yes ; if available, match user entry using the
91 ; 'username' field from the authentication line
92 ; instead of the From: field.
93 allowoverlap=no ; Disable overlap dialing support. (Default is yes)
94 ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
96 ;realm=mydomain.tld ; Realm for digest authentication
97 ; defaults to "asterisk". If you set a system name in
98 ; asterisk.conf, it defaults to that system name
99 ; Realms MUST be globally unique according to RFC 3261
100 ; Set this to your host name or domain name
101 ;domainsasrealm=no ; Use domans list as realms
102 ; You can serve multiple Realms specifying several
103 ; 'domain=...' directives (see below).
104 ; In this case Realm will be based on request 'From'/'To' header
105 ; and should match one of domain names.
106 ; Otherwise default 'realm=...' will be used.
107 udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
108 ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
110 ; When a dialog is started with another SIP endpoint, the other endpoint
111 ; should include an Allow header telling us what SIP methods the endpoint
112 ; implements. However, some endpoints either do not include an Allow header
113 ; or lie about what methods they implement. In the former case, Asterisk
114 ; makes the assumption that the endpoint supports all known SIP methods.
115 ; If you know that your SIP endpoint does not provide support for a specific
116 ; method, then you may provide a comma-separated list of methods that your
117 ; endpoint does not implement in the disallowed_methods option. Note that
118 ; if your endpoint is truthful with its Allow header, then there is no need
119 ; to set this option. This option may be set in the general section or may
120 ; be set per endpoint. If this option is set both in the general section and
121 ; in a peer section, then the peer setting completely overrides the general
122 ; setting (i.e. the result is *not* the union of the two options).
124 ; Note also that while Asterisk currently will parse an Allow header to learn
125 ; what methods an endpoint supports, the only actual use for this currently
126 ; is for determining if Asterisk may send connected line UPDATE requests. Its
127 ; use may be expanded in the future.
129 ; disallowed_methods = UPDATE
132 ; Note that the TCP and TLS support for chan_sip is currently considered
133 ; experimental. Since it is new, all of the related configuration options are
134 ; subject to change in any release. If they are changed, the changes will
135 ; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
137 tcpenable=no ; Enable server for incoming TCP connections (default is no)
138 tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
139 ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
141 ;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
142 ;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
143 ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
144 ; Remember that the DNS entry for the common name (server name) in the
145 ; certificate must point to the IP address you bind to,
146 ; so you don't want to bind a TLS socket to multiple IP addresses.
149 srvlookup=yes ; Enable DNS SRV lookups on outbound calls
150 ; Note: Asterisk only uses the first host
152 ; Disabling DNS SRV lookups disables the
153 ; ability to place SIP calls based on domain
154 ; names to some other SIP users on the Internet
155 ; Specifying a port in a SIP peer definition or
156 ; when dialing outbound calls will supress SRV
157 ; lookups for that peer or call.
159 ;pedantic=yes ; Enable checking of tags in headers,
160 ; international character conversions in URIs
161 ; and multiline formatted headers for strict
162 ; SIP compatibility (defaults to "no")
164 ; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
165 ;tos_sip=cs3 ; Sets TOS for SIP packets.
166 ;tos_audio=ef ; Sets TOS for RTP audio packets.
167 ;tos_video=af41 ; Sets TOS for RTP video packets.
168 ;tos_text=af41 ; Sets TOS for RTP text packets.
170 ;cos_sip=3 ; Sets 802.1p priority for SIP packets.
171 ;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
172 ;cos_video=4 ; Sets 802.1p priority for RTP video packets.
173 ;cos_text=3 ; Sets 802.1p priority for RTP text packets.
175 ;maxexpiry=3600 ; Maximum allowed time of incoming registrations
176 ; and subscriptions (seconds)
177 ;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
178 ;defaultexpiry=120 ; Default length of incoming/outgoing registration
179 ;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
180 ;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds
181 ; Set to low value if you use low timeout for NAT of UDP sessions
183 ;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
185 ;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
187 ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
188 ;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
189 ; fully. Enable this option to not get error messages
190 ; when sending MWI to phones with this bug.
191 ;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in
192 ; the From: header as the "name" portion. Also fill the
193 ; "user" portion of the URI in the From: header with this
194 ; value if no fromuser is set
196 ;vmexten=voicemail ; dialplan extension to reach mailbox sets the
197 ; Message-Account in the MWI notify message
198 ; defaults to "asterisk"
200 ;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
201 ; rather than advertising all joint codec capabilities. This
202 ; limits the other side's codec choice to exactly what we prefer.
204 ;disallow=all ; First disallow all codecs
205 ;allow=ulaw ; Allow codecs in order of preference
206 ;allow=ilbc ; see doc/rtp-packetization for framing options
208 ; This option specifies a preference for which music on hold class this channel
209 ; should listen to when put on hold if the music class has not been set on the
210 ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
211 ; channel putting this one on hold did not suggest a music class.
213 ; This option may be specified globally, or on a per-user or per-peer basis.
215 ;mohinterpret=default
217 ; This option specifies which music on hold class to suggest to the peer channel
218 ; when this channel places the peer on hold. It may be specified globally or on
219 ; a per-user or per-peer basis.
223 ;parkinglot=plaza ; Sets the default parking lot for call parking
224 ; This may also be set for individual users/peers
225 ; Parkinglots are configured in features.conf
226 ;language=en ; Default language setting for all users/peers
227 ; This may also be set for individual users/peers
228 ;relaxdtmf=yes ; Relax dtmf handling
229 ;trustrpid = no ; If Remote-Party-ID should be trusted
230 ;sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no)
231 ;sendrpid = rpid ; Use the "Remote-Party-ID" header
232 ; to send the identity of the remote party
233 ; This is identical to sendrpid=yes
234 ;sendrpid = pai ; Use the "P-Asserted-Identity" header
235 ; to send the identity of the remote party
236 ;rpid_update = no ; In certain cases, the only method by which a connected line
237 ; change may be immediately transmitted is with a SIP UPDATE request.
238 ; If communicating with another Asterisk server, and you wish to be able
239 ; transmit such UPDATE messages to it, then you must enable this option.
240 ; Otherwise, we will have to wait until we can send a reinvite to
241 ; transmit the information.
242 ;prematuremedia=no ; Some ISDN links send empty media frames before
243 ; the call is in ringing or progress state. The SIP
244 ; channel will then send 183 indicating early media
245 ; which will be empty - thus users get no ring signal.
246 ; Setting this to "no" will stop any media before we have
247 ; call progress. Default is "yes".
249 ; In order for "noanswer" applications to work, you need to run
250 ; the progress() application in the priority before the app.
252 ;progressinband=never ; If we should generate in-band ringing always
253 ; use 'never' to never use in-band signalling, even in cases
254 ; where some buggy devices might not render it
255 ; Valid values: yes, no, never Default: never
256 ;useragent=Asterisk PBX ; Allows you to change the user agent string
257 ; The default user agent string also contains the Asterisk
258 ; version. If you don't want to expose this, change the
260 ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
261 ; Note that promiscredir when redirects are made to the
262 ; local system will cause loops since Asterisk is incapable
263 ; of performing a "hairpin" call.
264 ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
265 ; a valid phone number
266 ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
268 ; info : SIP INFO messages (application/dtmf-relay)
269 ; shortinfo : SIP INFO messages (application/dtmf)
270 ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
271 ; auto : Use rfc2833 if offered, inband otherwise
273 ;compactheaders = yes ; send compact sip headers.
275 ;videosupport=yes ; Turn on support for SIP video. You need to turn this
276 ; on in this section to get any video support at all.
277 ; You can turn it off on a per peer basis if the general
278 ; video support is enabled, but you can't enable it for
279 ; one peer only without enabling in the general section.
280 ; If you set videosupport to "always", then RTP ports will
281 ; always be set up for video, even on clients that don't
282 ; support it. This assists callfile-derived calls and
283 ; certain transferred calls to use always use video when
284 ; available. [yes|NO|always]
286 ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
287 ; Videosupport and maxcallbitrate is settable
288 ; for peers and users as well
289 ;callevents=no ; generate manager events when sip ua
290 ; performs events (e.g. hold)
291 ;authfailureevents=no ; generate manager "peerstatus" events when peer can't
292 ; authenticate with Asterisk. Peerstatus will be "rejected".
293 ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
294 ; for any reason, always reject with an identical response
295 ; equivalent to valid username and invalid password/hash
296 ; instead of letting the requester know whether there was
297 ; a matching user or peer for their request. This reduces
298 ; the ability of an attacker to scan for valid SIP usernames.
300 ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
301 ; order instead of RFC3551 packing order (this is required
302 ; for Sipura and Grandstream ATAs, among others). This is
303 ; contrary to the RFC3551 specification, the peer _should_
304 ; be negotiating AAL2-G726-32 instead :-(
305 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
306 ;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
307 ;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
308 ;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls
309 ; ; (could also be tcp,udp) - defining transports on the proxy line only
310 ; ; applies for the global proxy, otherwise use the transport= option
311 ;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
312 ; your localnet setting. Unless you have some sort of strange network
313 ; setup you will not need to enable this.
315 ;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering
316 ; as any IP address used for staticly defined
317 ; hosts. This helps avoid the configuration
318 ; error of allowing your users to register at
319 ; the same address as a SIP provider.
321 ;contactdeny=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to
322 ;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may
323 ; register their phones.
325 ;engine=asterisk ; RTP engine to use when communicating with the device
328 ; If regcontext is specified, Asterisk will dynamically create and destroy a
329 ; NoOp priority 1 extension for a given peer who registers or unregisters with
330 ; us and have a "regexten=" configuration item.
331 ; Multiple contexts may be specified by separating them with '&'. The
332 ; actual extension is the 'regexten' parameter of the registering peer or its
333 ; name if 'regexten' is not provided. If more than one context is provided,
334 ; the context must be specified within regexten by appending the desired
335 ; context after '@'. More than one regexten may be supplied if they are
336 ; separated by '&'. Patterns may be used in regexten.
338 ;regcontext=sipregistrations
339 ;regextenonqualify=yes ; Default "no"
340 ; If you have qualify on and the peer becomes unreachable
341 ; this setting will enforce inactivation of the regexten
342 ; extension for the peer
344 ; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
345 ; in square brackets. For example, the caller id value 555.5555 becomes 5555555
346 ; when this option is enabled. Disabling this option results in no modification
347 ; of the caller id value, which is necessary when the caller id represents something
348 ; that must be preserved. This option can only be used in the [general] section.
349 ; By default this option is on.
351 ;shrinkcallerid=yes ; on by default
353 ;------------------------ TLS settings ------------------------------------------------------------
354 ;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem format only) to use for TLS connections
355 ; default is to look for "asterisk.pem" in current directory
357 ;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem format only) for TLS connections.
358 ; If no tlsprivatekey is specified, tlscertfile is searched for
359 ; for both public and private key.
361 ;tlscafile=</path/to/certificate>
362 ; If the server your connecting to uses a self signed certificate
363 ; you should have their certificate installed here so the code can
364 ; verify the authenticity of their certificate.
366 ;tlscadir=</path/to/ca/dir>
367 ; A directory full of CA certificates. The files must be named with
368 ; the CA subject name hash value.
369 ; (see man SSL_CTX_load_verify_locations for more info)
371 ;tlsdontverifyserver=[yes|no]
372 ; If set to yes, don't verify the servers certificate when acting as
373 ; a client. If you don't have the server's CA certificate you can
374 ; set this and it will connect without requiring tlscafile to be set.
377 ;tlscipher=<SSL cipher string>
378 ; A string specifying which SSL ciphers to use or not use
379 ; A list of valid SSL cipher strings can be found at:
380 ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
382 ;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2.
383 ; Specify protocol for outbound client connections.
384 ; If left unspecified, the default is sslv2.
386 ;--------------------------- SIP timers ----------------------------------------------------
387 ; These timers are used primarily in INVITE transactions.
388 ; The default for Timer T1 is 500 ms or the measured run-trip time between
389 ; Asterisk and the device if you have qualify=yes for the device.
391 ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
393 ;timert1=500 ; Default T1 timer
394 ; Defaults to 500 ms or the measured round-trip
395 ; time to a peer (qualify=yes).
396 ;timerb=32000 ; Call setup timer. If a provisional response is not received
397 ; in this amount of time, the call will autocongest
398 ; Defaults to 64*timert1
400 ;--------------------------- RTP timers ----------------------------------------------------
401 ; These timers are currently used for both audio and video streams. The RTP timeouts
402 ; are only applied to the audio channel.
403 ; The settings are settable in the global section as well as per device
405 ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
406 ; on the audio channel
407 ; when we're not on hold. This is to be able to hangup
408 ; a call in the case of a phone disappearing from the net,
409 ; like a powerloss or grandma tripping over a cable.
410 ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
411 ; on the audio channel
412 ; when we're on hold (must be > rtptimeout)
413 ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
414 ; (default is off - zero)
416 ;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
417 ; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
418 ; This mechanism can detect and reclaim SIP channels that do not terminate through normal
419 ; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
420 ; The operation of Session-Timers is driven by the following configuration parameters:
422 ; * session-timers - Session-Timers feature operates in the following three modes:
423 ; originate : Request and run session-timers always
424 ; accept : Run session-timers only when requested by other UA
425 ; refuse : Do not run session timers in any case
426 ; The default mode of operation is 'accept'.
427 ; * session-expires - Maximum session refresh interval in seconds. Defaults to 1800 secs.
428 ; * session-minse - Minimum session refresh interval in seconds. Defualts to 90 secs.
429 ; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
431 ;session-timers=originate
434 ;session-refresher=uas
436 ;--------------------------- HASH TABLE SIZES ------------------------------------------------
437 ; Hash tables are used internally by the SIP driver to locate objects in memory.
438 ; For every incoming call, Asterisk will match properties of the call with in-memory
439 ; hash tables to locate a matching device, peer or user.
441 ; For maximum efficiency, adjust the following
442 ; values to be slightly larger than the maximum number of in-memory objects (devices).
443 ; Too large, and space is wasted. Too small, and things will run slower.
444 ; 563 is probably way too big for small (home) applications, but it
445 ; should cover most small/medium sites.
446 ; It is recommended to make the sizes be a prime number!
447 ; This was internally set to 17 for small-memory applications...
448 ; All tables default to 563, except when compiled in LOW_MEMORY mode,
449 ; in which case, they default to 17. You can override this by uncommenting
450 ; the following, and changing the values.
455 ;--------------------------- SIP DEBUGGING ---------------------------------------------------
456 ;sipdebug = yes ; Turn on SIP debugging by default, from
457 ; the moment the channel loads this configuration
458 ;recordhistory=yes ; Record SIP history by default
459 ; (see sip history / sip no history)
460 ;dumphistory=yes ; Dump SIP history at end of SIP dialogue
461 ; SIP history is output to the DEBUG logging channel
464 ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
465 ; You can subscribe to the status of extensions with a "hint" priority
466 ; (See extensions.conf.sample for examples)
467 ; chan_sip support two major formats for notifications: dialog-info and SIMPLE
469 ; You will get more detailed reports (busy etc) if you have a call counter enabled
472 ; If you set the busylevel, we will indicate busy when we have a number of calls that
473 ; matches the busylevel treshold.
475 ; For queues, you will need this level of detail in status reporting, regardless
476 ; if you use SIP subscriptions. Queues and manager use the same internal interface
477 ; for reading status information.
479 ; Note: Subscriptions does not work if you have a realtime dialplan and use the
482 ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
483 ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
484 ; Useful to limit subscriptions to local extensions
485 ; Settable per peer/user also
486 ;notifyringing = no ; Control whether subscriptions already INUSE get sent
487 ; RINGING when another call is sent (default: yes)
488 ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
489 ; Turning on notifyringing and notifyhold will add a lot
490 ; more database transactions if you are using realtime.
491 ;notifycid = yes ; Control whether caller ID information is sent along with
492 ; dialog-info+xml notifications (supported by snom phones).
493 ; Note that this feature will only work properly when the
494 ; incoming call is using the same extension and context that
495 ; is being used as the hint for the called extension. This means
496 ; that it won't work when using subscribecontext for your sip
497 ; user or peer (if subscribecontext is different than context).
498 ; This is also limited to a single caller, meaning that if an
499 ; extension is ringing because multiple calls are incoming,
500 ; only one will be used as the source of caller ID. Specify
501 ; 'ignore-context' to ignore the called context when looking
502 ; for the caller's channel. The default value is 'no.' Setting
503 ; notifycid to 'ignore-context' also causes call-pickups attempted
504 ; via SNOM's NOTIFY mechanism to set the context for the call pickup
506 ;callcounter = yes ; Enable call counters on devices. This can be set per
509 ;----------------------------------------- T.38 FAX SUPPORT ----------------------------------
511 ; This setting is available in the [general] section as well as in device configurations.
512 ; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off.
514 ; t38pt_udptl = yes ; Enables T.38 with FEC error correction.
515 ; t38pt_udptl = yes,fec ; Enables T.38 with FEC error correction.
516 ; t38pt_udptl = yes,redundancy ; Enables T.38 with redundancy error correction.
517 ; t38pt_udptl = yes,none ; Enables T.38 with no error correction.
519 ; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that
520 ; is based on an incorrect interpretation of the T.38 recommendation, and results in failures
521 ; because Asterisk does not believe it can send T.38 packets of a reasonable size to that
522 ; endpoint (Cisco media gateways are one example of this situation). In these cases, during a
523 ; T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL
524 ; stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you
525 ; can set an override (globally, or on a per-device basis) to make Asterisk ignore the
526 ; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead.
527 ; This can be done by appending 'maxdatagram=<value>' to the t38pt_udptl configuration option,
530 ; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error correction and overrides
531 ; ; the other endpoint's provided value to assume we can
532 ; ; send 400 byte T.38 FAX packets to it.
534 ; FAX detection will cause the SIP channel to jump to the 'fax' extension (if it exists)
535 ; after T.38 is successfully negotiated.
537 ; faxdetect = yes ; Default false
539 ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
540 ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
541 ; Format for the register statement is:
542 ; register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
549 ; - the name of a peer defined below or in realtime
550 ; The domain is where you register your username, so your SIP uri you are registering to
553 ; If no extension is given, the 's' extension is used. The extension needs to
554 ; be defined in extensions.conf to be able to accept calls from this SIP proxy
557 ; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
558 ; this is equivalent to having the following line in the general section:
560 ; register => username:secret@host/callbackextension
562 ; and more readable because you don't have to write the parameters in two places
563 ; (note that the "port" is ignored - this is a bug that should be fixed).
565 ; Note that a register= line doesn't mean that we will match the incoming call in any
566 ; other way than described above. If you want to control where the call enters your
567 ; dialplan, which context, you want to define a peer with the hostname of the provider's
568 ; server. If the provider has multiple servers to place calls to your system, you need
569 ; a peer for each server.
571 ; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may
572 ; contain a port number. Since the logical separator between a host and port number is a
573 ; ':' character, and this character is already used to separate between the optional "secret"
574 ; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish
575 ; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
576 ; they are blank. See the third example below for an illustration.
581 ;register => 1234:password@mysipprovider.com
583 ; This will pass incoming calls to the 's' extension
586 ;register => 2345:password@sip_proxy/1234
588 ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
589 ; connect to local extension 1234 in extensions.conf, default context,
590 ; unless you configure a [sip_proxy] section below, and configure a
592 ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
593 ; Tip 2: Use separate inbound and outbound sections for SIP providers
594 ; (instead of type=friend) if you have calls in both directions
596 ;register => 3456@mydomain:5082::@mysipprovider.com
598 ; Note that in this example, the optional authuser and secret portions have
599 ; been left blank because we have specified a port in the user section
601 ;registertimeout=20 ; retry registration calls every 20 seconds (default)
602 ;registerattempts=10 ; Number of registration attempts before we give up
603 ; 0 = continue forever, hammering the other server
604 ; until it accepts the registration
605 ; Default is 0 tries, continue forever
607 ;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
608 ; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
610 ; Format for the mwi register statement is:
611 ; mwi => user[:secret[:authuser]]@host[:port][/mailbox]
614 ;mwi => 1234:password@mysipprovider.com/1234
616 ; MWI received will be stored in the 1234 mailbox of the SIP_Remote context. It can be used by other phones by following the below:
617 ; mailbox=1234@SIP_Remote
618 ;----------------------------------------- NAT SUPPORT ------------------------
620 ; WARNING: SIP operation behind a NAT is tricky and you really need
621 ; to read and understand well the following section.
623 ; When Asterisk is behind a NAT device, the "local" address (and port) that
624 ; a socket is bound to has different values when seen from the inside or
625 ; from the outside of the NATted network. Unfortunately this address must
626 ; be communicated to the outside (e.g. in SIP and SDP messages), and in
627 ; order to determine the correct value Asterisk needs to know:
629 ; + whether it is talking to someone "inside" or "outside" of the NATted network.
630 ; This is configured by assigning the "localnet" parameter with a list
631 ; of network addresses that are considered "inside" of the NATted network.
632 ; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
633 ; Multiple entries are allowed, e.g. a reasonable set is the following:
635 ; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
636 ; localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
637 ; localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
638 ; localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
640 ; + the "externally visible" address and port number to be used when talking
641 ; to a host outside the NAT. This information is derived by one of the
642 ; following (mutually exclusive) config file parameters:
644 ; a. "externip = hostname[:port]" specifies a static address[:port] to
645 ; be used in SIP and SDP messages.
646 ; The hostname is looked up only once, when [re]loading sip.conf .
647 ; If a port number is not present, use the "bindport" value (which is
648 ; not guaranteed to work correctly, because a NAT box might remap the
649 ; port number as well as the address).
650 ; This approach can be useful if you have a NAT device where you can
651 ; configure the mapping statically. Examples:
653 ; externip = 12.34.56.78 ; use this address.
654 ; externip = 12.34.56.78:9900 ; use this address and port.
655 ; externip = mynat.my.org:12600 ; Public address of my nat box.
656 ; externtcpport = 9900 ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT.
657 ; ; externtcpport will default to the externip or externhost port if either one is set.
658 ; externtlsport = 12600 ; The externally mapped tls port, when Asterisk is behind a static NAT or PAT.
659 ; ; externtlsport port will default to the RFC designated port of 5061.
661 ; b. "externhost = hostname[:port]" is similar to "externip" except
662 ; that the hostname is looked up every "externrefresh" seconds
663 ; (default 10s). This can be useful when your NAT device lets you choose
664 ; the port mapping, but the IP address is dynamic.
665 ; Beware, you might suffer from service disruption when the name server
666 ; resolution fails. Examples:
668 ; externhost=foo.dyndns.net ; refreshed periodically
669 ; externrefresh=180 ; change the refresh interval
671 ; c. "stunaddr = stun.server[:port]" queries the STUN server specified
672 ; as an argument to obtain the external address/port.
673 ; Queries are also sent periodically every "externrefresh" seconds
674 ; (as a side effect, sending the query also acts as a keepalive for
675 ; the state entry on the nat box):
677 ; stunaddr = foo.stun.com:3478
680 ; Note that at the moment all these mechanism work only for the SIP socket.
681 ; The IP address discovered with externip/externhost/STUN is reused for
682 ; media sessions as well, but the port numbers are not remapped so you
683 ; may still experience problems.
685 ; NOTE 1: in some cases, NAT boxes will use different port numbers in
686 ; the internal<->external mapping. In these cases, the "externip" and
687 ; "externhost" might not help you configure addresses properly, and you
688 ; really need to use STUN.
690 ; NOTE 2: when using "externip" or "externhost", the address part is
691 ; also used as the external address for media sessions.
692 ; If you use "stunaddr", STUN queries will be sent to the same server
693 ; also from media sockets, and this should permit a correct mapping of
694 ; the port numbers as well.
696 ; In addition to the above, Asterisk has an additional "nat" parameter to
697 ; address NAT-related issues in incoming SIP or media sessions.
698 ; In particular, depending on the 'nat= ' settings described below, Asterisk
699 ; may override the address/port information specified in the SIP/SDP messages,
700 ; and use the information (sender address) supplied by the network stack instead.
701 ; However, this is only useful if the external traffic can reach us.
702 ; The following settings are allowed (both globally and in individual sections):
704 ; nat = no ; Default. Use rport if the remote side says to use it.
705 ; nat = force_rport ; Force rport to always be on.
706 ; nat = yes ; Force rport to always be on and perform symmetric RTP.
707 ; nat = comedia ; Use rport if the remote side says to use it and perform symmetric RTP.
709 ; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using
710 ; the media_address configuration option. This is only applicable to the general section and
711 ; can not be set per-user or per-peer.
713 ; media_address = 172.16.42.1
715 ;----------------------------------- MEDIA HANDLING --------------------------------
716 ; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
717 ; no reason for Asterisk to stay in the media path, the media will be redirected.
718 ; This does not really work well in the case where Asterisk is outside and the
719 ; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
721 ;directmedia=yes ; Asterisk by default tries to redirect the
722 ; RTP media stream to go directly from
723 ; the caller to the callee. Some devices do not
724 ; support this (especially if one of them is behind a NAT).
725 ; The default setting is YES. If you have all clients
726 ; behind a NAT, or for some other reason want Asterisk to
727 ; stay in the audio path, you may want to turn this off.
729 ; This setting also affect direct RTP
730 ; at call setup (a new feature in 1.4 - setting up the
731 ; call directly between the endpoints instead of sending
734 ;directmedia=nonat ; An additional option is to allow media path redirection
735 ; (reinvite) but only when the peer where the media is being
736 ; sent is known to not be behind a NAT (as the RTP core can
737 ; determine it based on the apparent IP address the media
740 ;directmedia=update ; Yet a third option... use UPDATE for media path redirection,
741 ; instead of INVITE. This can be combined with 'nonat', as
742 ; 'directmedia=update,nonat'. It implies 'yes'.
744 ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
745 ; the call directly with media peer-2-peer without re-invites.
746 ; Will not work for video and cases where the callee sends
747 ; RTP payloads and fmtp headers in the 200 OK that does not match the
748 ; callers INVITE. This will also fail if directmedia is enabled when
749 ; the device is actually behind NAT.
751 ;ignoresdpversion=yes ; By default, Asterisk will honor the session version
752 ; number in SDP packets and will only modify the SDP
753 ; session if the version number changes. This option will
754 ; force asterisk to ignore the SDP session version number
755 ; and treat all SDP data as new data. This is required
756 ; for devices that send us non standard SDP packets
757 ; (observed with Microsoft OCS). By default this option is
760 ;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
761 ; Like the useragent parameter, the default user agent string
762 ; also contains the Asterisk version.
763 ;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
764 ; This field MUST NOT contain spaces
766 ;constantssrc=yes ; Don't change the RTP SSRC when our media stream changes
768 ;----------------------------------------- REALTIME SUPPORT ------------------------
769 ; For additional information on ARA, the Asterisk Realtime Architecture,
770 ; please read realtime.txt and extconfig.txt in the /doc directory of the
773 ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
774 ; just like friends added from the config file only on a
775 ; as-needed basis? (yes|no)
777 ;rtsavesysname=yes ; Save systemname in realtime database at registration
780 ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
781 ; If set to yes, when a SIP UA registers successfully, the ip address,
782 ; the origination port, the registration period, and the username of
783 ; the UA will be set to database via realtime.
784 ; If not present, defaults to 'yes'. Note: realtime peers will
785 ; probably not function across reloads in the way that you expect, if
786 ; you turn this option off.
787 ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
788 ; as if it had just registered? (yes|no|<seconds>)
789 ; If set to yes, when the registration expires, the friend will
790 ; vanish from the configuration until requested again. If set
791 ; to an integer, friends expire within this number of seconds
792 ; instead of the registration interval.
794 ;ignoreregexpire=yes ; Enabling this setting has two functions:
796 ; For non-realtime peers, when their registration expires, the
797 ; information will _not_ be removed from memory or the Asterisk database
798 ; if you attempt to place a call to the peer, the existing information
799 ; will be used in spite of it having expired
801 ; For realtime peers, when the peer is retrieved from realtime storage,
802 ; the registration information will be used regardless of whether
803 ; it has expired or not; if it expires while the realtime peer
804 ; is still in memory (due to caching or other reasons), the
805 ; information will not be removed from realtime storage
807 ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
808 ; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
809 ; domains, each of which can direct the call to a specific context if desired.
810 ; By default, all domains are accepted and sent to the default context or the
811 ; context associated with the user/peer placing the call.
812 ; REGISTER to non-local domains will be automatically denied if a domain
813 ; list is configured.
815 ; Domains can be specified using:
816 ; domain=<domain>[,<context>]
818 ; domain=myasterisk.dom
819 ; domain=customer.com,customer-context
821 ; In addition, all the 'default' domains associated with a server should be
822 ; added if incoming request filtering is desired.
825 ; To disallow requests for domains not serviced by this server:
826 ; allowexternaldomains=no
828 ;domain=mydomain.tld,mydomain-incoming
829 ; Add domain and configure incoming context
830 ; for external calls to this domain
831 ;domain=1.2.3.4 ; Add IP address as local domain
832 ; You can have several "domain" settings
833 ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
835 ;autodomain=yes ; Turn this on to have Asterisk add local host
836 ; name and local IP to domain list.
838 ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
839 ; non-peers, use your primary domain "identity"
840 ; for From: headers instead of just your IP
841 ; address. This is to be polite and
842 ; it may be a mandatory requirement for some
843 ; destinations which do not have a prior
844 ; account relationship with your server.
846 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
847 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
848 ; SIP channel. Defaults to "no". An enabled jitterbuffer will
849 ; be used only if the sending side can create and the receiving
850 ; side can not accept jitter. The SIP channel can accept jitter,
851 ; thus a jitterbuffer on the receive SIP side will be used only
852 ; if it is forced and enabled.
854 ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
855 ; channel. Defaults to "no".
857 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
859 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
860 ; resynchronized. Useful to improve the quality of the voice, with
861 ; big jumps in/broken timestamps, usually sent from exotic devices
862 ; and programs. Defaults to 1000.
864 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
865 ; channel. Two implementations are currently available - "fixed"
866 ; (with size always equals to jbmaxsize) and "adaptive" (with
867 ; variable size, actually the new jb of IAX2). Defaults to fixed.
869 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
870 ;-----------------------------------------------------------------------------------
873 ; Global credentials for outbound calls, i.e. when a proxy challenges your
874 ; Asterisk server for authentication. These credentials override
875 ; any credentials in peer/register definition if realm is matched.
877 ; This way, Asterisk can authenticate for outbound calls to other
878 ; realms. We match realm on the proxy challenge and pick an set of
879 ; credentials from this list
881 ; auth = <user>:<secret>@<realm>
882 ; auth = <user>#<md5secret>@<realm>
884 ;auth=mark:topsecret@digium.com
886 ; You may also add auth= statements to [peer] definitions
887 ; Peer auth= override all other authentication settings if we match on realm
889 ;------------------------------------------------------------------------------
890 ; DEVICE CONFIGURATION
892 ; The SIP channel has two types of devices, the friend and the peer.
893 ; * The type=friend is a device type that accepts both incoming and outbound calls,
894 ; where Asterisk match on the From: username on incoming calls.
895 ; (A synonym for friend is "user"). This is a type you use for your local
897 ; * The type=peer also handles both incoming and outbound calls. On inbound calls,
898 ; Asterisk only matches on IP/port, not on names. This is mostly used for SIP
901 ; For device names, we recommend using only a-z, numerics (0-9) and underscore
903 ; For local phones, type=friend works most of the time
905 ; If you have one-way audio, you probably have NAT problems.
906 ; If Asterisk is on a public IP, and the phone is inside of a NAT device
907 ; you will need to configure nat option for those phones.
908 ; Also, turn on qualify=yes to keep the nat session open
910 ; Configuration options available
911 ; --------------------
974 ; contactpermit ; Limit what a host may register as (a neat trick
975 ; contactdeny ; is to register at the same IP as a SIP provider,
976 ; ; then call oneself, and get redirected to that
978 ; unsolicited_mailbox
981 ; For incoming calls only. Example: FWD (Free World Dialup)
982 ; We match on IP address of the proxy for incoming calls
983 ; since we can not match on username (caller id)
989 ;type=peer ; we only want to call out, not be called
990 ;remotesecret=guessit ; Our password to their service
991 ;defaultuser=yourusername ; Authentication user for outbound proxies
992 ;fromuser=yourusername ; Many SIP providers require this!
993 ;fromdomain=provider.sip.domain
994 ;host=box.provider.com
995 ;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will
996 ; ; accept both tcp and udp. The default transport type is only used for
997 ; ; outbound messages until a Registration takes place. During the
998 ; ; peer Registration the transport type may change to another supported
999 ; ; type if the peer requests so.
1001 ;usereqphone=yes ; This provider requires ";user=phone" on URI
1002 ;callcounter=yes ; Enable call counter
1003 ;busylevel=2 ; Signal busy at 2 or more calls
1004 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
1005 ;port=80 ; The port number we want to connect to on the remote side
1006 ; Also used as "defaultport" in combination with "defaultip" settings
1008 ;--- sample definition for a provider
1011 ;host=sip.provider1.com
1012 ;fromuser=4015552299 ; how your provider knows you
1013 ;remotesecret=youwillneverguessit ; The password we use to authenticate to them
1014 ;secret=gissadetdu ; The password they use to contact us
1015 ;callbackextension=123 ; Register with this server and require calls coming back to this extension
1016 ;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
1017 ; ; accept both tcp and udp. Default is udp. The first transport
1018 ; ; listed will always be used for outgoing connections.
1019 ;unsolicited_mailbox=4015552299 ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old
1020 ; ; message count will be stored in the configured virtual mailbox. It can be used
1021 ; ; by any device supporting MWI by specifying <configured value>@SIP_Remote as the
1025 ; Because you might have a large number of similar sections, it is generally
1026 ; convenient to use templates for the common parameters, and add them
1027 ; the the various sections. Examples are below, and we can even leave
1028 ; the templates uncommented as they will not harm:
1030 [basic-options](!) ; a template
1035 [natted-phone](!,basic-options) ; another template inheriting basic-options
1040 [public-phone](!,basic-options) ; another template inheriting basic-options
1044 [my-codecs](!) ; a template for my preferred codecs
1052 [ulaw-phone](!) ; and another one for ulaw-only
1056 ; and finally instantiate a few phones
1058 ; [2133](natted-phone,my-codecs)
1060 ; [2134](natted-phone,ulaw-phone)
1061 ; secret = not_very_secret
1062 ; [2136](public-phone,ulaw-phone)
1063 ; secret = not_very_secret_either
1067 ; Standard configurations not using templates look like this:
1071 ;context=from-sip ; Where to start in the dialplan when this phone calls
1072 ;callerid=John Doe <1234> ; Full caller ID, to override the phones config
1073 ; on incoming calls to Asterisk
1074 ;host=192.168.0.23 ; we have a static but private IP address
1075 ; No registration allowed
1076 ;nat=no ; there is not NAT between phone and Asterisk
1077 ;directmedia=yes ; allow RTP voice traffic to bypass Asterisk
1078 ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
1079 ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
1080 ; from the phone to asterisk (deprecated)
1081 ; 1 for the explicit peer, 1 for the explicit user,
1082 ; remember that a friend equals 1 peer and 1 user in
1084 ; There is no combined call counter for a "friend"
1085 ; so there's currently no way in sip.conf to limit
1086 ; to one inbound or outbound call per phone. Use
1087 ; the group counters in the dial plan for that.
1089 ;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
1090 ;disallow=all ; need to disallow=all before we can use allow=
1091 ;allow=ulaw ; Note: In user sections the order of codecs
1092 ; listed with allow= does NOT matter!
1094 ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
1095 ;allow=g729 ; Pass-thru only unless g729 license obtained
1096 ;callingpres=allowed_passed_screen ; Set caller ID presentation
1097 ; See README.callingpres for more information
1100 ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
1101 ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
1103 ;regexten=1234 ; When they register, create extension 1234
1104 ;callerid="Jane Smith" <5678>
1105 ;host=dynamic ; This device needs to register
1106 ;nat=yes ; X-Lite is behind a NAT router
1107 ;directmedia=no ; Typically set to NO if behind NAT
1109 ;allow=gsm ; GSM consumes far less bandwidth than ulaw
1112 ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
1113 ;registertrying=yes ; Send a 100 Trying when the device registers.
1116 ;type=friend ; Friends place calls and receive calls
1117 ;context=from-sip ; Context for incoming calls from this user
1119 ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
1120 ;language=de ; Use German prompts for this user
1121 ;host=dynamic ; This peer register with us
1122 ;dtmfmode=inband ; Choices are inband, rfc2833, or info
1123 ;defaultip=192.168.0.59 ; IP used until peer registers
1124 ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
1125 ;subscribemwi=yes ; Only send notifications if this phone
1126 ; subscribes for mailbox notification
1127 ;vmexten=voicemail ; dialplan extension to reach mailbox
1128 ; sets the Message-Account in the MWI notify message
1129 ; defaults to global vmexten which defaults to "asterisk"
1131 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
1135 ;type=friend ; Friends place calls and receive calls
1136 ;context=from-sip ; Context for incoming calls from this user
1138 ;host=dynamic ; This peer register with us
1139 ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
1140 ;defaultuser=polly ; Username to use in INVITE until peer registers
1141 ;defaultip=192.168.40.123
1142 ; Normally you do NOT need to set this parameter
1144 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
1145 ;progressinband=no ; Polycom phones don't work properly with "never"
1152 ;insecure=port ; Allow matching of peer by IP address without
1153 ; matching port number
1154 ;insecure=invite ; Do not require authentication of incoming INVITEs
1155 ;insecure=port,invite ; (both)
1156 ;qualify=1000 ; Consider it down if it's 1 second to reply
1157 ; Helps with NAT session
1158 ; qualify=yes uses default value
1159 ;qualifyfreq=60 ; Qualification: How often to check for the
1160 ; host to be up in seconds
1161 ; Set to low value if you use low timeout for
1162 ; NAT of UDP sessions
1164 ; Call group and Pickup group should be in the range from 0 to 63
1166 ;callgroup=1,3-4 ; We are in caller groups 1,3,4
1167 ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
1168 ;defaultip=192.168.0.60 ; IP address to use if peer has not registered
1169 ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
1170 ;permit=192.168.0.60/255.255.255.0
1175 ;qualify=200 ; Qualify peer is no more than 200ms away
1176 ;nat=yes ; This phone may be natted
1177 ; Send SIP and RTP to the IP address that packet is
1178 ; received from instead of trusting SIP headers
1179 ;host=dynamic ; This device registers with us
1180 ;directmedia=no ; Asterisk by default tries to redirect the
1181 ; RTP media stream (audio) to go directly from
1182 ; the caller to the callee. Some devices do not
1183 ; support this (especially if one of them is
1185 ;defaultip=192.168.0.4 ; IP address to use until registration
1186 ;defaultuser=goran ; Username to use when calling this device before registration
1187 ; Normally you do NOT need to set this parameter
1188 ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from or to this device
1189 ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
1190 ; cause the given audio file to
1191 ; be played upon completion of
1192 ; an attended transfer.
1198 ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
1199 ; You must have this turned on or DTMF reception will work improperly.
1200 ;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
1201 ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
1202 ; external IP address of the remote device. If port forwarding is done at the client side
1203 ; then UDPTL will flow to the remote device.
projects / asterisk/asterisk.git / blob