2 ; SIP Configuration example for Asterisk
4 ; Note: Please read the security documentation for Asterisk in order to
5 ; understand the risks of installing Asterisk with the sample
6 ; configuration. If your Asterisk is installed on a public
7 ; IP address connected to the Internet, you will want to learn
8 ; about the various security settings BEFORE you start
10 ; Especially note the following settings:
11 ; - allowguest (default enabled)
12 ; - permit/deny - IP address filters
13 ; - contactpermit/contactdeny - IP address filters for registrations
14 ; - context - Which set of services you offer various users
17 ;-----------------------------------------------------------
18 ; In the dialplan (extensions.conf) you can use several
19 ; syntaxes for dialing SIP devices.
21 ; SIP/username@domain (SIP uri)
22 ; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
23 ; SIP/devicename/extension
27 ; devicename is defined as a peer in a section below.
30 ; Call any SIP user on the Internet
31 ; (Don't forget to enable DNS SRV records if you want to use this)
33 ; devicename/extension
34 ; If you define a SIP proxy as a peer below, you may call
35 ; SIP/proxyhostname/user or SIP/user@proxyhostname
36 ; where the proxyhostname is defined in a section below
37 ; This syntax also works with ATA's with FXO ports
39 ; SIP/username[:password[:md5secret[:authname]]]@host[:port]
40 ; This form allows you to specify password or md5secret and authname
41 ; without altering any authentication data in config.
45 ; SIP/sales:topsecret::account02@domain.com:5062
46 ; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
48 ; All of these dial strings specify the SIP request URI.
49 ; In addition, you can specify a specific To: header by adding an
50 ; exclamation mark after the dial string, like
52 ; SIP/sales@mysipproxy!sales@edvina.net
55 ; -------------------------------------------------------------
56 ; Useful CLI commands to check peers/users:
57 ; sip show peers Show all SIP peers (including friends)
58 ; sip show registry Show status of hosts we register with
60 ; sip set debug on Show all SIP messages
62 ; module reload chan_sip.so Reload configuration file
64 ;------- Naming devices ------------------------------------------------------
66 ; When naming devices, make sure you understand how Asterisk matches calls
68 ; 1. Asterisk checks the SIP From: address username and matches against
69 ; names of devices with type=user
70 ; The name is the text between square brackets [name]
71 ; 2. Asterisk checks the From: addres and matches the list of devices
73 ; 3. Asterisk checks the IP address (and port number) that the INVITE
74 ; was sent from and matches against any devices with type=peer
76 ; Don't mix extensions with the names of the devices. Devices need a unique
77 ; name. The device name is *not* used as phone numbers. Phone numbers are
78 ; anything you declare as an extension in the dialplan (extensions.conf).
80 ; When setting up trunks, make sure there's no risk that any From: username
81 ; (caller ID) will match any of your device names, because then Asterisk
82 ; might match the wrong device.
84 ; Note: The parameter "username" is not the username and in most cases is
85 ; not needed at all. Check below. In later releases, it's renamed
86 ; to "defaultuser" which is a better name, since it is used in
87 ; combination with the "defaultip" setting.
88 ;-----------------------------------------------------------------------------
90 ; ** Old configuration options **
91 ; The "call-limit" configuation option is considered old is replaced
92 ; by new functionality. To enable callcounters, you use the new
93 ; "callcounter" setting (for extension states in queue and subscriptions)
94 ; You are encouraged to use the dialplan groupcount functionality
95 ; to enforce call limits instead of using this channel-specific method.
96 ; You can still set limits per device in sip.conf or in a database by using
97 ; "setvar" to set variables that can be used in the dialplan for various limits.
100 context=default ; Default context for incoming calls
101 ;allowguest=no ; Allow or reject guest calls (default is yes)
102 ; If your Asterisk is connected to the Internet
103 ; and you have allowguest=yes
104 ; you want to check which services you offer everyone
105 ; out there, by enabling them in the default context (see below).
106 ;match_auth_username=yes ; if available, match user entry using the
107 ; 'username' field from the authentication line
108 ; instead of the From: field.
109 allowoverlap=no ; Disable overlap dialing support. (Default is yes)
110 ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
112 ;realm=mydomain.tld ; Realm for digest authentication
113 ; defaults to "asterisk". If you set a system name in
114 ; asterisk.conf, it defaults to that system name
115 ; Realms MUST be globally unique according to RFC 3261
116 ; Set this to your host name or domain name
117 ;domainsasrealm=no ; Use domans list as realms
118 ; You can serve multiple Realms specifying several
119 ; 'domain=...' directives (see below).
120 ; In this case Realm will be based on request 'From'/'To' header
121 ; and should match one of domain names.
122 ; Otherwise default 'realm=...' will be used.
123 udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
124 ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
126 ; When a dialog is started with another SIP endpoint, the other endpoint
127 ; should include an Allow header telling us what SIP methods the endpoint
128 ; implements. However, some endpoints either do not include an Allow header
129 ; or lie about what methods they implement. In the former case, Asterisk
130 ; makes the assumption that the endpoint supports all known SIP methods.
131 ; If you know that your SIP endpoint does not provide support for a specific
132 ; method, then you may provide a comma-separated list of methods that your
133 ; endpoint does not implement in the disallowed_methods option. Note that
134 ; if your endpoint is truthful with its Allow header, then there is no need
135 ; to set this option. This option may be set in the general section or may
136 ; be set per endpoint. If this option is set both in the general section and
137 ; in a peer section, then the peer setting completely overrides the general
138 ; setting (i.e. the result is *not* the union of the two options).
140 ; Note also that while Asterisk currently will parse an Allow header to learn
141 ; what methods an endpoint supports, the only actual use for this currently
142 ; is for determining if Asterisk may send connected line UPDATE requests. Its
143 ; use may be expanded in the future.
145 ; disallowed_methods = UPDATE
148 ; Note that the TCP and TLS support for chan_sip is currently considered
149 ; experimental. Since it is new, all of the related configuration options are
150 ; subject to change in any release. If they are changed, the changes will
151 ; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
153 tcpenable=no ; Enable server for incoming TCP connections (default is no)
154 tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
155 ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
157 ;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
158 ;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
159 ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
160 ; Remember that the DNS entry for the common name (server name) in the
161 ; certificate must point to the IP address you bind to,
162 ; so you don't want to bind a TLS socket to multiple IP addresses.
165 srvlookup=yes ; Enable DNS SRV lookups on outbound calls
166 ; Note: Asterisk only uses the first host
168 ; Disabling DNS SRV lookups disables the
169 ; ability to place SIP calls based on domain
170 ; names to some other SIP users on the Internet
171 ; Specifying a port in a SIP peer definition or
172 ; when dialing outbound calls will supress SRV
173 ; lookups for that peer or call.
175 ;pedantic=yes ; Enable checking of tags in headers,
176 ; international character conversions in URIs
177 ; and multiline formatted headers for strict
178 ; SIP compatibility (defaults to "no")
180 ; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
181 ;tos_sip=cs3 ; Sets TOS for SIP packets.
182 ;tos_audio=ef ; Sets TOS for RTP audio packets.
183 ;tos_video=af41 ; Sets TOS for RTP video packets.
184 ;tos_text=af41 ; Sets TOS for RTP text packets.
186 ;cos_sip=3 ; Sets 802.1p priority for SIP packets.
187 ;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
188 ;cos_video=4 ; Sets 802.1p priority for RTP video packets.
189 ;cos_text=3 ; Sets 802.1p priority for RTP text packets.
191 ;maxexpiry=3600 ; Maximum allowed time of incoming registrations
192 ; and subscriptions (seconds)
193 ;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
194 ;defaultexpiry=120 ; Default length of incoming/outgoing registration
195 ;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
196 ;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds
197 ; Set to low value if you use low timeout for NAT of UDP sessions
199 ;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
201 ;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
203 ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
204 ;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
205 ; fully. Enable this option to not get error messages
206 ; when sending MWI to phones with this bug.
207 ;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in
208 ; the From: header as the "name" portion. Also fill the
209 ; "user" portion of the URI in the From: header with this
210 ; value if no fromuser is set
212 ;vmexten=voicemail ; dialplan extension to reach mailbox sets the
213 ; Message-Account in the MWI notify message
214 ; defaults to "asterisk"
216 ;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
217 ; rather than advertising all joint codec capabilities. This
218 ; limits the other side's codec choice to exactly what we prefer.
220 ;disallow=all ; First disallow all codecs
221 ;allow=ulaw ; Allow codecs in order of preference
222 ;allow=ilbc ; see doc/rtp-packetization for framing options
224 ; This option specifies a preference for which music on hold class this channel
225 ; should listen to when put on hold if the music class has not been set on the
226 ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
227 ; channel putting this one on hold did not suggest a music class.
229 ; This option may be specified globally, or on a per-user or per-peer basis.
231 ;mohinterpret=default
233 ; This option specifies which music on hold class to suggest to the peer channel
234 ; when this channel places the peer on hold. It may be specified globally or on
235 ; a per-user or per-peer basis.
239 ;parkinglot=plaza ; Sets the default parking lot for call parking
240 ; This may also be set for individual users/peers
241 ; Parkinglots are configured in features.conf
242 ;language=en ; Default language setting for all users/peers
243 ; This may also be set for individual users/peers
244 ;relaxdtmf=yes ; Relax dtmf handling
245 ;trustrpid = no ; If Remote-Party-ID should be trusted
246 ;sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no)
247 ;sendrpid = rpid ; Use the "Remote-Party-ID" header
248 ; to send the identity of the remote party
249 ; This is identical to sendrpid=yes
250 ;sendrpid = pai ; Use the "P-Asserted-Identity" header
251 ; to send the identity of the remote party
252 ;rpid_update = no ; In certain cases, the only method by which a connected line
253 ; change may be immediately transmitted is with a SIP UPDATE request.
254 ; If communicating with another Asterisk server, and you wish to be able
255 ; transmit such UPDATE messages to it, then you must enable this option.
256 ; Otherwise, we will have to wait until we can send a reinvite to
257 ; transmit the information.
258 ;prematuremedia=no ; Some ISDN links send empty media frames before
259 ; the call is in ringing or progress state. The SIP
260 ; channel will then send 183 indicating early media
261 ; which will be empty - thus users get no ring signal.
262 ; Setting this to "no" will stop any media before we have
263 ; call progress. Default is "yes".
265 ; In order for "noanswer" applications to work, you need to run
266 ; the progress() application in the priority before the app.
268 ;progressinband=never ; If we should generate in-band ringing always
269 ; use 'never' to never use in-band signalling, even in cases
270 ; where some buggy devices might not render it
271 ; Valid values: yes, no, never Default: never
272 ;useragent=Asterisk PBX ; Allows you to change the user agent string
273 ; The default user agent string also contains the Asterisk
274 ; version. If you don't want to expose this, change the
276 ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
277 ; Note that promiscredir when redirects are made to the
278 ; local system will cause loops since Asterisk is incapable
279 ; of performing a "hairpin" call.
280 ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
281 ; a valid phone number
282 ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
284 ; info : SIP INFO messages (application/dtmf-relay)
285 ; shortinfo : SIP INFO messages (application/dtmf)
286 ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
287 ; auto : Use rfc2833 if offered, inband otherwise
289 ;compactheaders = yes ; send compact sip headers.
291 ;videosupport=yes ; Turn on support for SIP video. You need to turn this
292 ; on in this section to get any video support at all.
293 ; You can turn it off on a per peer basis if the general
294 ; video support is enabled, but you can't enable it for
295 ; one peer only without enabling in the general section.
296 ; If you set videosupport to "always", then RTP ports will
297 ; always be set up for video, even on clients that don't
298 ; support it. This assists callfile-derived calls and
299 ; certain transferred calls to use always use video when
300 ; available. [yes|NO|always]
302 ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
303 ; Videosupport and maxcallbitrate is settable
304 ; for peers and users as well
305 ;callevents=no ; generate manager events when sip ua
306 ; performs events (e.g. hold)
307 ;authfailureevents=no ; generate manager "peerstatus" events when peer can't
308 ; authenticate with Asterisk. Peerstatus will be "rejected".
309 ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
310 ; for any reason, always reject with an identical response
311 ; equivalent to valid username and invalid password/hash
312 ; instead of letting the requester know whether there was
313 ; a matching user or peer for their request. This reduces
314 ; the ability of an attacker to scan for valid SIP usernames.
316 ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
317 ; order instead of RFC3551 packing order (this is required
318 ; for Sipura and Grandstream ATAs, among others). This is
319 ; contrary to the RFC3551 specification, the peer _should_
320 ; be negotiating AAL2-G726-32 instead :-(
321 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
322 ;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
323 ;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
324 ;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls
325 ; ; (could also be tcp,udp) - defining transports on the proxy line only
326 ; ; applies for the global proxy, otherwise use the transport= option
327 ;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
328 ; your localnet setting. Unless you have some sort of strange network
329 ; setup you will not need to enable this.
331 ;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering
332 ; as any IP address used for staticly defined
333 ; hosts. This helps avoid the configuration
334 ; error of allowing your users to register at
335 ; the same address as a SIP provider.
337 ;contactdeny=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to
338 ;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may
339 ; register their phones.
341 ;engine=asterisk ; RTP engine to use when communicating with the device
344 ; If regcontext is specified, Asterisk will dynamically create and destroy a
345 ; NoOp priority 1 extension for a given peer who registers or unregisters with
346 ; us and have a "regexten=" configuration item.
347 ; Multiple contexts may be specified by separating them with '&'. The
348 ; actual extension is the 'regexten' parameter of the registering peer or its
349 ; name if 'regexten' is not provided. If more than one context is provided,
350 ; the context must be specified within regexten by appending the desired
351 ; context after '@'. More than one regexten may be supplied if they are
352 ; separated by '&'. Patterns may be used in regexten.
354 ;regcontext=sipregistrations
355 ;regextenonqualify=yes ; Default "no"
356 ; If you have qualify on and the peer becomes unreachable
357 ; this setting will enforce inactivation of the regexten
358 ; extension for the peer
360 ; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
361 ; in square brackets. For example, the caller id value 555.5555 becomes 5555555
362 ; when this option is enabled. Disabling this option results in no modification
363 ; of the caller id value, which is necessary when the caller id represents something
364 ; that must be preserved. This option can only be used in the [general] section.
365 ; By default this option is on.
367 ;shrinkcallerid=yes ; on by default
370 ;use_q850_reason = no ; Default "no"
371 ; Set to yes add Reason header and use Reason header if it is available.
373 ;------------------------ TLS settings ------------------------------------------------------------
374 ;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem format only) to use for TLS connections
375 ; default is to look for "asterisk.pem" in current directory
377 ;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem format only) for TLS connections.
378 ; If no tlsprivatekey is specified, tlscertfile is searched for
379 ; for both public and private key.
381 ;tlscafile=</path/to/certificate>
382 ; If the server your connecting to uses a self signed certificate
383 ; you should have their certificate installed here so the code can
384 ; verify the authenticity of their certificate.
386 ;tlscadir=</path/to/ca/dir>
387 ; A directory full of CA certificates. The files must be named with
388 ; the CA subject name hash value.
389 ; (see man SSL_CTX_load_verify_locations for more info)
391 ;tlsdontverifyserver=[yes|no]
392 ; If set to yes, don't verify the servers certificate when acting as
393 ; a client. If you don't have the server's CA certificate you can
394 ; set this and it will connect without requiring tlscafile to be set.
397 ;tlscipher=<SSL cipher string>
398 ; A string specifying which SSL ciphers to use or not use
399 ; A list of valid SSL cipher strings can be found at:
400 ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
402 ;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2.
403 ; Specify protocol for outbound client connections.
404 ; If left unspecified, the default is sslv2.
406 ;--------------------------- SIP timers ----------------------------------------------------
407 ; These timers are used primarily in INVITE transactions.
408 ; The default for Timer T1 is 500 ms or the measured run-trip time between
409 ; Asterisk and the device if you have qualify=yes for the device.
411 ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
413 ;timert1=500 ; Default T1 timer
414 ; Defaults to 500 ms or the measured round-trip
415 ; time to a peer (qualify=yes).
416 ;timerb=32000 ; Call setup timer. If a provisional response is not received
417 ; in this amount of time, the call will autocongest
418 ; Defaults to 64*timert1
420 ;--------------------------- RTP timers ----------------------------------------------------
421 ; These timers are currently used for both audio and video streams. The RTP timeouts
422 ; are only applied to the audio channel.
423 ; The settings are settable in the global section as well as per device
425 ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
426 ; on the audio channel
427 ; when we're not on hold. This is to be able to hangup
428 ; a call in the case of a phone disappearing from the net,
429 ; like a powerloss or grandma tripping over a cable.
430 ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
431 ; on the audio channel
432 ; when we're on hold (must be > rtptimeout)
433 ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
434 ; (default is off - zero)
436 ;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
437 ; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
438 ; This mechanism can detect and reclaim SIP channels that do not terminate through normal
439 ; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
440 ; The operation of Session-Timers is driven by the following configuration parameters:
442 ; * session-timers - Session-Timers feature operates in the following three modes:
443 ; originate : Request and run session-timers always
444 ; accept : Run session-timers only when requested by other UA
445 ; refuse : Do not run session timers in any case
446 ; The default mode of operation is 'accept'.
447 ; * session-expires - Maximum session refresh interval in seconds. Defaults to 1800 secs.
448 ; * session-minse - Minimum session refresh interval in seconds. Defualts to 90 secs.
449 ; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
451 ;session-timers=originate
454 ;session-refresher=uas
456 ;--------------------------- HASH TABLE SIZES ------------------------------------------------
457 ; Hash tables are used internally by the SIP driver to locate objects in memory.
458 ; For every incoming call, Asterisk will match properties of the call with in-memory
459 ; hash tables to locate a matching device, peer or user.
461 ; For maximum efficiency, adjust the following
462 ; values to be slightly larger than the maximum number of in-memory objects (devices).
463 ; Too large, and space is wasted. Too small, and things will run slower.
464 ; 563 is probably way too big for small (home) applications, but it
465 ; should cover most small/medium sites.
466 ; It is recommended to make the sizes be a prime number!
467 ; This was internally set to 17 for small-memory applications...
468 ; All tables default to 563, except when compiled in LOW_MEMORY mode,
469 ; in which case, they default to 17. You can override this by uncommenting
470 ; the following, and changing the values.
475 ;--------------------------- SIP DEBUGGING ---------------------------------------------------
476 ;sipdebug = yes ; Turn on SIP debugging by default, from
477 ; the moment the channel loads this configuration
478 ;recordhistory=yes ; Record SIP history by default
479 ; (see sip history / sip no history)
480 ;dumphistory=yes ; Dump SIP history at end of SIP dialogue
481 ; SIP history is output to the DEBUG logging channel
484 ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
485 ; You can subscribe to the status of extensions with a "hint" priority
486 ; (See extensions.conf.sample for examples)
487 ; chan_sip support two major formats for notifications: dialog-info and SIMPLE
489 ; You will get more detailed reports (busy etc) if you have a call counter enabled
492 ; If you set the busylevel, we will indicate busy when we have a number of calls that
493 ; matches the busylevel treshold.
495 ; For queues, you will need this level of detail in status reporting, regardless
496 ; if you use SIP subscriptions. Queues and manager use the same internal interface
497 ; for reading status information.
499 ; Note: Subscriptions does not work if you have a realtime dialplan and use the
502 ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
503 ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
504 ; Useful to limit subscriptions to local extensions
505 ; Settable per peer/user also
506 ;notifyringing = no ; Control whether subscriptions already INUSE get sent
507 ; RINGING when another call is sent (default: yes)
508 ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
509 ; Turning on notifyringing and notifyhold will add a lot
510 ; more database transactions if you are using realtime.
511 ;notifycid = yes ; Control whether caller ID information is sent along with
512 ; dialog-info+xml notifications (supported by snom phones).
513 ; Note that this feature will only work properly when the
514 ; incoming call is using the same extension and context that
515 ; is being used as the hint for the called extension. This means
516 ; that it won't work when using subscribecontext for your sip
517 ; user or peer (if subscribecontext is different than context).
518 ; This is also limited to a single caller, meaning that if an
519 ; extension is ringing because multiple calls are incoming,
520 ; only one will be used as the source of caller ID. Specify
521 ; 'ignore-context' to ignore the called context when looking
522 ; for the caller's channel. The default value is 'no.' Setting
523 ; notifycid to 'ignore-context' also causes call-pickups attempted
524 ; via SNOM's NOTIFY mechanism to set the context for the call pickup
526 ;callcounter = yes ; Enable call counters on devices. This can be set per
529 ;----------------------------------------- T.38 FAX SUPPORT ----------------------------------
531 ; This setting is available in the [general] section as well as in device configurations.
532 ; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off.
534 ; t38pt_udptl = yes ; Enables T.38 with FEC error correction.
535 ; t38pt_udptl = yes,fec ; Enables T.38 with FEC error correction.
536 ; t38pt_udptl = yes,redundancy ; Enables T.38 with redundancy error correction.
537 ; t38pt_udptl = yes,none ; Enables T.38 with no error correction.
539 ; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that
540 ; is based on an incorrect interpretation of the T.38 recommendation, and results in failures
541 ; because Asterisk does not believe it can send T.38 packets of a reasonable size to that
542 ; endpoint (Cisco media gateways are one example of this situation). In these cases, during a
543 ; T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL
544 ; stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you
545 ; can set an override (globally, or on a per-device basis) to make Asterisk ignore the
546 ; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead.
547 ; This can be done by appending 'maxdatagram=<value>' to the t38pt_udptl configuration option,
550 ; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error correction and overrides
551 ; ; the other endpoint's provided value to assume we can
552 ; ; send 400 byte T.38 FAX packets to it.
554 ; FAX detection will cause the SIP channel to jump to the 'fax' extension (if it exists)
555 ; when a CNG tone is detected on an incoming call.
557 ; faxdetect = yes ; Default false
559 ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
560 ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
561 ; Format for the register statement is:
562 ; register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
569 ; - the name of a peer defined below or in realtime
570 ; The domain is where you register your username, so your SIP uri you are registering to
573 ; If no extension is given, the 's' extension is used. The extension needs to
574 ; be defined in extensions.conf to be able to accept calls from this SIP proxy
577 ; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
578 ; this is equivalent to having the following line in the general section:
580 ; register => username:secret@host/callbackextension
582 ; and more readable because you don't have to write the parameters in two places
583 ; (note that the "port" is ignored - this is a bug that should be fixed).
585 ; Note that a register= line doesn't mean that we will match the incoming call in any
586 ; other way than described above. If you want to control where the call enters your
587 ; dialplan, which context, you want to define a peer with the hostname of the provider's
588 ; server. If the provider has multiple servers to place calls to your system, you need
589 ; a peer for each server.
591 ; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may
592 ; contain a port number. Since the logical separator between a host and port number is a
593 ; ':' character, and this character is already used to separate between the optional "secret"
594 ; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish
595 ; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
596 ; they are blank. See the third example below for an illustration.
601 ;register => 1234:password@mysipprovider.com
603 ; This will pass incoming calls to the 's' extension
606 ;register => 2345:password@sip_proxy/1234
608 ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
609 ; connect to local extension 1234 in extensions.conf, default context,
610 ; unless you configure a [sip_proxy] section below, and configure a
612 ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
613 ; Tip 2: Use separate inbound and outbound sections for SIP providers
614 ; (instead of type=friend) if you have calls in both directions
616 ;register => 3456@mydomain:5082::@mysipprovider.com
618 ; Note that in this example, the optional authuser and secret portions have
619 ; been left blank because we have specified a port in the user section
621 ;registertimeout=20 ; retry registration calls every 20 seconds (default)
622 ;registerattempts=10 ; Number of registration attempts before we give up
623 ; 0 = continue forever, hammering the other server
624 ; until it accepts the registration
625 ; Default is 0 tries, continue forever
627 ;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
628 ; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
630 ; Format for the mwi register statement is:
631 ; mwi => user[:secret[:authuser]]@host[:port][/mailbox]
634 ;mwi => 1234:password@mysipprovider.com/1234
636 ; 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:
637 ; mailbox=1234@SIP_Remote
638 ;----------------------------------------- NAT SUPPORT ------------------------
640 ; WARNING: SIP operation behind a NAT is tricky and you really need
641 ; to read and understand well the following section.
643 ; When Asterisk is behind a NAT device, the "local" address (and port) that
644 ; a socket is bound to has different values when seen from the inside or
645 ; from the outside of the NATted network. Unfortunately this address must
646 ; be communicated to the outside (e.g. in SIP and SDP messages), and in
647 ; order to determine the correct value Asterisk needs to know:
649 ; + whether it is talking to someone "inside" or "outside" of the NATted network.
650 ; This is configured by assigning the "localnet" parameter with a list
651 ; of network addresses that are considered "inside" of the NATted network.
652 ; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
653 ; Multiple entries are allowed, e.g. a reasonable set is the following:
655 ; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
656 ; localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
657 ; localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
658 ; localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
660 ; + the "externally visible" address and port number to be used when talking
661 ; to a host outside the NAT. This information is derived by one of the
662 ; following (mutually exclusive) config file parameters:
664 ; a. "externip = hostname[:port]" specifies a static address[:port] to
665 ; be used in SIP and SDP messages.
666 ; The hostname is looked up only once, when [re]loading sip.conf .
667 ; If a port number is not present, use the "bindport" value (which is
668 ; not guaranteed to work correctly, because a NAT box might remap the
669 ; port number as well as the address).
670 ; This approach can be useful if you have a NAT device where you can
671 ; configure the mapping statically. Examples:
673 ; externip = 12.34.56.78 ; use this address.
674 ; externip = 12.34.56.78:9900 ; use this address and port.
675 ; externip = mynat.my.org:12600 ; Public address of my nat box.
676 ; externtcpport = 9900 ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT.
677 ; ; externtcpport will default to the externip or externhost port if either one is set.
678 ; externtlsport = 12600 ; The externally mapped tls port, when Asterisk is behind a static NAT or PAT.
679 ; ; externtlsport port will default to the RFC designated port of 5061.
681 ; b. "externhost = hostname[:port]" is similar to "externip" except
682 ; that the hostname is looked up every "externrefresh" seconds
683 ; (default 10s). This can be useful when your NAT device lets you choose
684 ; the port mapping, but the IP address is dynamic.
685 ; Beware, you might suffer from service disruption when the name server
686 ; resolution fails. Examples:
688 ; externhost=foo.dyndns.net ; refreshed periodically
689 ; externrefresh=180 ; change the refresh interval
691 ; c. "stunaddr = stun.server[:port]" queries the STUN server specified
692 ; as an argument to obtain the external address/port.
693 ; Queries are also sent periodically every "externrefresh" seconds
694 ; (as a side effect, sending the query also acts as a keepalive for
695 ; the state entry on the nat box):
697 ; stunaddr = foo.stun.com:3478
700 ; Note that at the moment all these mechanism work only for the SIP socket.
701 ; The IP address discovered with externip/externhost/STUN is reused for
702 ; media sessions as well, but the port numbers are not remapped so you
703 ; may still experience problems.
705 ; NOTE 1: in some cases, NAT boxes will use different port numbers in
706 ; the internal<->external mapping. In these cases, the "externip" and
707 ; "externhost" might not help you configure addresses properly, and you
708 ; really need to use STUN.
710 ; NOTE 2: when using "externip" or "externhost", the address part is
711 ; also used as the external address for media sessions.
712 ; If you use "stunaddr", STUN queries will be sent to the same server
713 ; also from media sockets, and this should permit a correct mapping of
714 ; the port numbers as well.
716 ; In addition to the above, Asterisk has an additional "nat" parameter to
717 ; address NAT-related issues in incoming SIP or media sessions.
718 ; In particular, depending on the 'nat= ' settings described below, Asterisk
719 ; may override the address/port information specified in the SIP/SDP messages,
720 ; and use the information (sender address) supplied by the network stack instead.
721 ; However, this is only useful if the external traffic can reach us.
722 ; The following settings are allowed (both globally and in individual sections):
724 ; nat = no ; Default. Use rport if the remote side says to use it.
725 ; nat = force_rport ; Force rport to always be on.
726 ; nat = yes ; Force rport to always be on and perform symmetric RTP.
727 ; nat = comedia ; Use rport if the remote side says to use it and perform symmetric RTP.
729 ; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using
730 ; the media_address configuration option. This is only applicable to the general section and
731 ; can not be set per-user or per-peer.
733 ; media_address = 172.16.42.1
735 ;----------------------------------- MEDIA HANDLING --------------------------------
736 ; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
737 ; no reason for Asterisk to stay in the media path, the media will be redirected.
738 ; This does not really work well in the case where Asterisk is outside and the
739 ; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
741 ;directmedia=yes ; Asterisk by default tries to redirect the
742 ; RTP media stream to go directly from
743 ; the caller to the callee. Some devices do not
744 ; support this (especially if one of them is behind a NAT).
745 ; The default setting is YES. If you have all clients
746 ; behind a NAT, or for some other reason want Asterisk to
747 ; stay in the audio path, you may want to turn this off.
749 ; This setting also affect direct RTP
750 ; at call setup (a new feature in 1.4 - setting up the
751 ; call directly between the endpoints instead of sending
754 ; Additionally this option does not disable all reINVITE operations.
755 ; It only controls Asterisk generating reINVITEs for the specific
756 ; purpose of setting up a direct media path. If a reINVITE is
757 ; needed to switch a media stream to inactive (when placed on
758 ; hold) or to T.38, it will still be done, regardless of this
761 ;directmedia=nonat ; An additional option is to allow media path redirection
762 ; (reinvite) but only when the peer where the media is being
763 ; sent is known to not be behind a NAT (as the RTP core can
764 ; determine it based on the apparent IP address the media
767 ;directmedia=update ; Yet a third option... use UPDATE for media path redirection,
768 ; instead of INVITE. This can be combined with 'nonat', as
769 ; 'directmedia=update,nonat'. It implies 'yes'.
771 ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
772 ; the call directly with media peer-2-peer without re-invites.
773 ; Will not work for video and cases where the callee sends
774 ; RTP payloads and fmtp headers in the 200 OK that does not match the
775 ; callers INVITE. This will also fail if directmedia is enabled when
776 ; the device is actually behind NAT.
778 ;ignoresdpversion=yes ; By default, Asterisk will honor the session version
779 ; number in SDP packets and will only modify the SDP
780 ; session if the version number changes. This option will
781 ; force asterisk to ignore the SDP session version number
782 ; and treat all SDP data as new data. This is required
783 ; for devices that send us non standard SDP packets
784 ; (observed with Microsoft OCS). By default this option is
787 ;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
788 ; Like the useragent parameter, the default user agent string
789 ; also contains the Asterisk version.
790 ;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
791 ; This field MUST NOT contain spaces
793 ;constantssrc=yes ; Don't change the RTP SSRC when our media stream changes
795 ;----------------------------------------- REALTIME SUPPORT ------------------------
796 ; For additional information on ARA, the Asterisk Realtime Architecture,
797 ; please read realtime.txt and extconfig.txt in the /doc directory of the
800 ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
801 ; just like friends added from the config file only on a
802 ; as-needed basis? (yes|no)
804 ;rtsavesysname=yes ; Save systemname in realtime database at registration
807 ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
808 ; If set to yes, when a SIP UA registers successfully, the ip address,
809 ; the origination port, the registration period, and the username of
810 ; the UA will be set to database via realtime.
811 ; If not present, defaults to 'yes'. Note: realtime peers will
812 ; probably not function across reloads in the way that you expect, if
813 ; you turn this option off.
814 ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
815 ; as if it had just registered? (yes|no|<seconds>)
816 ; If set to yes, when the registration expires, the friend will
817 ; vanish from the configuration until requested again. If set
818 ; to an integer, friends expire within this number of seconds
819 ; instead of the registration interval.
821 ;ignoreregexpire=yes ; Enabling this setting has two functions:
823 ; For non-realtime peers, when their registration expires, the
824 ; information will _not_ be removed from memory or the Asterisk database
825 ; if you attempt to place a call to the peer, the existing information
826 ; will be used in spite of it having expired
828 ; For realtime peers, when the peer is retrieved from realtime storage,
829 ; the registration information will be used regardless of whether
830 ; it has expired or not; if it expires while the realtime peer
831 ; is still in memory (due to caching or other reasons), the
832 ; information will not be removed from realtime storage
834 ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
835 ; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
836 ; domains, each of which can direct the call to a specific context if desired.
837 ; By default, all domains are accepted and sent to the default context or the
838 ; context associated with the user/peer placing the call.
839 ; REGISTER to non-local domains will be automatically denied if a domain
840 ; list is configured.
842 ; Domains can be specified using:
843 ; domain=<domain>[,<context>]
845 ; domain=myasterisk.dom
846 ; domain=customer.com,customer-context
848 ; In addition, all the 'default' domains associated with a server should be
849 ; added if incoming request filtering is desired.
852 ; To disallow requests for domains not serviced by this server:
853 ; allowexternaldomains=no
855 ;domain=mydomain.tld,mydomain-incoming
856 ; Add domain and configure incoming context
857 ; for external calls to this domain
858 ;domain=1.2.3.4 ; Add IP address as local domain
859 ; You can have several "domain" settings
860 ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
862 ;autodomain=yes ; Turn this on to have Asterisk add local host
863 ; name and local IP to domain list.
865 ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
866 ; non-peers, use your primary domain "identity"
867 ; for From: headers instead of just your IP
868 ; address. This is to be polite and
869 ; it may be a mandatory requirement for some
870 ; destinations which do not have a prior
871 ; account relationship with your server.
873 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
874 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
875 ; SIP channel. Defaults to "no". An enabled jitterbuffer will
876 ; be used only if the sending side can create and the receiving
877 ; side can not accept jitter. The SIP channel can accept jitter,
878 ; thus a jitterbuffer on the receive SIP side will be used only
879 ; if it is forced and enabled.
881 ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
882 ; channel. Defaults to "no".
884 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
886 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
887 ; resynchronized. Useful to improve the quality of the voice, with
888 ; big jumps in/broken timestamps, usually sent from exotic devices
889 ; and programs. Defaults to 1000.
891 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
892 ; channel. Two implementations are currently available - "fixed"
893 ; (with size always equals to jbmaxsize) and "adaptive" (with
894 ; variable size, actually the new jb of IAX2). Defaults to fixed.
896 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
897 ;-----------------------------------------------------------------------------------
900 ; Global credentials for outbound calls, i.e. when a proxy challenges your
901 ; Asterisk server for authentication. These credentials override
902 ; any credentials in peer/register definition if realm is matched.
904 ; This way, Asterisk can authenticate for outbound calls to other
905 ; realms. We match realm on the proxy challenge and pick an set of
906 ; credentials from this list
908 ; auth = <user>:<secret>@<realm>
909 ; auth = <user>#<md5secret>@<realm>
911 ;auth=mark:topsecret@digium.com
913 ; You may also add auth= statements to [peer] definitions
914 ; Peer auth= override all other authentication settings if we match on realm
916 ;------------------------------------------------------------------------------
917 ; DEVICE CONFIGURATION
919 ; The SIP channel has two types of devices, the friend and the peer.
920 ; * The type=friend is a device type that accepts both incoming and outbound calls,
921 ; where Asterisk match on the From: username on incoming calls.
922 ; (A synonym for friend is "user"). This is a type you use for your local
924 ; * The type=peer also handles both incoming and outbound calls. On inbound calls,
925 ; Asterisk only matches on IP/port, not on names. This is mostly used for SIP
928 ; For device names, we recommend using only a-z, numerics (0-9) and underscore
930 ; For local phones, type=friend works most of the time
932 ; If you have one-way audio, you probably have NAT problems.
933 ; If Asterisk is on a public IP, and the phone is inside of a NAT device
934 ; you will need to configure nat option for those phones.
935 ; Also, turn on qualify=yes to keep the nat session open
937 ; Configuration options available
938 ; --------------------
1001 ; contactpermit ; Limit what a host may register as (a neat trick
1002 ; contactdeny ; is to register at the same IP as a SIP provider,
1003 ; ; then call oneself, and get redirected to that
1005 ; unsolicited_mailbox
1009 ; For incoming calls only. Example: FWD (Free World Dialup)
1010 ; We match on IP address of the proxy for incoming calls
1011 ; since we can not match on username (caller id)
1014 ;host=fwd.pulver.com
1017 ;type=peer ; we only want to call out, not be called
1018 ;remotesecret=guessit ; Our password to their service
1019 ;defaultuser=yourusername ; Authentication user for outbound proxies
1020 ;fromuser=yourusername ; Many SIP providers require this!
1021 ;fromdomain=provider.sip.domain
1022 ;host=box.provider.com
1023 ;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will
1024 ; ; accept both tcp and udp. The default transport type is only used for
1025 ; ; outbound messages until a Registration takes place. During the
1026 ; ; peer Registration the transport type may change to another supported
1027 ; ; type if the peer requests so.
1029 ;usereqphone=yes ; This provider requires ";user=phone" on URI
1030 ;callcounter=yes ; Enable call counter
1031 ;busylevel=2 ; Signal busy at 2 or more calls
1032 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
1033 ;port=80 ; The port number we want to connect to on the remote side
1034 ; Also used as "defaultport" in combination with "defaultip" settings
1036 ;--- sample definition for a provider
1039 ;host=sip.provider1.com
1040 ;fromuser=4015552299 ; how your provider knows you
1041 ;remotesecret=youwillneverguessit ; The password we use to authenticate to them
1042 ;secret=gissadetdu ; The password they use to contact us
1043 ;callbackextension=123 ; Register with this server and require calls coming back to this extension
1044 ;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
1045 ; ; accept both tcp and udp. Default is udp. The first transport
1046 ; ; listed will always be used for outgoing connections.
1047 ;unsolicited_mailbox=4015552299 ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old
1048 ; ; message count will be stored in the configured virtual mailbox. It can be used
1049 ; ; by any device supporting MWI by specifying <configured value>@SIP_Remote as the
1053 ; Because you might have a large number of similar sections, it is generally
1054 ; convenient to use templates for the common parameters, and add them
1055 ; the the various sections. Examples are below, and we can even leave
1056 ; the templates uncommented as they will not harm:
1058 [basic-options](!) ; a template
1063 [natted-phone](!,basic-options) ; another template inheriting basic-options
1068 [public-phone](!,basic-options) ; another template inheriting basic-options
1072 [my-codecs](!) ; a template for my preferred codecs
1080 [ulaw-phone](!) ; and another one for ulaw-only
1084 ; and finally instantiate a few phones
1086 ; [2133](natted-phone,my-codecs)
1088 ; [2134](natted-phone,ulaw-phone)
1089 ; secret = not_very_secret
1090 ; [2136](public-phone,ulaw-phone)
1091 ; secret = not_very_secret_either
1095 ; Standard configurations not using templates look like this:
1099 ;context=from-sip ; Where to start in the dialplan when this phone calls
1100 ;callerid=John Doe <1234> ; Full caller ID, to override the phones config
1101 ; on incoming calls to Asterisk
1102 ;host=192.168.0.23 ; we have a static but private IP address
1103 ; No registration allowed
1104 ;nat=no ; there is not NAT between phone and Asterisk
1105 ;directmedia=yes ; allow RTP voice traffic to bypass Asterisk
1106 ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
1107 ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
1108 ; from the phone to asterisk (deprecated)
1109 ; 1 for the explicit peer, 1 for the explicit user,
1110 ; remember that a friend equals 1 peer and 1 user in
1112 ; There is no combined call counter for a "friend"
1113 ; so there's currently no way in sip.conf to limit
1114 ; to one inbound or outbound call per phone. Use
1115 ; the group counters in the dial plan for that.
1117 ;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
1118 ;disallow=all ; need to disallow=all before we can use allow=
1119 ;allow=ulaw ; Note: In user sections the order of codecs
1120 ; listed with allow= does NOT matter!
1122 ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
1123 ;allow=g729 ; Pass-thru only unless g729 license obtained
1124 ;callingpres=allowed_passed_screen ; Set caller ID presentation
1125 ; See README.callingpres for more information
1128 ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
1129 ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
1131 ;regexten=1234 ; When they register, create extension 1234
1132 ;callerid="Jane Smith" <5678>
1133 ;host=dynamic ; This device needs to register
1134 ;nat=yes ; X-Lite is behind a NAT router
1135 ;directmedia=no ; Typically set to NO if behind NAT
1137 ;allow=gsm ; GSM consumes far less bandwidth than ulaw
1140 ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
1141 ;registertrying=yes ; Send a 100 Trying when the device registers.
1144 ;type=friend ; Friends place calls and receive calls
1145 ;context=from-sip ; Context for incoming calls from this user
1147 ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
1148 ;language=de ; Use German prompts for this user
1149 ;host=dynamic ; This peer register with us
1150 ;dtmfmode=inband ; Choices are inband, rfc2833, or info
1151 ;defaultip=192.168.0.59 ; IP used until peer registers
1152 ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
1153 ;subscribemwi=yes ; Only send notifications if this phone
1154 ; subscribes for mailbox notification
1155 ;vmexten=voicemail ; dialplan extension to reach mailbox
1156 ; sets the Message-Account in the MWI notify message
1157 ; defaults to global vmexten which defaults to "asterisk"
1159 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
1163 ;type=friend ; Friends place calls and receive calls
1164 ;context=from-sip ; Context for incoming calls from this user
1166 ;host=dynamic ; This peer register with us
1167 ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
1168 ;defaultuser=polly ; Username to use in INVITE until peer registers
1169 ;defaultip=192.168.40.123
1170 ; Normally you do NOT need to set this parameter
1172 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
1173 ;progressinband=no ; Polycom phones don't work properly with "never"
1180 ;insecure=port ; Allow matching of peer by IP address without
1181 ; matching port number
1182 ;insecure=invite ; Do not require authentication of incoming INVITEs
1183 ;insecure=port,invite ; (both)
1184 ;qualify=1000 ; Consider it down if it's 1 second to reply
1185 ; Helps with NAT session
1186 ; qualify=yes uses default value
1187 ;qualifyfreq=60 ; Qualification: How often to check for the
1188 ; host to be up in seconds
1189 ; Set to low value if you use low timeout for
1190 ; NAT of UDP sessions
1192 ; Call group and Pickup group should be in the range from 0 to 63
1194 ;callgroup=1,3-4 ; We are in caller groups 1,3,4
1195 ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
1196 ;defaultip=192.168.0.60 ; IP address to use if peer has not registered
1197 ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
1198 ;permit=192.168.0.60/255.255.255.0
1203 ;qualify=200 ; Qualify peer is no more than 200ms away
1204 ;nat=yes ; This phone may be natted
1205 ; Send SIP and RTP to the IP address that packet is
1206 ; received from instead of trusting SIP headers
1207 ;host=dynamic ; This device registers with us
1208 ;directmedia=no ; Asterisk by default tries to redirect the
1209 ; RTP media stream (audio) to go directly from
1210 ; the caller to the callee. Some devices do not
1211 ; support this (especially if one of them is
1213 ;defaultip=192.168.0.4 ; IP address to use until registration
1214 ;defaultuser=goran ; Username to use when calling this device before registration
1215 ; Normally you do NOT need to set this parameter
1216 ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from or to this device
1217 ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
1218 ; cause the given audio file to
1219 ; be played upon completion of
1220 ; an attended transfer.
1226 ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
1227 ; You must have this turned on or DTMF reception will work improperly.
1228 ;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
1229 ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
1230 ; external IP address of the remote device. If port forwarding is done at the client side
1231 ; then UDPTL will flow to the remote device.
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