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
11 ; Especially note the following settings:
12 ; - allowguest (default enabled)
13 ; - permit/deny - IP address filters
14 ; - contactpermit/contactdeny - IP address filters for registrations
15 ; - context - Which set of services you offer various users
18 ;-----------------------------------------------------------
19 ; In the dialplan (extensions.conf) you can use several
20 ; syntaxes for dialing SIP devices.
22 ; SIP/username@domain (SIP uri)
23 ; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
24 ; SIP/devicename/extension
25 ; SIP/devicename/extension/IPorHost
26 ; SIP/username@domain//IPorHost
30 ; devicename is defined as a peer in a section below.
33 ; Call any SIP user on the Internet
34 ; (Don't forget to enable DNS SRV records if you want to use this)
36 ; devicename/extension
37 ; If you define a SIP proxy as a peer below, you may call
38 ; SIP/proxyhostname/user or SIP/user@proxyhostname
39 ; where the proxyhostname is defined in a section below
40 ; This syntax also works with ATA's with FXO ports
42 ; SIP/username[:password[:md5secret[:authname]]]@host[:port]
43 ; This form allows you to specify password or md5secret and authname
44 ; without altering any authentication data in config.
48 ; SIP/sales:topsecret::account02@domain.com:5062
49 ; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
52 ; The next server for this call regardless of domain/peer
54 ; All of these dial strings specify the SIP request URI.
55 ; In addition, you can specify a specific To: header by adding an
56 ; exclamation mark after the dial string, like
58 ; SIP/sales@mysipproxy!sales@edvina.net
60 ; A new feature for 1.8 allows one to specify a host or IP address to use
61 ; when routing the call. This is typically used in tandem with func_srv if
62 ; multiple methods of reaching the same domain exist. The host or IP address
63 ; is specified after the third slash in the dialstring. Examples:
65 ; SIP/devicename/extension/IPorHost
66 ; SIP/username@domain//IPorHost
69 ; -------------------------------------------------------------
70 ; Useful CLI commands to check peers/users:
71 ; sip show peers Show all SIP peers (including friends)
72 ; sip show registry Show status of hosts we register with
74 ; sip set debug on Show all SIP messages
76 ; sip reload Reload configuration file
77 ; sip show settings Show the current channel configuration
79 ;------- Naming devices ------------------------------------------------------
81 ; When naming devices, make sure you understand how Asterisk matches calls
83 ; 1. Asterisk checks the SIP From: address username and matches against
84 ; names of devices with type=user
85 ; The name is the text between square brackets [name]
86 ; 2. Asterisk checks the From: addres and matches the list of devices
88 ; 3. Asterisk checks the IP address (and port number) that the INVITE
89 ; was sent from and matches against any devices with type=peer
91 ; Don't mix extensions with the names of the devices. Devices need a unique
92 ; name. The device name is *not* used as phone numbers. Phone numbers are
93 ; anything you declare as an extension in the dialplan (extensions.conf).
95 ; When setting up trunks, make sure there's no risk that any From: username
96 ; (caller ID) will match any of your device names, because then Asterisk
97 ; might match the wrong device.
99 ; Note: The parameter "username" is not the username and in most cases is
100 ; not needed at all. Check below. In later releases, it's renamed
101 ; to "defaultuser" which is a better name, since it is used in
102 ; combination with the "defaultip" setting.
103 ;-----------------------------------------------------------------------------
105 ; ** Old configuration options **
106 ; The "call-limit" configuation option is considered old is replaced
107 ; by new functionality. To enable callcounters, you use the new
108 ; "callcounter" setting (for extension states in queue and subscriptions)
109 ; You are encouraged to use the dialplan groupcount functionality
110 ; to enforce call limits instead of using this channel-specific method.
111 ; You can still set limits per device in sip.conf or in a database by using
112 ; "setvar" to set variables that can be used in the dialplan for various limits.
115 context=default ; Default context for incoming calls
116 ;allowguest=no ; Allow or reject guest calls (default is yes)
117 ; If your Asterisk is connected to the Internet
118 ; and you have allowguest=yes
119 ; you want to check which services you offer everyone
120 ; out there, by enabling them in the default context (see below).
121 ;match_auth_username=yes ; if available, match user entry using the
122 ; 'username' field from the authentication line
123 ; instead of the From: field.
124 allowoverlap=no ; Disable overlap dialing support. (Default is yes)
125 ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
126 ; Default is enabled. The Dial() options 't' and 'T' are not
127 ; related as to whether SIP transfers are allowed or not.
128 ;realm=mydomain.tld ; Realm for digest authentication
129 ; defaults to "asterisk". If you set a system name in
130 ; asterisk.conf, it defaults to that system name
131 ; Realms MUST be globally unique according to RFC 3261
132 ; Set this to your host name or domain name
133 ;domainsasrealm=no ; Use domans list as realms
134 ; You can serve multiple Realms specifying several
135 ; 'domain=...' directives (see below).
136 ; In this case Realm will be based on request 'From'/'To' header
137 ; and should match one of domain names.
138 ; Otherwise default 'realm=...' will be used.
140 ; With the current situation, you can do one of four things:
141 ; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1
142 ; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1
143 ; c) Listen on the IPv4 wildcard. Example: bindaddr=0.0.0.0
144 ; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=::
145 ; (You can choose independently for UDP, TCP, and TLS, by specifying different values for
146 ; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".)
147 ; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat.
148 ; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)
150 ; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061
152 ; IPv4 example: bindaddr=0.0.0.0:5062
153 ; IPv6 example: bindaddr=[::]:5062
155 ; The address family of the bound UDP address is used to determine how Asterisk performs
156 ; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only
157 ; AAAA records are considered. In case d), both A and AAAA records are considered. Note,
158 ; however, that Asterisk ignores all records except the first one. In case d), when both A
159 ; and AAAA records are available, either an A or AAAA record will be first, and which one
160 ; depends on the operating system. On systems using glibc, AAAA records are given
163 udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
164 ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
166 ; When a dialog is started with another SIP endpoint, the other endpoint
167 ; should include an Allow header telling us what SIP methods the endpoint
168 ; implements. However, some endpoints either do not include an Allow header
169 ; or lie about what methods they implement. In the former case, Asterisk
170 ; makes the assumption that the endpoint supports all known SIP methods.
171 ; If you know that your SIP endpoint does not provide support for a specific
172 ; method, then you may provide a comma-separated list of methods that your
173 ; endpoint does not implement in the disallowed_methods option. Note that
174 ; if your endpoint is truthful with its Allow header, then there is no need
175 ; to set this option. This option may be set in the general section or may
176 ; be set per endpoint. If this option is set both in the general section and
177 ; in a peer section, then the peer setting completely overrides the general
178 ; setting (i.e. the result is *not* the union of the two options).
180 ; Note also that while Asterisk currently will parse an Allow header to learn
181 ; what methods an endpoint supports, the only actual use for this currently
182 ; is for determining if Asterisk may send connected line UPDATE requests. Its
183 ; use may be expanded in the future.
185 ; disallowed_methods = UPDATE
188 ; Note that the TCP and TLS support for chan_sip is currently considered
189 ; experimental. Since it is new, all of the related configuration options are
190 ; subject to change in any release. If they are changed, the changes will
191 ; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
193 tcpenable=no ; Enable server for incoming TCP connections (default is no)
194 tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
195 ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
197 ;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
198 ;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
199 ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
200 ; Remember that the IP address must match the common name (hostname) in the
201 ; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
202 ; For details how to construct a certificate for SIP see
203 ; http://tools.ietf.org/html/draft-ietf-sip-domain-certs
205 srvlookup=yes ; Enable DNS SRV lookups on outbound calls
206 ; Note: Asterisk only uses the first host
208 ; Disabling DNS SRV lookups disables the
209 ; ability to place SIP calls based on domain
210 ; names to some other SIP users on the Internet
211 ; Specifying a port in a SIP peer definition or
212 ; when dialing outbound calls will supress SRV
213 ; lookups for that peer or call.
215 ;pedantic=yes ; Enable checking of tags in headers,
216 ; international character conversions in URIs
217 ; and multiline formatted headers for strict
218 ; SIP compatibility (defaults to "yes")
220 ; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
221 ;tos_sip=cs3 ; Sets TOS for SIP packets.
222 ;tos_audio=ef ; Sets TOS for RTP audio packets.
223 ;tos_video=af41 ; Sets TOS for RTP video packets.
224 ;tos_text=af41 ; Sets TOS for RTP text packets.
226 ;cos_sip=3 ; Sets 802.1p priority for SIP packets.
227 ;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
228 ;cos_video=4 ; Sets 802.1p priority for RTP video packets.
229 ;cos_text=3 ; Sets 802.1p priority for RTP text packets.
231 ;maxexpiry=3600 ; Maximum allowed time of incoming registrations
232 ; and subscriptions (seconds)
233 ;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
234 ;defaultexpiry=120 ; Default length of incoming/outgoing registration
235 ;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
236 ;maxforwards=70 ; Setting for the SIP Max-Forwards: header (loop prevention)
237 ; Default value is 70
238 ;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds
239 ; Set to low value if you use low timeout for NAT of UDP sessions
241 ;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
243 ;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
245 ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
246 ;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
247 ; fully. Enable this option to not get error messages
248 ; when sending MWI to phones with this bug.
249 ;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in
250 ; the From: header as the "name" portion. Also fill the
251 ; "user" portion of the URI in the From: header with this
252 ; value if no fromuser is set
254 ;vmexten=voicemail ; dialplan extension to reach mailbox sets the
255 ; Message-Account in the MWI notify message
256 ; defaults to "asterisk"
260 ; When Asterisk is receiving a call, the codec will initially be set to the
261 ; first codec in the allowed codecs defined for the user receiving the call
262 ; that the caller also indicates that it supports. But, after the caller
263 ; starts sending RTP, Asterisk will switch to using whatever codec the caller
266 ; When Asterisk is placing a call, the codec used will be the first codec in
267 ; the allowed codecs that the callee indicates that it supports. Asterisk will
268 ; *not* switch to whatever codec the callee is sending.
270 ;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
271 ; rather than advertising all joint codec capabilities. This
272 ; limits the other side's codec choice to exactly what we prefer.
274 ;disallow=all ; First disallow all codecs
275 ;allow=ulaw ; Allow codecs in order of preference
276 ;allow=ilbc ; see doc/rtp-packetization for framing options
278 ; This option specifies a preference for which music on hold class this channel
279 ; should listen to when put on hold if the music class has not been set on the
280 ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
281 ; channel putting this one on hold did not suggest a music class.
283 ; This option may be specified globally, or on a per-user or per-peer basis.
285 ;mohinterpret=default
287 ; This option specifies which music on hold class to suggest to the peer channel
288 ; when this channel places the peer on hold. It may be specified globally or on
289 ; a per-user or per-peer basis.
293 ;parkinglot=plaza ; Sets the default parking lot for call parking
294 ; This may also be set for individual users/peers
295 ; Parkinglots are configured in features.conf
296 ;language=en ; Default language setting for all users/peers
297 ; This may also be set for individual users/peers
298 ;relaxdtmf=yes ; Relax dtmf handling
299 ;trustrpid = no ; If Remote-Party-ID should be trusted
300 ;sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no)
301 ;sendrpid = rpid ; Use the "Remote-Party-ID" header
302 ; to send the identity of the remote party
303 ; This is identical to sendrpid=yes
304 ;sendrpid = pai ; Use the "P-Asserted-Identity" header
305 ; to send the identity of the remote party
306 ;rpid_update = no ; In certain cases, the only method by which a connected line
307 ; change may be immediately transmitted is with a SIP UPDATE request.
308 ; If communicating with another Asterisk server, and you wish to be able
309 ; transmit such UPDATE messages to it, then you must enable this option.
310 ; Otherwise, we will have to wait until we can send a reinvite to
311 ; transmit the information.
312 ;prematuremedia=no ; Some ISDN links send empty media frames before
313 ; the call is in ringing or progress state. The SIP
314 ; channel will then send 183 indicating early media
315 ; which will be empty - thus users get no ring signal.
316 ; Setting this to "yes" will stop any media before we have
317 ; call progress (meaning the SIP channel will not send 183 Session
318 ; Progress for early media). Default is "yes". Also make sure that
319 ; the SIP peer is configured with progressinband=never.
321 ; In order for "noanswer" applications to work, you need to run
322 ; the progress() application in the priority before the app.
324 ;progressinband=never ; If we should generate in-band ringing always
325 ; use 'never' to never use in-band signalling, even in cases
326 ; where some buggy devices might not render it
327 ; Valid values: yes, no, never Default: never
328 ;useragent=Asterisk PBX ; Allows you to change the user agent string
329 ; The default user agent string also contains the Asterisk
330 ; version. If you don't want to expose this, change the
332 ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
333 ; Note that promiscredir when redirects are made to the
334 ; local system will cause loops since Asterisk is incapable
335 ; of performing a "hairpin" call.
336 ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
337 ; a valid phone number
338 ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
340 ; info : SIP INFO messages (application/dtmf-relay)
341 ; shortinfo : SIP INFO messages (application/dtmf)
342 ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
343 ; auto : Use rfc2833 if offered, inband otherwise
345 ;compactheaders = yes ; send compact sip headers.
347 ;videosupport=yes ; Turn on support for SIP video. You need to turn this
348 ; on in this section to get any video support at all.
349 ; You can turn it off on a per peer basis if the general
350 ; video support is enabled, but you can't enable it for
351 ; one peer only without enabling in the general section.
352 ; If you set videosupport to "always", then RTP ports will
353 ; always be set up for video, even on clients that don't
354 ; support it. This assists callfile-derived calls and
355 ; certain transferred calls to use always use video when
356 ; available. [yes|NO|always]
358 ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
359 ; Videosupport and maxcallbitrate is settable
360 ; for peers and users as well
361 ;callevents=no ; generate manager events when sip ua
362 ; performs events (e.g. hold)
363 ;authfailureevents=no ; generate manager "peerstatus" events when peer can't
364 ; authenticate with Asterisk. Peerstatus will be "rejected".
365 ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
366 ; for any reason, always reject with an identical response
367 ; equivalent to valid username and invalid password/hash
368 ; instead of letting the requester know whether there was
369 ; a matching user or peer for their request. This reduces
370 ; the ability of an attacker to scan for valid SIP usernames.
371 ; This option is set to "yes" by default.
373 ;auth_options_requests = yes ; Enabling this option will authenticate OPTIONS requests just like
374 ; INVITE requests are. By default this option is disabled.
376 ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
377 ; order instead of RFC3551 packing order (this is required
378 ; for Sipura and Grandstream ATAs, among others). This is
379 ; contrary to the RFC3551 specification, the peer _should_
380 ; be negotiating AAL2-G726-32 instead :-(
381 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
382 ;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
383 ;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
384 ;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls
385 ;outboundproxy=192.0.2.1 ; IPv4 address literal (default port is 5060)
386 ;outboundproxy=2001:db8::1 ; IPv6 address literal (default port is 5060)
387 ;outboundproxy=192.168.0.2.1:5062 ; IPv4 address literal with explicit port
388 ;outboundproxy=[2001:db8::1]:5062 ; IPv6 address literal with explicit port
389 ; ; (could also be tcp,udp) - defining transports on the proxy line only
390 ; ; applies for the global proxy, otherwise use the transport= option
391 ;matchexternaddrlocally = yes ; Only substitute the externaddr or externhost setting if it matches
392 ; your localnet setting. Unless you have some sort of strange network
393 ; setup you will not need to enable this.
395 ;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering
396 ; as any IP address used for staticly defined
397 ; hosts. This helps avoid the configuration
398 ; error of allowing your users to register at
399 ; the same address as a SIP provider.
401 ;contactdeny=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to
402 ;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may
403 ; register their phones.
405 ;engine=asterisk ; RTP engine to use when communicating with the device
408 ; If regcontext is specified, Asterisk will dynamically create and destroy a
409 ; NoOp priority 1 extension for a given peer who registers or unregisters with
410 ; us and have a "regexten=" configuration item.
411 ; Multiple contexts may be specified by separating them with '&'. The
412 ; actual extension is the 'regexten' parameter of the registering peer or its
413 ; name if 'regexten' is not provided. If more than one context is provided,
414 ; the context must be specified within regexten by appending the desired
415 ; context after '@'. More than one regexten may be supplied if they are
416 ; separated by '&'. Patterns may be used in regexten.
418 ;regcontext=sipregistrations
419 ;regextenonqualify=yes ; Default "no"
420 ; If you have qualify on and the peer becomes unreachable
421 ; this setting will enforce inactivation of the regexten
422 ; extension for the peer
424 ; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
425 ; in square brackets. For example, the caller id value 555.5555 becomes 5555555
426 ; when this option is enabled. Disabling this option results in no modification
427 ; of the caller id value, which is necessary when the caller id represents something
428 ; that must be preserved. This option can only be used in the [general] section.
429 ; By default this option is on.
431 ;shrinkcallerid=yes ; on by default
434 ;use_q850_reason = no ; Default "no"
435 ; Set to yes add Reason header and use Reason header if it is available.
437 ;------------------------ TLS settings ------------------------------------------------------------
438 ;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem format only) to use for TLS connections
439 ; default is to look for "asterisk.pem" in current directory
441 ;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem format only) for TLS connections.
442 ; If no tlsprivatekey is specified, tlscertfile is searched for
443 ; for both public and private key.
445 ;tlscafile=</path/to/certificate>
446 ; If the server your connecting to uses a self signed certificate
447 ; you should have their certificate installed here so the code can
448 ; verify the authenticity of their certificate.
450 ;tlscadir=</path/to/ca/dir>
451 ; A directory full of CA certificates. The files must be named with
452 ; the CA subject name hash value.
453 ; (see man SSL_CTX_load_verify_locations for more info)
455 ;tlsdontverifyserver=[yes|no]
456 ; If set to yes, don't verify the servers certificate when acting as
457 ; a client. If you don't have the server's CA certificate you can
458 ; set this and it will connect without requiring tlscafile to be set.
461 ;tlscipher=<SSL cipher string>
462 ; A string specifying which SSL ciphers to use or not use
463 ; A list of valid SSL cipher strings can be found at:
464 ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
466 ;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2.
467 ; Specify protocol for outbound client connections.
468 ; If left unspecified, the default is sslv2.
470 ;--------------------------- SIP timers ----------------------------------------------------
471 ; These timers are used primarily in INVITE transactions.
472 ; The default for Timer T1 is 500 ms or the measured run-trip time between
473 ; Asterisk and the device if you have qualify=yes for the device.
475 ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
477 ;timert1=500 ; Default T1 timer
478 ; Defaults to 500 ms or the measured round-trip
479 ; time to a peer (qualify=yes).
480 ;timerb=32000 ; Call setup timer. If a provisional response is not received
481 ; in this amount of time, the call will autocongest
482 ; Defaults to 64*timert1
484 ;--------------------------- RTP timers ----------------------------------------------------
485 ; These timers are currently used for both audio and video streams. The RTP timeouts
486 ; are only applied to the audio channel.
487 ; The settings are settable in the global section as well as per device
489 ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
490 ; on the audio channel
491 ; when we're not on hold. This is to be able to hangup
492 ; a call in the case of a phone disappearing from the net,
493 ; like a powerloss or grandma tripping over a cable.
494 ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
495 ; on the audio channel
496 ; when we're on hold (must be > rtptimeout)
497 ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
498 ; (default is off - zero)
500 ;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
501 ; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
502 ; This mechanism can detect and reclaim SIP channels that do not terminate through normal
503 ; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
504 ; The operation of Session-Timers is driven by the following configuration parameters:
506 ; * session-timers - Session-Timers feature operates in the following three modes:
507 ; originate : Request and run session-timers always
508 ; accept : Run session-timers only when requested by other UA
509 ; refuse : Do not run session timers in any case
510 ; The default mode of operation is 'accept'.
511 ; * session-expires - Maximum session refresh interval in seconds. Defaults to 1800 secs.
512 ; * session-minse - Minimum session refresh interval in seconds. Defualts to 90 secs.
513 ; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
515 ;session-timers=originate
518 ;session-refresher=uas
520 ;--------------------------- SIP DEBUGGING ---------------------------------------------------
521 ;sipdebug = yes ; Turn on SIP debugging by default, from
522 ; the moment the channel loads this configuration
523 ;recordhistory=yes ; Record SIP history by default
524 ; (see sip history / sip no history)
525 ;dumphistory=yes ; Dump SIP history at end of SIP dialogue
526 ; SIP history is output to the DEBUG logging channel
529 ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
530 ; You can subscribe to the status of extensions with a "hint" priority
531 ; (See extensions.conf.sample for examples)
532 ; chan_sip support two major formats for notifications: dialog-info and SIMPLE
534 ; You will get more detailed reports (busy etc) if you have a call counter enabled
537 ; If you set the busylevel, we will indicate busy when we have a number of calls that
538 ; matches the busylevel treshold.
540 ; For queues, you will need this level of detail in status reporting, regardless
541 ; if you use SIP subscriptions. Queues and manager use the same internal interface
542 ; for reading status information.
544 ; Note: Subscriptions does not work if you have a realtime dialplan and use the
547 ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
548 ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
549 ; Useful to limit subscriptions to local extensions
550 ; Settable per peer/user also
551 ;notifyringing = no ; Control whether subscriptions already INUSE get sent
552 ; RINGING when another call is sent (default: yes)
553 ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
554 ; Turning on notifyringing and notifyhold will add a lot
555 ; more database transactions if you are using realtime.
556 ;notifycid = yes ; Control whether caller ID information is sent along with
557 ; dialog-info+xml notifications (supported by snom phones).
558 ; Note that this feature will only work properly when the
559 ; incoming call is using the same extension and context that
560 ; is being used as the hint for the called extension. This means
561 ; that it won't work when using subscribecontext for your sip
562 ; user or peer (if subscribecontext is different than context).
563 ; This is also limited to a single caller, meaning that if an
564 ; extension is ringing because multiple calls are incoming,
565 ; only one will be used as the source of caller ID. Specify
566 ; 'ignore-context' to ignore the called context when looking
567 ; for the caller's channel. The default value is 'no.' Setting
568 ; notifycid to 'ignore-context' also causes call-pickups attempted
569 ; via SNOM's NOTIFY mechanism to set the context for the call pickup
571 ;callcounter = yes ; Enable call counters on devices. This can be set per
574 ;----------------------------------------- T.38 FAX SUPPORT ----------------------------------
576 ; This setting is available in the [general] section as well as in device configurations.
577 ; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off.
579 ; t38pt_udptl = yes ; Enables T.38 with FEC error correction.
580 ; t38pt_udptl = yes,fec ; Enables T.38 with FEC error correction.
581 ; t38pt_udptl = yes,redundancy ; Enables T.38 with redundancy error correction.
582 ; t38pt_udptl = yes,none ; Enables T.38 with no error correction.
584 ; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that
585 ; is based on an incorrect interpretation of the T.38 recommendation, and results in failures
586 ; because Asterisk does not believe it can send T.38 packets of a reasonable size to that
587 ; endpoint (Cisco media gateways are one example of this situation). In these cases, during a
588 ; T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL
589 ; stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you
590 ; can set an override (globally, or on a per-device basis) to make Asterisk ignore the
591 ; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead.
592 ; This can be done by appending 'maxdatagram=<value>' to the t38pt_udptl configuration option,
595 ; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error correction and overrides
596 ; ; the other endpoint's provided value to assume we can
597 ; ; send 400 byte T.38 FAX packets to it.
599 ; FAX detection will cause the SIP channel to jump to the 'fax' extension (if it exists)
600 ; based one or more events being detected. The events that can be detected are an incoming
601 ; CNG tone or an incoming T.38 re-INVITE request.
603 ; faxdetect = yes ; Default 'no', 'yes' enables both CNG and T.38 detection
604 ; faxdetect = cng ; Enables only CNG detection
605 ; faxdetect = t38 ; Enables only T.38 detection
607 ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
608 ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
609 ; Format for the register statement is:
610 ; register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
617 ; - the name of a peer defined below or in realtime
618 ; The domain is where you register your username, so your SIP uri you are registering to
621 ; If no extension is given, the 's' extension is used. The extension needs to
622 ; be defined in extensions.conf to be able to accept calls from this SIP proxy
625 ; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
626 ; this is equivalent to having the following line in the general section:
628 ; register => username:secret@host/callbackextension
630 ; and more readable because you don't have to write the parameters in two places
631 ; (note that the "port" is ignored - this is a bug that should be fixed).
633 ; Note that a register= line doesn't mean that we will match the incoming call in any
634 ; other way than described above. If you want to control where the call enters your
635 ; dialplan, which context, you want to define a peer with the hostname of the provider's
636 ; server. If the provider has multiple servers to place calls to your system, you need
637 ; a peer for each server.
639 ; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may
640 ; contain a port number. Since the logical separator between a host and port number is a
641 ; ':' character, and this character is already used to separate between the optional "secret"
642 ; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish
643 ; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
644 ; they are blank. See the third example below for an illustration.
649 ;register => 1234:password@mysipprovider.com
651 ; This will pass incoming calls to the 's' extension
654 ;register => 2345:password@sip_proxy/1234
656 ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
657 ; connect to local extension 1234 in extensions.conf, default context,
658 ; unless you configure a [sip_proxy] section below, and configure a
660 ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
661 ; Tip 2: Use separate inbound and outbound sections for SIP providers
662 ; (instead of type=friend) if you have calls in both directions
664 ;register => 3456@mydomain:5082::@mysipprovider.com
666 ; Note that in this example, the optional authuser and secret portions have
667 ; been left blank because we have specified a port in the user section
669 ;register => tls://username:xxxxxx@sip-tls-proxy.example.org
671 ; The 'transport' part defaults to 'udp' but may also be 'tcp' or 'tls'.
672 ; Using 'udp://' explicitly is also useful in case the username part
673 ; contains a '/' ('user/name').
675 ;registertimeout=20 ; retry registration calls every 20 seconds (default)
676 ;registerattempts=10 ; Number of registration attempts before we give up
677 ; 0 = continue forever, hammering the other server
678 ; until it accepts the registration
679 ; Default is 0 tries, continue forever
681 ;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
682 ; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
683 ; by other phones. At this time, you can only subscribe using UDP as the transport.
684 ; Format for the mwi register statement is:
685 ; mwi => user[:secret[:authuser]]@host[:port]/mailbox
688 ;mwi => 1234:password@mysipprovider.com/1234
689 ;mwi => 1234:password@myportprovider.com:6969/1234
690 ;mwi => 1234:password:authuser@myauthprovider.com/1234
691 ;mwi => 1234:password:authuser@myauthportprovider.com:6969/1234
693 ; 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:
694 ; mailbox=1234@SIP_Remote
695 ;----------------------------------------- NAT SUPPORT ------------------------
697 ; WARNING: SIP operation behind a NAT is tricky and you really need
698 ; to read and understand well the following section.
700 ; When Asterisk is behind a NAT device, the "local" address (and port) that
701 ; a socket is bound to has different values when seen from the inside or
702 ; from the outside of the NATted network. Unfortunately this address must
703 ; be communicated to the outside (e.g. in SIP and SDP messages), and in
704 ; order to determine the correct value Asterisk needs to know:
706 ; + whether it is talking to someone "inside" or "outside" of the NATted network.
707 ; This is configured by assigning the "localnet" parameter with a list
708 ; of network addresses that are considered "inside" of the NATted network.
709 ; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
710 ; Multiple entries are allowed, e.g. a reasonable set is the following:
712 ; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
713 ; localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
714 ; localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
715 ; localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
717 ; + the "externally visible" address and port number to be used when talking
718 ; to a host outside the NAT. This information is derived by one of the
719 ; following (mutually exclusive) config file parameters:
721 ; a. "externaddr = hostname[:port]" specifies a static address[:port] to
722 ; be used in SIP and SDP messages.
723 ; The hostname is looked up only once, when [re]loading sip.conf .
724 ; If a port number is not present, use the port specified in the "udpbindaddr"
725 ; (which is not guaranteed to work correctly, because a NAT box might remap the
726 ; port number as well as the address).
727 ; This approach can be useful if you have a NAT device where you can
728 ; configure the mapping statically. Examples:
730 ; externaddr = 12.34.56.78 ; use this address.
731 ; externaddr = 12.34.56.78:9900 ; use this address and port.
732 ; externaddr = mynat.my.org:12600 ; Public address of my nat box.
733 ; externtcpport = 9900 ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT.
734 ; ; externtcpport will default to the externaddr or externhost port if either one is set.
735 ; externtlsport = 12600 ; The externally mapped tls port, when Asterisk is behind a static NAT or PAT.
736 ; ; externtlsport port will default to the RFC designated port of 5061.
738 ; b. "externhost = hostname[:port]" is similar to "externaddr" except
739 ; that the hostname is looked up every "externrefresh" seconds
740 ; (default 10s). This can be useful when your NAT device lets you choose
741 ; the port mapping, but the IP address is dynamic.
742 ; Beware, you might suffer from service disruption when the name server
743 ; resolution fails. Examples:
745 ; externhost=foo.dyndns.net ; refreshed periodically
746 ; externrefresh=180 ; change the refresh interval
748 ; Note that at the moment all these mechanism work only for the SIP socket.
749 ; The IP address discovered with externaddr/externhost is reused for
750 ; media sessions as well, but the port numbers are not remapped so you
751 ; may still experience problems.
753 ; NOTE 1: in some cases, NAT boxes will use different port numbers in
754 ; the internal<->external mapping. In these cases, the "externaddr" and
755 ; "externhost" might not help you configure addresses properly.
757 ; NOTE 2: when using "externaddr" or "externhost", the address part is
758 ; also used as the external address for media sessions. Thus, the port
759 ; information in the SDP may be wrong!
761 ; In addition to the above, Asterisk has an additional "nat" parameter to
762 ; address NAT-related issues in incoming SIP or media sessions.
763 ; In particular, depending on the 'nat= ' settings described below, Asterisk
764 ; may override the address/port information specified in the SIP/SDP messages,
765 ; and use the information (sender address) supplied by the network stack instead.
766 ; However, this is only useful if the external traffic can reach us.
767 ; The following settings are allowed (both globally and in individual sections):
769 ; nat = no ; Default. Use rport if the remote side says to use it.
770 ; nat = force_rport ; Force rport to always be on.
771 ; nat = yes ; Force rport to always be on and perform comedia RTP handling.
772 ; nat = comedia ; Use rport if the remote side says to use it and perform comedia RTP handling.
774 ; 'comedia RTP handling' refers to the technique of sending RTP to the port that the
775 ; the other endpoint's RTP arrived from, and means 'connection-oriented media'. This is
776 ; only partially related to RFC 4145 which was referred to as COMEDIA while it was in
777 ; draft form. This method is used to accomodate endpoints that may be located behind
778 ; NAT devices, and as such the port number they tell Asterisk to send RTP packets to
779 ; for their media streams is not actual port number that will be used on the nearer
782 ; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by
783 ; RFC 4961; Asterisk will always send RTP packets from the same port number it expects
784 ; to receive them on.
786 ; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using
787 ; the media_address configuration option. This is only applicable to the general section and
788 ; can not be set per-user or per-peer.
790 ; media_address = 172.16.42.1
792 ; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the
793 ; perceived external network address has changed. When the stun_monitor is installed and
794 ; configured, chan_sip will renew all outbound registrations when the monitor detects any sort
795 ; of network change has occurred. By default this option is enabled, but only takes effect once
796 ; res_stun_monitor is configured. If res_stun_monitor is enabled and you wish to not
797 ; generate all outbound registrations on a network change, use the option below to disable
800 ; subscribe_network_change_event = yes ; on by default
802 ;----------------------------------- MEDIA HANDLING --------------------------------
803 ; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
804 ; no reason for Asterisk to stay in the media path, the media will be redirected.
805 ; This does not really work well in the case where Asterisk is outside and the
806 ; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
808 ;directmedia=yes ; Asterisk by default tries to redirect the
809 ; RTP media stream to go directly from
810 ; the caller to the callee. Some devices do not
811 ; support this (especially if one of them is behind a NAT).
812 ; The default setting is YES. If you have all clients
813 ; behind a NAT, or for some other reason want Asterisk to
814 ; stay in the audio path, you may want to turn this off.
816 ; This setting also affect direct RTP
817 ; at call setup (a new feature in 1.4 - setting up the
818 ; call directly between the endpoints instead of sending
821 ; Additionally this option does not disable all reINVITE operations.
822 ; It only controls Asterisk generating reINVITEs for the specific
823 ; purpose of setting up a direct media path. If a reINVITE is
824 ; needed to switch a media stream to inactive (when placed on
825 ; hold) or to T.38, it will still be done, regardless of this
826 ; setting. Note that direct T.38 is not supported.
828 ;directmedia=nonat ; An additional option is to allow media path redirection
829 ; (reinvite) but only when the peer where the media is being
830 ; sent is known to not be behind a NAT (as the RTP core can
831 ; determine it based on the apparent IP address the media
834 ;directmedia=update ; Yet a third option... use UPDATE for media path redirection,
835 ; instead of INVITE. This can be combined with 'nonat', as
836 ; 'directmedia=update,nonat'. It implies 'yes'.
838 ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
839 ; the call directly with media peer-2-peer without re-invites.
840 ; Will not work for video and cases where the callee sends
841 ; RTP payloads and fmtp headers in the 200 OK that does not match the
842 ; callers INVITE. This will also fail if directmedia is enabled when
843 ; the device is actually behind NAT.
845 ;directmediadeny=0.0.0.0/0 ; Use directmediapermit and directmediadeny to restrict
846 ;directmediapermit=172.16.0.0/16; which peers should be able to pass directmedia to each other
847 ; (There is no default setting, this is just an example)
848 ; Use this if some of your phones are on IP addresses that
849 ; can not reach each other directly. This way you can force
850 ; RTP to always flow through asterisk in such cases.
852 ;ignoresdpversion=yes ; By default, Asterisk will honor the session version
853 ; number in SDP packets and will only modify the SDP
854 ; session if the version number changes. This option will
855 ; force asterisk to ignore the SDP session version number
856 ; and treat all SDP data as new data. This is required
857 ; for devices that send us non standard SDP packets
858 ; (observed with Microsoft OCS). By default this option is
861 ;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
862 ; Like the useragent parameter, the default user agent string
863 ; also contains the Asterisk version.
864 ;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
865 ; This field MUST NOT contain spaces
867 ;----------------------------------------- REALTIME SUPPORT ------------------------
868 ; For additional information on ARA, the Asterisk Realtime Architecture,
869 ; please read realtime.txt and extconfig.txt in the /doc directory of the
872 ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
873 ; just like friends added from the config file only on a
874 ; as-needed basis? (yes|no)
876 ;rtsavesysname=yes ; Save systemname in realtime database at registration
879 ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
880 ; If set to yes, when a SIP UA registers successfully, the ip address,
881 ; the origination port, the registration period, and the username of
882 ; the UA will be set to database via realtime.
883 ; If not present, defaults to 'yes'. Note: realtime peers will
884 ; probably not function across reloads in the way that you expect, if
885 ; you turn this option off.
886 ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
887 ; as if it had just registered? (yes|no|<seconds>)
888 ; If set to yes, when the registration expires, the friend will
889 ; vanish from the configuration until requested again. If set
890 ; to an integer, friends expire within this number of seconds
891 ; instead of the registration interval.
893 ;ignoreregexpire=yes ; Enabling this setting has two functions:
895 ; For non-realtime peers, when their registration expires, the
896 ; information will _not_ be removed from memory or the Asterisk database
897 ; if you attempt to place a call to the peer, the existing information
898 ; will be used in spite of it having expired
900 ; For realtime peers, when the peer is retrieved from realtime storage,
901 ; the registration information will be used regardless of whether
902 ; it has expired or not; if it expires while the realtime peer
903 ; is still in memory (due to caching or other reasons), the
904 ; information will not be removed from realtime storage
906 ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
907 ; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
908 ; domains, each of which can direct the call to a specific context if desired.
909 ; By default, all domains are accepted and sent to the default context or the
910 ; context associated with the user/peer placing the call.
911 ; REGISTER to non-local domains will be automatically denied if a domain
912 ; list is configured.
914 ; Domains can be specified using:
915 ; domain=<domain>[,<context>]
917 ; domain=myasterisk.dom
918 ; domain=customer.com,customer-context
920 ; In addition, all the 'default' domains associated with a server should be
921 ; added if incoming request filtering is desired.
924 ; To disallow requests for domains not serviced by this server:
925 ; allowexternaldomains=no
927 ;domain=mydomain.tld,mydomain-incoming
928 ; Add domain and configure incoming context
929 ; for external calls to this domain
930 ;domain=1.2.3.4 ; Add IP address as local domain
931 ; You can have several "domain" settings
932 ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
934 ;autodomain=yes ; Turn this on to have Asterisk add local host
935 ; name and local IP to domain list.
937 ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
938 ; non-peers, use your primary domain "identity"
939 ; for From: headers instead of just your IP
940 ; address. This is to be polite and
941 ; it may be a mandatory requirement for some
942 ; destinations which do not have a prior
943 ; account relationship with your server.
945 ;------------------------------ Advice of Charge CONFIGURATION --------------------------
946 ; snom_aoc_enabled = yes; ; This options turns on and off support for sending AOC-D and
947 ; AOC-E to snom endpoints. This option can be used both in the
948 ; peer and global scope. The default for this option is off.
951 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
952 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
953 ; SIP channel. Defaults to "no". An enabled jitterbuffer will
954 ; be used only if the sending side can create and the receiving
955 ; side can not accept jitter. The SIP channel can accept jitter,
956 ; thus a jitterbuffer on the receive SIP side will be used only
957 ; if it is forced and enabled.
959 ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
960 ; channel. Defaults to "no".
962 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
964 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
965 ; resynchronized. Useful to improve the quality of the voice, with
966 ; big jumps in/broken timestamps, usually sent from exotic devices
967 ; and programs. Defaults to 1000.
969 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
970 ; channel. Two implementations are currently available - "fixed"
971 ; (with size always equals to jbmaxsize) and "adaptive" (with
972 ; variable size, actually the new jb of IAX2). Defaults to fixed.
974 ; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
975 ; The option represents the number of milliseconds by which the new jitter buffer
976 ; will pad its size. the default is 40, so without modification, the new
977 ; jitter buffer will set its size to the jitter value plus 40 milliseconds.
978 ; increasing this value may help if your network normally has low jitter,
979 ; but occasionally has spikes.
981 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
982 ;-----------------------------------------------------------------------------------
985 ; Global credentials for outbound calls, i.e. when a proxy challenges your
986 ; Asterisk server for authentication. These credentials override
987 ; any credentials in peer/register definition if realm is matched.
989 ; This way, Asterisk can authenticate for outbound calls to other
990 ; realms. We match realm on the proxy challenge and pick an set of
991 ; credentials from this list
993 ; auth = <user>:<secret>@<realm>
994 ; auth = <user>#<md5secret>@<realm>
996 ;auth=mark:topsecret@digium.com
998 ; You may also add auth= statements to [peer] definitions
999 ; Peer auth= override all other authentication settings if we match on realm
1001 ;------------------------------------------------------------------------------
1002 ; DEVICE CONFIGURATION
1004 ; The SIP channel has two types of devices, the friend and the peer.
1005 ; * The type=friend is a device type that accepts both incoming and outbound calls,
1006 ; where Asterisk match on the From: username on incoming calls.
1007 ; (A synonym for friend is "user"). This is a type you use for your local
1009 ; * The type=peer also handles both incoming and outbound calls. On inbound calls,
1010 ; Asterisk only matches on IP/port, not on names. This is mostly used for SIP
1013 ; For device names, we recommend using only a-z, numerics (0-9) and underscore
1015 ; For local phones, type=friend works most of the time
1017 ; If you have one-way audio, you probably have NAT problems.
1018 ; If Asterisk is on a public IP, and the phone is inside of a NAT device
1019 ; you will need to configure nat option for those phones.
1020 ; Also, turn on qualify=yes to keep the nat session open
1022 ; Configuration options available
1023 ; --------------------
1065 ; t38pt_usertpsource
1084 ; t38pt_usertpsource
1085 ; contactpermit ; Limit what a host may register as (a neat trick
1086 ; contactdeny ; is to register at the same IP as a SIP provider,
1087 ; ; then call oneself, and get redirected to that
1091 ; unsolicited_mailbox
1096 ; For incoming calls only. Example: FWD (Free World Dialup)
1097 ; We match on IP address of the proxy for incoming calls
1098 ; since we can not match on username (caller id)
1101 ;host=fwd.pulver.com
1104 ;type=peer ; we only want to call out, not be called
1105 ;remotesecret=guessit ; Our password to their service
1106 ;defaultuser=yourusername ; Authentication user for outbound proxies
1107 ;fromuser=yourusername ; Many SIP providers require this!
1108 ;fromdomain=provider.sip.domain
1109 ;host=box.provider.com
1110 ;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will
1111 ; ; accept both tcp and udp. The default transport type is only used for
1112 ; ; outbound messages until a Registration takes place. During the
1113 ; ; peer Registration the transport type may change to another supported
1114 ; ; type if the peer requests so.
1116 ;usereqphone=yes ; This provider requires ";user=phone" on URI
1117 ;callcounter=yes ; Enable call counter
1118 ;busylevel=2 ; Signal busy at 2 or more calls
1119 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
1120 ;port=80 ; The port number we want to connect to on the remote side
1121 ; Also used as "defaultport" in combination with "defaultip" settings
1123 ;--- sample definition for a provider
1126 ;host=sip.provider1.com
1127 ;fromuser=4015552299 ; how your provider knows you
1128 ;remotesecret=youwillneverguessit ; The password we use to authenticate to them
1129 ;secret=gissadetdu ; The password they use to contact us
1130 ;callbackextension=123 ; Register with this server and require calls coming back to this extension
1131 ;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
1132 ; ; accept both tcp and udp. Default is udp. The first transport
1133 ; ; listed will always be used for outgoing connections.
1134 ;unsolicited_mailbox=4015552299 ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old
1135 ; ; message count will be stored in the configured virtual mailbox. It can be used
1136 ; ; by any device supporting MWI by specifying <configured value>@SIP_Remote as the
1140 ; Because you might have a large number of similar sections, it is generally
1141 ; convenient to use templates for the common parameters, and add them
1142 ; the the various sections. Examples are below, and we can even leave
1143 ; the templates uncommented as they will not harm:
1145 [basic-options](!) ; a template
1150 [natted-phone](!,basic-options) ; another template inheriting basic-options
1155 [public-phone](!,basic-options) ; another template inheriting basic-options
1159 [my-codecs](!) ; a template for my preferred codecs
1167 [ulaw-phone](!) ; and another one for ulaw-only
1171 ; and finally instantiate a few phones
1173 ; [2133](natted-phone,my-codecs)
1175 ; [2134](natted-phone,ulaw-phone)
1176 ; secret = not_very_secret
1177 ; [2136](public-phone,ulaw-phone)
1178 ; secret = not_very_secret_either
1182 ; Standard configurations not using templates look like this:
1186 ;context=from-sip ; Where to start in the dialplan when this phone calls
1187 ;callerid=John Doe <1234> ; Full caller ID, to override the phones config
1188 ; on incoming calls to Asterisk
1189 ;host=192.168.0.23 ; we have a static but private IP address
1190 ; No registration allowed
1191 ;nat=no ; there is not NAT between phone and Asterisk
1192 ;directmedia=yes ; allow RTP voice traffic to bypass Asterisk
1193 ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
1194 ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
1195 ; from the phone to asterisk (deprecated)
1196 ; 1 for the explicit peer, 1 for the explicit user,
1197 ; remember that a friend equals 1 peer and 1 user in
1199 ; There is no combined call counter for a "friend"
1200 ; so there's currently no way in sip.conf to limit
1201 ; to one inbound or outbound call per phone. Use
1202 ; the group counters in the dial plan for that.
1204 ;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
1205 ;disallow=all ; need to disallow=all before we can use allow=
1206 ;allow=ulaw ; Note: In user sections the order of codecs
1207 ; listed with allow= does NOT matter!
1209 ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
1210 ;allow=g729 ; Pass-thru only unless g729 license obtained
1211 ;callingpres=allowed_passed_screen ; Set caller ID presentation
1212 ; See README.callingpres for more information
1215 ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
1216 ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
1218 ;regexten=1234 ; When they register, create extension 1234
1219 ;callerid="Jane Smith" <5678>
1220 ;host=dynamic ; This device needs to register
1221 ;nat=yes ; X-Lite is behind a NAT router
1222 ;directmedia=no ; Typically set to NO if behind NAT
1224 ;allow=gsm ; GSM consumes far less bandwidth than ulaw
1227 ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
1228 ;registertrying=yes ; Send a 100 Trying when the device registers.
1231 ;type=friend ; Friends place calls and receive calls
1232 ;context=from-sip ; Context for incoming calls from this user
1234 ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
1235 ;language=de ; Use German prompts for this user
1236 ;host=dynamic ; This peer register with us
1237 ;dtmfmode=inband ; Choices are inband, rfc2833, or info
1238 ;defaultip=192.168.0.59 ; IP used until peer registers
1239 ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
1240 ;subscribemwi=yes ; Only send notifications if this phone
1241 ; subscribes for mailbox notification
1242 ;vmexten=voicemail ; dialplan extension to reach mailbox
1243 ; sets the Message-Account in the MWI notify message
1244 ; defaults to global vmexten which defaults to "asterisk"
1246 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
1250 ;type=friend ; Friends place calls and receive calls
1251 ;context=from-sip ; Context for incoming calls from this user
1253 ;host=dynamic ; This peer register with us
1254 ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
1255 ;defaultuser=polly ; Username to use in INVITE until peer registers
1256 ;defaultip=192.168.40.123
1257 ; Normally you do NOT need to set this parameter
1259 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
1260 ;progressinband=no ; Polycom phones don't work properly with "never"
1267 ;insecure=port ; Allow matching of peer by IP address without
1268 ; matching port number
1269 ;insecure=invite ; Do not require authentication of incoming INVITEs
1270 ;insecure=port,invite ; (both)
1271 ;qualify=1000 ; Consider it down if it's 1 second to reply
1272 ; Helps with NAT session
1273 ; qualify=yes uses default value
1274 ;qualifyfreq=60 ; Qualification: How often to check for the
1275 ; host to be up in seconds
1276 ; Set to low value if you use low timeout for
1277 ; NAT of UDP sessions
1279 ; Call group and Pickup group should be in the range from 0 to 63
1281 ;callgroup=1,3-4 ; We are in caller groups 1,3,4
1282 ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
1283 ;defaultip=192.168.0.60 ; IP address to use if peer has not registered
1284 ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
1285 ;permit=192.168.0.60/255.255.255.0
1286 ;permit=192.168.0.60/24 ; we can also use CIDR notation for subnet masks
1287 ;permit=2001:db8::/32 ; IPv6 ACLs can be specified if desired. IPv6 ACLs
1288 ; apply only to IPv6 addresses, and IPv4 ACLs apply
1289 ; only to IPv4 addresses.
1294 ;qualify=200 ; Qualify peer is no more than 200ms away
1295 ;nat=yes ; This phone may be natted
1296 ; Send SIP and RTP to the IP address that packet is
1297 ; received from instead of trusting SIP headers
1298 ;host=dynamic ; This device registers with us
1299 ;directmedia=no ; Asterisk by default tries to redirect the
1300 ; RTP media stream (audio) to go directly from
1301 ; the caller to the callee. Some devices do not
1302 ; support this (especially if one of them is
1304 ;defaultip=192.168.0.4 ; IP address to use until registration
1305 ;defaultuser=goran ; Username to use when calling this device before registration
1306 ; Normally you do NOT need to set this parameter
1307 ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from or to this device
1308 ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
1309 ; cause the given audio file to
1310 ; be played upon completion of
1311 ; an attended transfer.
1317 ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
1318 ; You must have this turned on or DTMF reception will work improperly.
1319 ;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
1320 ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
1321 ; external IP address of the remote device. If port forwarding is done at the client side
1322 ; then UDPTL will flow to the remote device.
projects / asterisk/asterisk.git / blob