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/acl - IP address filters
14 ; - contactpermit/contactdeny/contactacl - 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=public ; Default context for incoming calls. Defaults to 'default'
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 ;allowoverlap=yes ; Enable RFC3578 overlap dialing support.
126 ; Can use the Incomplete application to collect the
127 ; needed digits from an ambiguous dialplan match.
128 ;allowoverlap=dtmf ; Enable overlap dialing support using DTMF delivery
129 ; methods (inband, RFC2833, SIP INFO) in the early
130 ; media phase. Uses the Incomplete application to
131 ; collect the needed digits.
132 ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
133 ; Default is enabled. The Dial() options 't' and 'T' are not
134 ; related as to whether SIP transfers are allowed or not.
135 ;realm=mydomain.tld ; Realm for digest authentication
136 ; defaults to "asterisk". If you set a system name in
137 ; asterisk.conf, it defaults to that system name
138 ; Realms MUST be globally unique according to RFC 3261
139 ; Set this to your host name or domain name
140 ;domainsasrealm=no ; Use domains list as realms
141 ; You can serve multiple Realms specifying several
142 ; 'domain=...' directives (see below).
143 ; In this case Realm will be based on request 'From'/'To' header
144 ; and should match one of domain names.
145 ; Otherwise default 'realm=...' will be used.
146 ;recordonfeature=automixmon ; Default feature to use when receiving 'Record: on' header
147 ; from an INFO message. Defaults to 'automon'. Works with
148 ; dynamic features. Feature must be usable on requesting
149 ; channel for it to work. Setting this value to a blank
151 ;recordofffeature=automixmon ; Default feature to use when receiving 'Record: off' header
152 ; from an INFO message. Defaults to 'automon'. Works with
153 ; dynamic features. Feature must be usable on requesting
154 ; channel for it to work. Setting this value to a blank
157 ; With the current situation, you can do one of four things:
158 ; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1
159 ; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1
160 ; c) Listen on the IPv4 wildcard. Example: bindaddr=0.0.0.0
161 ; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=::
162 ; (You can choose independently for UDP, TCP, and TLS, by specifying different values for
163 ; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".)
164 ; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat.
165 ; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)
167 ; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061
169 ; IPv4 example: bindaddr=0.0.0.0:5062
170 ; IPv6 example: bindaddr=[::]:5062
172 ; The address family of the bound UDP address is used to determine how Asterisk performs
173 ; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only
174 ; AAAA records are considered. In case d), both A and AAAA records are considered. Note,
175 ; however, that Asterisk ignores all records except the first one. In case d), when both A
176 ; and AAAA records are available, either an A or AAAA record will be first, and which one
177 ; depends on the operating system. On systems using glibc, AAAA records are given
180 udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
181 ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
183 ; When a dialog is started with another SIP endpoint, the other endpoint
184 ; should include an Allow header telling us what SIP methods the endpoint
185 ; implements. However, some endpoints either do not include an Allow header
186 ; or lie about what methods they implement. In the former case, Asterisk
187 ; makes the assumption that the endpoint supports all known SIP methods.
188 ; If you know that your SIP endpoint does not provide support for a specific
189 ; method, then you may provide a comma-separated list of methods that your
190 ; endpoint does not implement in the disallowed_methods option. Note that
191 ; if your endpoint is truthful with its Allow header, then there is no need
192 ; to set this option. This option may be set in the general section or may
193 ; be set per endpoint. If this option is set both in the general section and
194 ; in a peer section, then the peer setting completely overrides the general
195 ; setting (i.e. the result is *not* the union of the two options).
197 ; Note also that while Asterisk currently will parse an Allow header to learn
198 ; what methods an endpoint supports, the only actual use for this currently
199 ; is for determining if Asterisk may send connected line UPDATE requests and
200 ; MESSAGE requests. Its use may be expanded in the future.
202 ; disallowed_methods = UPDATE
205 ; Note that the TCP and TLS support for chan_sip is currently considered
206 ; experimental. Since it is new, all of the related configuration options are
207 ; subject to change in any release. If they are changed, the changes will
208 ; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
210 tcpenable=no ; Enable server for incoming TCP connections (default is no)
211 tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
212 ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
214 ;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
215 ;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
216 ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
217 ; Remember that the IP address must match the common name (hostname) in the
218 ; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
219 ; For details how to construct a certificate for SIP see
220 ; http://tools.ietf.org/html/draft-ietf-sip-domain-certs
222 ;tcpauthtimeout = 30 ; tcpauthtimeout specifies the maximum number
223 ; of seconds a client has to authenticate. If
224 ; the client does not authenticate beofre this
225 ; timeout expires, the client will be
226 ; disconnected. (default: 30 seconds)
228 ;tcpauthlimit = 100 ; tcpauthlimit specifies the maximum number of
229 ; unauthenticated sessions that will be allowed
230 ; to connect at any given time. (default: 100)
232 transport=udp ; Set the default transports. The order determines the primary default transport.
233 ; If tcpenable=no and the transport set is tcp, we will fallback to UDP.
235 srvlookup=yes ; Enable DNS SRV lookups on outbound calls
236 ; Note: Asterisk only uses the first host
238 ; Disabling DNS SRV lookups disables the
239 ; ability to place SIP calls based on domain
240 ; names to some other SIP users on the Internet
241 ; Specifying a port in a SIP peer definition or
242 ; when dialing outbound calls will supress SRV
243 ; lookups for that peer or call.
245 ;pedantic=yes ; Enable checking of tags in headers,
246 ; international character conversions in URIs
247 ; and multiline formatted headers for strict
248 ; SIP compatibility (defaults to "yes")
250 ; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
251 ;tos_sip=cs3 ; Sets TOS for SIP packets.
252 ;tos_audio=ef ; Sets TOS for RTP audio packets.
253 ;tos_video=af41 ; Sets TOS for RTP video packets.
254 ;tos_text=af41 ; Sets TOS for RTP text packets.
256 ;cos_sip=3 ; Sets 802.1p priority for SIP packets.
257 ;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
258 ;cos_video=4 ; Sets 802.1p priority for RTP video packets.
259 ;cos_text=3 ; Sets 802.1p priority for RTP text packets.
261 ;maxexpiry=3600 ; Maximum allowed time of incoming registrations (seconds)
262 ;minexpiry=60 ; Minimum length of registrations (default 60)
263 ;defaultexpiry=120 ; Default length of incoming/outgoing registration
264 ;submaxexpiry=3600 ; Maximum allowed time of incoming subscriptions (seconds), default: maxexpiry
265 ;subminexpiry=60 ; Minimum length of subscriptions, default: minexpiry
266 ;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
267 ;maxforwards=70 ; Setting for the SIP Max-Forwards: header (loop prevention)
268 ; Default value is 70
269 ;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds
270 ; and reported in milliseconds with sip show settings.
271 ; Set to low value if you use low timeout for NAT of UDP sessions
273 ;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
275 ;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
277 ;keepalive=60 ; Interval at which keepalive packets should be sent to a peer
278 ; Valid options are yes (60 seconds), no, or the number of seconds.
280 ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
281 ;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
282 ; fully. Enable this option to not get error messages
283 ; when sending MWI to phones with this bug.
284 ;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in
285 ; the From: header as the "name" portion. Also fill the
286 ; "user" portion of the URI in the From: header with this
287 ; value if no fromuser is set
289 ;vmexten=voicemail ; dialplan extension to reach mailbox sets the
290 ; Message-Account in the MWI notify message
291 ; defaults to "asterisk"
295 ; When Asterisk is receiving a call, the codec will initially be set to the
296 ; first codec in the allowed codecs defined for the user receiving the call
297 ; that the caller also indicates that it supports. But, after the caller
298 ; starts sending RTP, Asterisk will switch to using whatever codec the caller
301 ; When Asterisk is placing a call, the codec used will be the first codec in
302 ; the allowed codecs that the callee indicates that it supports. Asterisk will
303 ; *not* switch to whatever codec the callee is sending.
305 ;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
306 ; rather than advertising all joint codec capabilities. This
307 ; limits the other side's codec choice to exactly what we prefer.
309 ;disallow=all ; First disallow all codecs
310 ;allow=ulaw ; Allow codecs in order of preference
311 ;allow=ilbc ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
312 ; for framing options
313 ;autoframing=yes ; Set packetization based on the remote endpoint's (ptime)
314 ; preferences. Defaults to no.
316 ; This option specifies a preference for which music on hold class this channel
317 ; should listen to when put on hold if the music class has not been set on the
318 ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
319 ; channel putting this one on hold did not suggest a music class.
321 ; This option may be specified globally, or on a per-user or per-peer basis.
323 ;mohinterpret=default
325 ; This option specifies which music on hold class to suggest to the peer channel
326 ; when this channel places the peer on hold. It may be specified globally or on
327 ; a per-user or per-peer basis.
331 ;parkinglot=plaza ; Sets the default parking lot for call parking
332 ; This may also be set for individual users/peers
333 ; Parkinglots are configured in features.conf
334 ;language=en ; Default language setting for all users/peers
335 ; This may also be set for individual users/peers
336 ;tonezone=se ; Default tonezone for all users/peers
337 ; This may also be set for individual users/peers
339 ;relaxdtmf=yes ; Relax dtmf handling
340 ;trustrpid = no ; If Remote-Party-ID should be trusted
341 ;sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no)
342 ;sendrpid = rpid ; Use the "Remote-Party-ID" header
343 ; to send the identity of the remote party
344 ; This is identical to sendrpid=yes
345 ;sendrpid = pai ; Use the "P-Asserted-Identity" header
346 ; to send the identity of the remote party
347 ;rpid_update = no ; In certain cases, the only method by which a connected line
348 ; change may be immediately transmitted is with a SIP UPDATE request.
349 ; If communicating with another Asterisk server, and you wish to be able
350 ; transmit such UPDATE messages to it, then you must enable this option.
351 ; Otherwise, we will have to wait until we can send a reinvite to
352 ; transmit the information.
353 ;trust_id_outbound = no ; Controls whether or not we trust this peer with private identity
354 ; information (when the remote party has callingpres=prohib or equivalent).
355 ; no - RPID/PAI headers will not be included for private peer information
356 ; yes - RPID/PAI headers will include the private peer information. Privacy
357 ; requirements will be indicated in a Privacy header for sendrpid=pai
358 ; legacy - RPID/PAI will be included for private peer information. In the
359 ; case of sendrpid=pai, private data that would be included in them
360 ; will be anonymized. For sendrpid=rpid, private data may be included
361 ; but the remote party's domain will be anonymized. The way legacy
362 ; behaves may violate RFC-3325, but it follows historic behavior.
363 ; This option is set to 'legacy' by default
364 ;prematuremedia=no ; Some ISDN links send empty media frames before
365 ; the call is in ringing or progress state. The SIP
366 ; channel will then send 183 indicating early media
367 ; which will be empty - thus users get no ring signal.
368 ; Setting this to "yes" will stop any media before we have
369 ; call progress (meaning the SIP channel will not send 183 Session
370 ; Progress for early media). Default is "yes". Also make sure that
371 ; the SIP peer is configured with progressinband=never.
373 ; In order for "noanswer" applications to work, you need to run
374 ; the progress() application in the priority before the app.
376 ;progressinband=never ; If we should generate in-band ringing always
377 ; use 'never' to never use in-band signalling, even in cases
378 ; where some buggy devices might not render it
379 ; Valid values: yes, no, never Default: never
380 ;useragent=Asterisk PBX ; Allows you to change the user agent string
381 ; The default user agent string also contains the Asterisk
382 ; version. If you don't want to expose this, change the
384 ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
385 ; Note that promiscredir when redirects are made to the
386 ; local system will cause loops since Asterisk is incapable
387 ; of performing a "hairpin" call.
388 ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
389 ; a valid phone number
390 ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
392 ; info : SIP INFO messages (application/dtmf-relay)
393 ; shortinfo : SIP INFO messages (application/dtmf)
394 ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
395 ; auto : Use rfc2833 if offered, inband otherwise
397 ;compactheaders = yes ; send compact sip headers.
399 ;videosupport=yes ; Turn on support for SIP video. You need to turn this
400 ; on in this section to get any video support at all.
401 ; You can turn it off on a per peer basis if the general
402 ; video support is enabled, but you can't enable it for
403 ; one peer only without enabling in the general section.
404 ; If you set videosupport to "always", then RTP ports will
405 ; always be set up for video, even on clients that don't
406 ; support it. This assists callfile-derived calls and
407 ; certain transferred calls to use always use video when
408 ; available. [yes|NO|always]
410 ;textsupport=no ; Support for ITU-T T.140 realtime text.
411 ; The default value is "no".
413 ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
414 ; Videosupport and maxcallbitrate is settable
415 ; for peers and users as well
416 ;authfailureevents=no ; generate manager "peerstatus" events when peer can't
417 ; authenticate with Asterisk. Peerstatus will be "rejected".
418 ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
419 ; for any reason, always reject with an identical response
420 ; equivalent to valid username and invalid password/hash
421 ; instead of letting the requester know whether there was
422 ; a matching user or peer for their request. This reduces
423 ; the ability of an attacker to scan for valid SIP usernames.
424 ; This option is set to "yes" by default.
426 ;auth_options_requests = yes ; Enabling this option will authenticate OPTIONS requests just like
427 ; INVITE requests are. By default this option is disabled.
429 ;accept_outofcall_message = no ; Disable this option to reject all MESSAGE requests outside of a
430 ; call. By default, this option is enabled. When enabled, MESSAGE
431 ; requests are passed in to the dialplan.
433 ;outofcall_message_context = messages ; Context all out of dialog msgs are sent to. When this
434 ; option is not set, the context used during peer matching
435 ; is used. This option can be defined at both the peer and
438 ;auth_message_requests = yes ; Enabling this option will authenticate MESSAGE requests.
439 ; By default this option is enabled. However, it can be disabled
440 ; should an application desire to not load the Asterisk server with
441 ; doing authentication and implement end to end security in the
444 ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
445 ; order instead of RFC3551 packing order (this is required
446 ; for Sipura and Grandstream ATAs, among others). This is
447 ; contrary to the RFC3551 specification, the peer _should_
448 ; be negotiating AAL2-G726-32 instead :-(
449 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
450 ;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
451 ;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
452 ;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls
453 ;outboundproxy=192.0.2.1 ; IPv4 address literal (default port is 5060)
454 ;outboundproxy=2001:db8::1 ; IPv6 address literal (default port is 5060)
455 ;outboundproxy=192.168.0.2.1:5062 ; IPv4 address literal with explicit port
456 ;outboundproxy=[2001:db8::1]:5062 ; IPv6 address literal with explicit port
457 ; ; (could also be tcp,udp) - defining transports on the proxy line only
458 ; ; applies for the global proxy, otherwise use the transport= option
460 ;supportpath=yes ; This activates parsing and handling of Path header as defined in RFC 3327. This enables
461 ; Asterisk to route outgoing out-of-dialog requests via a set of proxies by using a pre-loaded
462 ; route-set defined by the Path headers in the REGISTER request.
463 ; NOTE: There are multiple things to consider with this setting:
464 ; * As this influences routing of SIP requests make sure to not trust Path headers provided
465 ; by the user's SIP client (the proxy in front of Asterisk should remove existing user
466 ; provided Path headers).
467 ; * When a peer has both a path and outboundproxy set, the path will be added to Route: header
468 ; but routing to next hop is done using the outboundproxy.
469 ; * If set globally, not only will all peers use the Path header, but outbound REGISTER
470 ; requests from Asterisk will add path to the Supported header.
472 ;rtsavepath=yes ; If using dynamic realtime, store the path headers
474 ;matchexternaddrlocally = yes ; Only substitute the externaddr or externhost setting if it matches
475 ; your localnet setting. Unless you have some sort of strange network
476 ; setup you will not need to enable this.
478 ;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering
479 ; as any IP address used for staticly defined
480 ; hosts. This helps avoid the configuration
481 ; error of allowing your users to register at
482 ; the same address as a SIP provider.
484 ;contactdeny=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to
485 ;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may
486 ; register their phones.
487 ;contactacl=named_acl_example ; Use named ACLs defined in acl.conf
489 ;rtp_engine=asterisk ; RTP engine to use when communicating with the device
492 ; If regcontext is specified, Asterisk will dynamically create and destroy a
493 ; NoOp priority 1 extension for a given peer who registers or unregisters with
494 ; us and have a "regexten=" configuration item.
495 ; Multiple contexts may be specified by separating them with '&'. The
496 ; actual extension is the 'regexten' parameter of the registering peer or its
497 ; name if 'regexten' is not provided. If more than one context is provided,
498 ; the context must be specified within regexten by appending the desired
499 ; context after '@'. More than one regexten may be supplied if they are
500 ; separated by '&'. Patterns may be used in regexten.
502 ;regcontext=sipregistrations
503 ;regextenonqualify=yes ; Default "no"
504 ; If you have qualify on and the peer becomes unreachable
505 ; this setting will enforce inactivation of the regexten
506 ; extension for the peer
507 ;legacy_useroption_parsing=yes ; Default "no" ; If you have this option enabled and there are semicolons
508 ; in the user field of a sip URI, the field be truncated
509 ; at the first semicolon seen. This effectively makes
510 ; semicolon a non-usable character for peer names, extensions,
511 ; and maybe other, less tested things. This can be useful
512 ; for improving compatability with devices that like to use
513 ; user options for whatever reason. The behavior is similar to
514 ; how SIP URI's were typically handled in 1.6.2, hence the name.
516 ;send_diversion=no ; Default "yes" ; Asterisk normally sends Diversion headers with certain SIP
517 ; invites to relay data about forwarded calls. If this option
518 ; is disabled, Asterisk won't send Diversion headers unless
519 ; they are added manually.
521 ; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
522 ; in square brackets. For example, the caller id value 555.5555 becomes 5555555
523 ; when this option is enabled. Disabling this option results in no modification
524 ; of the caller id value, which is necessary when the caller id represents something
525 ; that must be preserved. This option can only be used in the [general] section.
526 ; By default this option is on.
528 ;shrinkcallerid=yes ; on by default
531 ;use_q850_reason = no ; Default "no"
532 ; Set to yes add Reason header and use Reason header if it is available.
534 ; When the Transfer() application sends a REFER SIP message, extra headers specified in
535 ; the dialplan by way of SIPAddHeader are sent out with that message. 1.8 and earlier did not
536 ; add the extra headers. To revert to 1.8- behavior, call SIPRemoveHeader with no arguments
537 ; before calling Transfer() to remove all additional headers from the channel. The setting
538 ; below is for transitional compatibility only.
540 ;refer_addheaders=yes ; on by default
542 ;autocreatepeer=no ; Allow any UAC not explicitly defined to register
543 ; WITHOUT AUTHENTICATION. Enabling this options poses a high
544 ; potential security risk and should be avoided unless the
545 ; server is behind a trusted firewall.
546 ; If set to "yes", then peers created in this fashion
547 ; are purged during SIP reloads.
548 ; When set to "persist", the peers created in this fashion
549 ; are not purged during SIP reloads.
552 ;------------------------ TLS settings ------------------------------------------------------------
553 ;tlscertfile=</path/to/certificate.pem> ; Certificate chain (*.pem format only) to use for TLS connections
554 ; The certificates must be sorted starting with the subject's certificate
555 ; and followed by intermediate CA certificates if applicable.
556 ; Default is to look for "asterisk.pem" in current directory
558 ;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem format only) for TLS connections.
559 ; If no tlsprivatekey is specified, tlscertfile is searched for
560 ; for both public and private key.
562 ;tlscafile=</path/to/certificate>
563 ; If the server your connecting to uses a self signed certificate
564 ; you should have their certificate installed here so the code can
565 ; verify the authenticity of their certificate.
567 ;tlscapath=</path/to/ca/dir>
568 ; A directory full of CA certificates. The files must be named with
569 ; the CA subject name hash value.
570 ; (see man SSL_CTX_load_verify_locations for more info)
572 ;tlsdontverifyserver=[yes|no]
573 ; If set to yes, don't verify the servers certificate when acting as
574 ; a client. If you don't have the server's CA certificate you can
575 ; set this and it will connect without requiring tlscafile to be set.
578 ;tlscipher=<SSL cipher string>
579 ; A string specifying which SSL ciphers to use or not use
580 ; A list of valid SSL cipher strings can be found at:
581 ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
583 ;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2.
584 ; Specify protocol for outbound client connections.
585 ; If left unspecified, the default is sslv2.
587 ;--------------------------- SIP timers ----------------------------------------------------
588 ; These timers are used primarily in INVITE transactions.
589 ; The default for Timer T1 is 500 ms or the measured run-trip time between
590 ; Asterisk and the device if you have qualify=yes for the device.
592 ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
594 ;timert1=500 ; Default T1 timer
595 ; Defaults to 500 ms or the measured round-trip
596 ; time to a peer (qualify=yes).
597 ;timerb=32000 ; Call setup timer. If a provisional response is not received
598 ; in this amount of time, the call will autocongest
599 ; Defaults to 64*timert1
601 ;--------------------------- RTP timers ----------------------------------------------------
602 ; These timers are currently used for both audio and video streams. The RTP timeouts
603 ; are only applied to the audio channel.
604 ; The settings are settable in the global section as well as per device
606 ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
607 ; on the audio channel
608 ; when we're not on hold. This is to be able to hangup
609 ; a call in the case of a phone disappearing from the net,
610 ; like a powerloss or grandma tripping over a cable.
611 ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
612 ; on the audio channel
613 ; when we're on hold (must be > rtptimeout)
614 ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
615 ; (default is off - zero)
617 ;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
618 ; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
619 ; This mechanism can detect and reclaim SIP channels that do not terminate through normal
620 ; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
621 ; The operation of Session-Timers is driven by the following configuration parameters:
623 ; * session-timers - Session-Timers feature operates in the following three modes:
624 ; originate : Request and run session-timers always
625 ; accept : Run session-timers only when requested by other UA
626 ; refuse : Do not run session timers in any case
627 ; The default mode of operation is 'accept'.
628 ; * session-expires - Maximum session refresh interval in seconds. Defaults to 1800 secs.
629 ; * session-minse - Minimum session refresh interval in seconds. Defualts to 90 secs.
630 ; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
631 ; uac - Default to the caller initially refreshing when possible
632 ; uas - Default to the callee initially refreshing when possible
634 ; Note that, due to recommendations in RFC 4028, Asterisk will always honor the other
635 ; endpoint's preference for who will handle refreshes. Asterisk will never override the
636 ; preferences of the other endpoint. Doing so could result in Asterisk and the endpoint
637 ; fighting over who sends the refreshes. This holds true for the initiation of session
638 ; timers and subsequent re-INVITE requests whether Asterisk is the caller or callee, or
639 ; whether Asterisk is currently the refresher or not.
641 ;session-timers=originate
644 ;session-refresher=uac
646 ;--------------------------- SIP DEBUGGING ---------------------------------------------------
647 ;sipdebug = yes ; Turn on SIP debugging by default, from
648 ; the moment the channel loads this configuration
649 ;recordhistory=yes ; Record SIP history by default
650 ; (see sip history / sip no history)
651 ;dumphistory=yes ; Dump SIP history at end of SIP dialogue
652 ; SIP history is output to the DEBUG logging channel
655 ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
656 ; You can subscribe to the status of extensions with a "hint" priority
657 ; (See extensions.conf.sample for examples)
658 ; chan_sip support two major formats for notifications: dialog-info and SIMPLE
660 ; You will get more detailed reports (busy etc) if you have a call counter enabled
663 ; If you set the busylevel, we will indicate busy when we have a number of calls that
664 ; matches the busylevel treshold.
666 ; For queues, you will need this level of detail in status reporting, regardless
667 ; if you use SIP subscriptions. Queues and manager use the same internal interface
668 ; for reading status information.
670 ; Note: Subscriptions does not work if you have a realtime dialplan and use the
673 ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
674 ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
675 ; Useful to limit subscriptions to local extensions
676 ; Settable per peer/user also
677 ;notifyringing = no ; Control whether subscriptions already INUSE get sent
678 ; RINGING when another call is sent (default: yes)
679 ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
680 ; Turning on notifyringing and notifyhold will add a lot
681 ; more database transactions if you are using realtime.
682 ;notifycid = yes ; Control whether caller ID information is sent along with
683 ; dialog-info+xml notifications (supported by snom phones).
684 ; Note that this feature will only work properly when the
685 ; incoming call is using the same extension and context that
686 ; is being used as the hint for the called extension. This means
687 ; that it won't work when using subscribecontext for your sip
688 ; user or peer (if subscribecontext is different than context).
689 ; This is also limited to a single caller, meaning that if an
690 ; extension is ringing because multiple calls are incoming,
691 ; only one will be used as the source of caller ID. Specify
692 ; 'ignore-context' to ignore the called context when looking
693 ; for the caller's channel. The default value is 'no.' Setting
694 ; notifycid to 'ignore-context' also causes call-pickups attempted
695 ; via SNOM's NOTIFY mechanism to set the context for the call pickup
697 ;callcounter = yes ; Enable call counters on devices. This can be set per
700 ;----------------------------------------- T.38 FAX SUPPORT ----------------------------------
702 ; This setting is available in the [general] section as well as in device configurations.
703 ; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off.
705 ; t38pt_udptl = yes ; Enables T.38 with FEC error correction.
706 ; t38pt_udptl = yes,fec ; Enables T.38 with FEC error correction.
707 ; t38pt_udptl = yes,redundancy ; Enables T.38 with redundancy error correction.
708 ; t38pt_udptl = yes,none ; Enables T.38 with no error correction.
710 ; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that
711 ; is based on an incorrect interpretation of the T.38 recommendation, and results in failures
712 ; because Asterisk does not believe it can send T.38 packets of a reasonable size to that
713 ; endpoint (Cisco media gateways are one example of this situation). In these cases, during a
714 ; T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL
715 ; stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you
716 ; can set an override (globally, or on a per-device basis) to make Asterisk ignore the
717 ; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead.
718 ; This can be done by appending 'maxdatagram=<value>' to the t38pt_udptl configuration option,
721 ; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error correction and overrides
722 ; ; the other endpoint's provided value to assume we can
723 ; ; send 400 byte T.38 FAX packets to it.
725 ; FAX detection will cause the SIP channel to jump to the 'fax' extension (if it exists)
726 ; based one or more events being detected. The events that can be detected are an incoming
727 ; CNG tone or an incoming T.38 re-INVITE request.
729 ; faxdetect = yes ; Default 'no', 'yes' enables both CNG and T.38 detection
730 ; faxdetect = cng ; Enables only CNG detection
731 ; faxdetect = t38 ; Enables only T.38 detection
733 ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
734 ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
735 ; Format for the register statement is:
736 ; register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
743 ; - the name of a peer defined below or in realtime
744 ; The domain is where you register your username, so your SIP uri you are registering to
747 ; If no extension is given, the 's' extension is used. The extension needs to
748 ; be defined in extensions.conf to be able to accept calls from this SIP proxy
751 ; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
752 ; this is equivalent to having the following line in the general section:
754 ; register => username:secret@host/callbackextension
756 ; and more readable because you don't have to write the parameters in two places
757 ; (note that the "port" is ignored - this is a bug that should be fixed).
759 ; Note that a register= line doesn't mean that we will match the incoming call in any
760 ; other way than described above. If you want to control where the call enters your
761 ; dialplan, which context, you want to define a peer with the hostname of the provider's
762 ; server. If the provider has multiple servers to place calls to your system, you need
763 ; a peer for each server.
765 ; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may
766 ; contain a port number. Since the logical separator between a host and port number is a
767 ; ':' character, and this character is already used to separate between the optional "secret"
768 ; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish
769 ; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
770 ; they are blank. See the third example below for an illustration.
775 ;register => 1234:password@mysipprovider.com
777 ; This will pass incoming calls to the 's' extension
780 ;register => 2345:password@sip_proxy/1234
782 ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
783 ; connect to local extension 1234 in extensions.conf, default context,
784 ; unless you configure a [sip_proxy] section below, and configure a
786 ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
787 ; Tip 2: Use separate inbound and outbound sections for SIP providers
788 ; (instead of type=friend) if you have calls in both directions
790 ;register => 3456@mydomain:5082::@mysipprovider.com
792 ; Note that in this example, the optional authuser and secret portions have
793 ; been left blank because we have specified a port in the user section
795 ;register => tls://username:xxxxxx@sip-tls-proxy.example.org
797 ; The 'transport' part defaults to 'udp' but may also be 'tcp', 'tls', 'ws', or 'wss'.
798 ; Using 'udp://' explicitly is also useful in case the username part
799 ; contains a '/' ('user/name').
801 ;registertimeout=20 ; retry registration calls every 20 seconds (default)
802 ;registerattempts=10 ; Number of registration attempts before we give up
803 ; 0 = continue forever, hammering the other server
804 ; until it accepts the registration
805 ; Default is 0 tries, continue forever
806 ;register_retry_403=yes ; Treat 403 responses to registrations as if they were
807 ; 401 responses and continue retrying according to normal
810 ;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
811 ; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
812 ; by other phones. At this time, you can only subscribe using UDP as the transport.
813 ; Format for the mwi register statement is:
814 ; mwi => user[:secret[:authuser]]@host[:port]/mailbox
817 ;mwi => 1234:password@mysipprovider.com/1234
818 ;mwi => 1234:password@myportprovider.com:6969/1234
819 ;mwi => 1234:password:authuser@myauthprovider.com/1234
820 ;mwi => 1234:password:authuser@myauthportprovider.com:6969/1234
822 ; MWI received will be stored in the 1234 mailbox of the SIP_Remote context.
823 ; It can be used by other phones by following the below:
824 ; mailbox=1234@SIP_Remote
825 ;----------------------------------------- NAT SUPPORT ------------------------
827 ; WARNING: SIP operation behind a NAT is tricky and you really need
828 ; to read and understand well the following section.
830 ; When Asterisk is behind a NAT device, the "local" address (and port) that
831 ; a socket is bound to has different values when seen from the inside or
832 ; from the outside of the NATted network. Unfortunately this address must
833 ; be communicated to the outside (e.g. in SIP and SDP messages), and in
834 ; order to determine the correct value Asterisk needs to know:
836 ; + whether it is talking to someone "inside" or "outside" of the NATted network.
837 ; This is configured by assigning the "localnet" parameter with a list
838 ; of network addresses that are considered "inside" of the NATted network.
839 ; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
840 ; Multiple entries are allowed, e.g. a reasonable set is the following:
842 ; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
843 ; localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
844 ; localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
845 ; localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
847 ; + the "externally visible" address and port number to be used when talking
848 ; to a host outside the NAT. This information is derived by one of the
849 ; following (mutually exclusive) config file parameters:
851 ; a. "externaddr = hostname[:port]" specifies a static address[:port] to
852 ; be used in SIP and SDP messages.
853 ; The hostname is looked up only once, when [re]loading sip.conf .
854 ; If a port number is not present, use the port specified in the "udpbindaddr"
855 ; (which is not guaranteed to work correctly, because a NAT box might remap the
856 ; port number as well as the address).
857 ; This approach can be useful if you have a NAT device where you can
858 ; configure the mapping statically. Examples:
860 ; externaddr = 12.34.56.78 ; use this address.
861 ; externaddr = 12.34.56.78:9900 ; use this address and port.
862 ; externaddr = mynat.my.org:12600 ; Public address of my nat box.
863 ; externtcpport = 9900 ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT.
864 ; ; externtcpport will default to the externaddr or externhost port if either one is set.
865 ; externtlsport = 12600 ; The externally mapped tls port, when Asterisk is behind a static NAT or PAT.
866 ; ; externtlsport port will default to the RFC designated port of 5061.
868 ; b. "externhost = hostname[:port]" is similar to "externaddr" except
869 ; that the hostname is looked up every "externrefresh" seconds
870 ; (default 10s). This can be useful when your NAT device lets you choose
871 ; the port mapping, but the IP address is dynamic.
872 ; Beware, you might suffer from service disruption when the name server
873 ; resolution fails. Examples:
875 ; externhost=foo.dyndns.net ; refreshed periodically
876 ; externrefresh=180 ; change the refresh interval
878 ; Note that at the moment all these mechanism work only for the SIP socket.
879 ; The IP address discovered with externaddr/externhost is reused for
880 ; media sessions as well, but the port numbers are not remapped so you
881 ; may still experience problems.
883 ; NOTE 1: in some cases, NAT boxes will use different port numbers in
884 ; the internal<->external mapping. In these cases, the "externaddr" and
885 ; "externhost" might not help you configure addresses properly.
887 ; NOTE 2: when using "externaddr" or "externhost", the address part is
888 ; also used as the external address for media sessions. Thus, the port
889 ; information in the SDP may be wrong!
891 ; In addition to the above, Asterisk has an additional "nat" parameter to
892 ; address NAT-related issues in incoming SIP or media sessions.
893 ; In particular, depending on the 'nat= ' settings described below, Asterisk
894 ; may override the address/port information specified in the SIP/SDP messages,
895 ; and use the information (sender address) supplied by the network stack instead.
896 ; However, this is only useful if the external traffic can reach us.
897 ; The following settings are allowed (both globally and in individual sections):
899 ; nat = no ; Do no special NAT handling other than RFC3581
900 ; nat = force_rport ; Pretend there was an rport parameter even if there wasn't
901 ; nat = comedia ; Send media to the port Asterisk received it from regardless
902 ; ; of where the SDP says to send it.
903 ; nat = auto_force_rport ; Set the force_rport option if Asterisk detects NAT (default)
904 ; nat = auto_comedia ; Set the comedia option if Asterisk detects NAT
906 ; The nat settings can be combined. For example, to set both force_rport and comedia
907 ; one would set nat=force_rport,comedia. If any of the comma-separated options is 'no',
908 ; Asterisk will ignore any other settings and set nat=no. If one of the "auto" settings
909 ; is used in conjunction with its non-auto counterpart (nat=comedia,auto_comedia), then
910 ; the non-auto option will be ignored.
912 ; The RFC 3581-defined 'rport' parameter allows a client to request that Asterisk send
913 ; SIP responses to it via the source IP and port from which the request originated
914 ; instead of the address/port listed in the top-most Via header. This is useful if a
915 ; client knows that it is behind a NAT and therefore cannot guess from what address/port
916 ; its request will be sent. Asterisk will always honor the 'rport' parameter if it is
917 ; sent. The force_rport setting causes Asterisk to always send responses back to the
918 ; address/port from which it received requests; even if the other side doesn't support
919 ; adding the 'rport' parameter.
921 ; 'comedia RTP handling' refers to the technique of sending RTP to the port that the
922 ; the other endpoint's RTP arrived from, and means 'connection-oriented media'. This is
923 ; only partially related to RFC 4145 which was referred to as COMEDIA while it was in
924 ; draft form. This method is used to accomodate endpoints that may be located behind
925 ; NAT devices, and as such the address/port they tell Asterisk to send RTP packets to
926 ; for their media streams is not the actual address/port that will be used on the nearer
929 ; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
930 ; the nat setting in a peer definition, then the peer username will be discoverable
931 ; by outside parties as Asterisk will respond to different ports for defined and
932 ; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
933 ; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the
934 ; other, then valid peers with settings differing from those in the general section will
937 ; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by
938 ; RFC 4961; Asterisk will always send RTP packets from the same port number it expects
939 ; to receive them on.
941 ; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using
942 ; the media_address configuration option. This is only applicable to the general section and
943 ; can not be set per-user or per-peer.
945 ; media_address = 172.16.42.1
947 ; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the
948 ; perceived external network address has changed. When the stun_monitor is installed and
949 ; configured, chan_sip will renew all outbound registrations when the monitor detects any sort
950 ; of network change has occurred. By default this option is enabled, but only takes effect once
951 ; res_stun_monitor is configured. If res_stun_monitor is enabled and you wish to not
952 ; generate all outbound registrations on a network change, use the option below to disable
955 ; subscribe_network_change_event = yes ; on by default
957 ; ICE/STUN/TURN usage can be enabled globally or on a per-peer basis using the icesupport
958 ; configuration option. When set to yes ICE support is enabled. When set to no it is disabled.
959 ; It is disabled by default.
963 ;----------------------------------- MEDIA HANDLING --------------------------------
964 ; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
965 ; no reason for Asterisk to stay in the media path, the media will be redirected.
966 ; This does not really work well in the case where Asterisk is outside and the
967 ; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
969 ;directmedia=yes ; Asterisk by default tries to redirect the
970 ; RTP media stream to go directly from
971 ; the caller to the callee. Some devices do not
972 ; support this (especially if one of them is behind a NAT).
973 ; The default setting is YES. If you have all clients
974 ; behind a NAT, or for some other reason want Asterisk to
975 ; stay in the audio path, you may want to turn this off.
977 ; This setting also affect direct RTP
978 ; at call setup (a new feature in 1.4 - setting up the
979 ; call directly between the endpoints instead of sending
982 ; Additionally this option does not disable all reINVITE operations.
983 ; It only controls Asterisk generating reINVITEs for the specific
984 ; purpose of setting up a direct media path. If a reINVITE is
985 ; needed to switch a media stream to inactive (when placed on
986 ; hold) or to T.38, it will still be done, regardless of this
987 ; setting. Note that direct T.38 is not supported.
989 ;directmedia=nonat ; An additional option is to allow media path redirection
990 ; (reinvite) but only when the peer where the media is being
991 ; sent is known to not be behind a NAT (as the RTP core can
992 ; determine it based on the apparent IP address the media
995 ;directmedia=update ; Yet a third option... use UPDATE for media path redirection,
996 ; instead of INVITE. This can be combined with 'nonat', as
997 ; 'directmedia=update,nonat'. It implies 'yes'.
999 ;directmedia=outgoing ; When sending directmedia reinvites, do not send an immediate
1000 ; reinvite on an incoming call leg. This option is useful when
1001 ; peered with another SIP user agent that is known to send
1002 ; immediate direct media reinvites upon call establishment. Setting
1003 ; the option in this situation helps to prevent potential glares.
1004 ; Setting this option implies 'yes'.
1006 ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
1007 ; the call directly with media peer-2-peer without re-invites.
1008 ; Will not work for video and cases where the callee sends
1009 ; RTP payloads and fmtp headers in the 200 OK that does not match the
1010 ; callers INVITE. This will also fail if directmedia is enabled when
1011 ; the device is actually behind NAT.
1013 ;directmediadeny=0.0.0.0/0 ; Use directmediapermit and directmediadeny to restrict
1014 ;directmediapermit=172.16.0.0/16; which peers should be able to pass directmedia to each other
1015 ; (There is no default setting, this is just an example)
1016 ; Use this if some of your phones are on IP addresses that
1017 ; can not reach each other directly. This way you can force
1018 ; RTP to always flow through asterisk in such cases.
1019 ;directmediaacl=acl_example ; Use named ACLs defined in acl.conf
1021 ;ignoresdpversion=yes ; By default, Asterisk will honor the session version
1022 ; number in SDP packets and will only modify the SDP
1023 ; session if the version number changes. This option will
1024 ; force asterisk to ignore the SDP session version number
1025 ; and treat all SDP data as new data. This is required
1026 ; for devices that send us non standard SDP packets
1027 ; (observed with Microsoft OCS). By default this option is
1030 ;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
1031 ; Like the useragent parameter, the default user agent string
1032 ; also contains the Asterisk version.
1033 ;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
1034 ; This field MUST NOT contain spaces
1035 ;encryption=no ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
1036 ; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
1037 ; the peer does not support SRTP. Defaults to no.
1038 ;encryption_taglen=80 ; Set the auth tag length offered in the INVITE either 32/80 default 80
1040 ;avpf=yes ; Enable inter-operability with media streams using the AVPF RTP profile.
1041 ; This will cause all offers and answers to use AVPF (or SAVPF). This
1042 ; option may be specified at the global or peer scope.
1043 ;----------------------------------------- REALTIME SUPPORT ------------------------
1044 ; For additional information on ARA, the Asterisk Realtime Architecture,
1045 ; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
1047 ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
1048 ; just like friends added from the config file only on a
1049 ; as-needed basis? (yes|no)
1051 ;rtsavesysname=yes ; Save systemname in realtime database at registration
1054 ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
1055 ; If set to yes, when a SIP UA registers successfully, the ip address,
1056 ; the origination port, the registration period, and the username of
1057 ; the UA will be set to database via realtime.
1058 ; If not present, defaults to 'yes'. Note: realtime peers will
1059 ; probably not function across reloads in the way that you expect, if
1060 ; you turn this option off.
1061 ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
1062 ; as if it had just registered? (yes|no|<seconds>)
1063 ; If set to yes, when the registration expires, the friend will
1064 ; vanish from the configuration until requested again. If set
1065 ; to an integer, friends expire within this number of seconds
1066 ; instead of the registration interval.
1068 ;ignoreregexpire=yes ; Enabling this setting has two functions:
1070 ; For non-realtime peers, when their registration expires, the
1071 ; information will _not_ be removed from memory or the Asterisk database
1072 ; if you attempt to place a call to the peer, the existing information
1073 ; will be used in spite of it having expired
1075 ; For realtime peers, when the peer is retrieved from realtime storage,
1076 ; the registration information will be used regardless of whether
1077 ; it has expired or not; if it expires while the realtime peer
1078 ; is still in memory (due to caching or other reasons), the
1079 ; information will not be removed from realtime storage
1081 ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
1082 ; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
1083 ; domains, each of which can direct the call to a specific context if desired.
1084 ; By default, all domains are accepted and sent to the default context or the
1085 ; context associated with the user/peer placing the call.
1086 ; REGISTER to non-local domains will be automatically denied if a domain
1087 ; list is configured.
1089 ; Domains can be specified using:
1090 ; domain=<domain>[,<context>]
1092 ; domain=myasterisk.dom
1093 ; domain=customer.com,customer-context
1095 ; In addition, all the 'default' domains associated with a server should be
1096 ; added if incoming request filtering is desired.
1099 ; To disallow requests for domains not serviced by this server:
1100 ; allowexternaldomains=no
1102 ;domain=mydomain.tld,mydomain-incoming
1103 ; Add domain and configure incoming context
1104 ; for external calls to this domain
1105 ;domain=1.2.3.4 ; Add IP address as local domain
1106 ; You can have several "domain" settings
1107 ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
1109 ;autodomain=yes ; Turn this on to have Asterisk add local host
1110 ; name and local IP to domain list.
1112 ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
1113 ; non-peers, use your primary domain "identity"
1114 ; for From: headers instead of just your IP
1115 ; address. This is to be polite and
1116 ; it may be a mandatory requirement for some
1117 ; destinations which do not have a prior
1118 ; account relationship with your server.
1120 ;------------------------------ Advice of Charge CONFIGURATION --------------------------
1121 ; snom_aoc_enabled = yes; ; This options turns on and off support for sending AOC-D and
1122 ; AOC-E to snom endpoints. This option can be used both in the
1123 ; peer and global scope. The default for this option is off.
1126 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
1127 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
1128 ; SIP channel. Defaults to "no". An enabled jitterbuffer will
1129 ; be used only if the sending side can create and the receiving
1130 ; side can not accept jitter. The SIP channel can accept jitter,
1131 ; thus a jitterbuffer on the receive SIP side will be used only
1132 ; if it is forced and enabled.
1134 ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
1135 ; channel. Defaults to "no".
1137 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
1139 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
1140 ; resynchronized. Useful to improve the quality of the voice, with
1141 ; big jumps in/broken timestamps, usually sent from exotic devices
1142 ; and programs. Defaults to 1000.
1144 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
1145 ; channel. Two implementations are currently available - "fixed"
1146 ; (with size always equals to jbmaxsize) and "adaptive" (with
1147 ; variable size, actually the new jb of IAX2). Defaults to fixed.
1149 ; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
1150 ; The option represents the number of milliseconds by which the new jitter buffer
1151 ; will pad its size. the default is 40, so without modification, the new
1152 ; jitter buffer will set its size to the jitter value plus 40 milliseconds.
1153 ; increasing this value may help if your network normally has low jitter,
1154 ; but occasionally has spikes.
1156 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
1158 ;-----------------------------------------------------------------------------------
1161 ; Global credentials for outbound calls, i.e. when a proxy challenges your
1162 ; Asterisk server for authentication. These credentials override
1163 ; any credentials in peer/register definition if realm is matched.
1165 ; This way, Asterisk can authenticate for outbound calls to other
1166 ; realms. We match realm on the proxy challenge and pick an set of
1167 ; credentials from this list
1169 ; auth = <user>:<secret>@<realm>
1170 ; auth = <user>#<md5secret>@<realm>
1172 ;auth=mark:topsecret@digium.com
1174 ; You may also add auth= statements to [peer] definitions
1175 ; Peer auth= override all other authentication settings if we match on realm
1177 ;------------------------------------------------------------------------------
1178 ; DEVICE CONFIGURATION
1180 ; SIP entities have a 'type' which determines their roles within Asterisk.
1181 ; * For entities with 'type=peer':
1182 ; Peers handle both inbound and outbound calls and are matched by ip/port, so for
1183 ; The case of incoming calls from the peer, the IP address must match in order for
1184 ; The invitation to work. This means calls made from either direction won't work if
1185 ; The peer is unregistered while host=dynamic or if the host is otherise not set to
1186 ; the correct IP of the sender.
1187 ; * For entities with 'type=user':
1188 ; Asterisk users handle inbound calls only (meaning they call Asterisk, Asterisk can't
1189 ; call them) and are matched by their authorization information (authname and secret).
1190 ; Asterisk doesn't rely on their IP and will accept calls regardless of the host setting
1191 ; as long as the incoming SIP invite authorizes successfully.
1192 ; * For entities with 'type=friend':
1193 ; Asterisk will create the entity as both a friend and a peer. Asterisk will accept
1194 ; calls from friends like it would for users, requiring only that the authorization
1195 ; matches rather than the IP address. Since it is also a peer, a friend entity can
1196 ; be called as long as its IP is known to Asterisk. In the case of host=dynamic,
1197 ; this means it is necessary for the entity to register before Asterisk can call it.
1199 ; Use remotesecret for outbound authentication, and secret for authenticating
1200 ; inbound requests. For historical reasons, if no remotesecret is supplied for an
1201 ; outbound registration or call, the secret will be used.
1203 ; For device names, we recommend using only a-z, numerics (0-9) and underscore
1205 ; For local phones, type=friend works most of the time
1207 ; If you have one-way audio, you probably have NAT problems.
1208 ; If Asterisk is on a public IP, and the phone is inside of a NAT device
1209 ; you will need to configure nat option for those phones.
1210 ; Also, turn on qualify=yes to keep the nat session open
1212 ; Configuration options available
1213 ; --------------------
1252 ; Note: app_voicemail mailboxes must be in the form of mailbox@context.
1258 ; t38pt_usertpsource
1277 ; t38pt_usertpsource
1278 ; contactpermit ; Limit what a host may register as (a neat trick
1279 ; contactdeny ; is to register at the same IP as a SIP provider,
1280 ; contactacl ; then call oneself, and get redirected to that
1285 ; unsolicited_mailbox
1289 ; description ; Used to provide a description of the peer in console output
1299 ; ignore_requested_pref ; Ignore the requested codec and determine the preferred codec
1300 ; ; from the peer's configuration.
1303 ;------------------------------------------------------------------------------
1304 ; DTLS-SRTP CONFIGURATION
1306 ; DTLS-SRTP support is available if the underlying RTP engine in use supports it.
1308 ; dtlsenable = yes ; Enable or disable DTLS-SRTP support
1309 ; dtlsverify = yes ; Verify that the provided peer certificate is valid
1310 ; dtlsrekey = 60 ; Interval at which to renegotiate the TLS session and rekey the SRTP session
1311 ; ; If this is not set or the value provided is 0 rekeying will be disabled
1312 ; dtlscertfile = file ; Path to certificate file to present
1313 ; dtlsprivatekey = file ; Path to private key for certificate file
1314 ; dtlscipher = <SSL cipher string> ; Cipher to use for TLS negotiation
1315 ; ; A list of valid SSL cipher strings can be found at:
1316 ; ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
1317 ; dtlscafile = file ; Path to certificate authority certificate
1318 ; dtlscapath = path ; Path to a directory containing certificate authority certificates
1319 ; dtlssetup = actpass ; Whether we are willing to accept connections, connect to the other party, or both.
1320 ; ; Valid options are active (we want to connect to the other party), passive (we want to
1321 ; ; accept connections only), and actpass (we will do both). This value will be used in
1322 ; ; the outgoing SDP when offering and for incoming SDP offers when the remote party sends
1326 ; For incoming calls only. Example: FWD (Free World Dialup)
1327 ; We match on IP address of the proxy for incoming calls
1328 ; since we can not match on username (caller id)
1331 ;host=fwd.pulver.com
1334 ;type=peer ; we only want to call out, not be called
1335 ;remotesecret=guessit ; Our password to their service
1336 ;defaultuser=yourusername ; Authentication user for outbound proxies
1337 ;fromuser=yourusername ; Many SIP providers require this!
1338 ;fromdomain=provider.sip.domain
1339 ;host=box.provider.com
1340 ;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will
1341 ; ; accept both tcp and udp. The default transport type is only used for
1342 ; ; outbound messages until a Registration takes place. During the
1343 ; ; peer Registration the transport type may change to another supported
1344 ; ; type if the peer requests so.
1346 ;usereqphone=yes ; This provider requires ";user=phone" on URI
1347 ;callcounter=yes ; Enable call counter
1348 ;busylevel=2 ; Signal busy at 2 or more calls
1349 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
1350 ;port=80 ; The port number we want to connect to on the remote side
1351 ; Also used as "defaultport" in combination with "defaultip" settings
1353 ;--- sample definition for a provider
1356 ;host=sip.provider1.com
1357 ;fromuser=4015552299 ; how your provider knows you
1358 ;remotesecret=youwillneverguessit ; The password we use to authenticate to them
1359 ;secret=gissadetdu ; The password they use to contact us
1360 ;callbackextension=123 ; Register with this server and require calls coming back to this extension
1361 ;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
1362 ; ; accept both tcp and udp. Default is udp. The first transport
1363 ; ; listed will always be used for outgoing connections.
1364 ;unsolicited_mailbox=4015552299 ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old
1365 ; ; message count will be stored in the configured virtual mailbox. It can be used
1366 ; ; by any device supporting MWI by specifying <configured value>@SIP_Remote as the
1370 ; Because you might have a large number of similar sections, it is generally
1371 ; convenient to use templates for the common parameters, and add them
1372 ; the the various sections. Examples are below, and we can even leave
1373 ; the templates uncommented as they will not harm:
1375 [basic-options](!) ; a template
1380 [natted-phone](!,basic-options) ; another template inheriting basic-options
1384 [public-phone](!,basic-options) ; another template inheriting basic-options
1387 [my-codecs](!) ; a template for my preferred codecs
1395 ;allow=!all,ilbc,g729,gsm,g723,ulaw
1397 [ulaw-phone](!) ; and another one for ulaw-only
1400 ; Again, more simply:
1403 ; and finally instantiate a few phones
1405 ; [2133](natted-phone,my-codecs)
1407 ; [2134](natted-phone,ulaw-phone)
1408 ; secret = not_very_secret
1409 ; [2136](public-phone,ulaw-phone)
1410 ; secret = not_very_secret_either
1414 ; Standard configurations not using templates look like this:
1418 ;context=from-sip ; Where to start in the dialplan when this phone calls
1419 ;recordonfeature=dynamicfeature1 ; Feature to use when INFO with Record: on is received.
1420 ;recordofffeature=dynamicfeature2 ; Feature to use when INFO with Record: off is received.
1421 ;callerid=John Doe <1234> ; Full caller ID, to override the phones config
1422 ; on incoming calls to Asterisk
1423 ;description=Courtesy Phone ; Description of the peer. Shown when doing 'sip show peers'.
1424 ;host=192.168.0.23 ; we have a static but private IP address
1425 ; No registration allowed
1426 ;directmedia=yes ; allow RTP voice traffic to bypass Asterisk
1427 ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
1428 ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
1429 ; from the phone to asterisk (deprecated)
1430 ; 1 for the explicit peer, 1 for the explicit user,
1431 ; remember that a friend equals 1 peer and 1 user in
1433 ; There is no combined call counter for a "friend"
1434 ; so there's currently no way in sip.conf to limit
1435 ; to one inbound or outbound call per phone. Use
1436 ; the group counters in the dial plan for that.
1438 ;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
1439 ;disallow=all ; need to disallow=all before we can use allow=
1440 ;allow=ulaw ; Note: In user sections the order of codecs
1441 ; listed with allow= does NOT matter!
1443 ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
1444 ;allow=g729 ; Pass-thru only unless g729 license obtained
1445 ;callingpres=allowed_passed_screen ; Set caller ID presentation
1446 ; See function CALLERPRES documentation for possible
1450 ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
1451 ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
1453 ;regexten=1234 ; When they register, create extension 1234
1454 ;callerid="Jane Smith" <5678>
1455 ;host=dynamic ; This device needs to register
1456 ;directmedia=no ; Typically set to NO if behind NAT
1458 ;allow=gsm ; GSM consumes far less bandwidth than ulaw
1461 ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
1462 ;registertrying=yes ; Send a 100 Trying when the device registers.
1465 ;type=friend ; Friends place calls and receive calls
1466 ;context=from-sip ; Context for incoming calls from this user
1468 ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
1469 ;language=de ; Use German prompts for this user
1470 ;host=dynamic ; This peer register with us
1471 ;dtmfmode=inband ; Choices are inband, rfc2833, or info
1472 ;defaultip=192.168.0.59 ; IP used until peer registers
1473 ;mailbox=1234@context,2345@context ; Mailbox(-es) for message waiting indicator
1474 ;subscribemwi=yes ; Only send notifications if this phone
1475 ; subscribes for mailbox notification
1476 ;vmexten=voicemail ; dialplan extension to reach mailbox
1477 ; sets the Message-Account in the MWI notify message
1478 ; defaults to global vmexten which defaults to "asterisk"
1480 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
1484 ;type=friend ; Friends place calls and receive calls
1485 ;context=from-sip ; Context for incoming calls from this user
1487 ;host=dynamic ; This peer register with us
1488 ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
1489 ;defaultuser=polly ; Username to use in INVITE until peer registers
1490 ;defaultip=192.168.40.123
1491 ; Normally you do NOT need to set this parameter
1493 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
1494 ;progressinband=no ; Polycom phones don't work properly with "never"
1501 ;insecure=port ; Allow matching of peer by IP address without
1502 ; matching port number
1503 ;insecure=invite ; Do not require authentication of incoming INVITEs
1504 ;insecure=port,invite ; (both)
1505 ;qualify=1000 ; Consider it down if it's 1 second to reply
1506 ; Helps with NAT session
1507 ; qualify=yes uses default value
1508 ;qualifyfreq=60 ; Qualification: How often to check for the
1509 ; host to be up in seconds
1510 ; Set to low value if you use low timeout for
1511 ; NAT of UDP sessions
1513 ; Call group and Pickup group should be in the range from 0 to 63
1515 ;callgroup=1,3-4 ; We are in caller groups 1,3,4
1516 ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
1517 ;namedcallgroup=engineering,sales,netgroup,protgroup ; We are in named call groups engineering,sales,netgroup,protgroup
1518 ;namedpickupgroup=sales ; We can do call pick-p for named call group sales
1519 ;defaultip=192.168.0.60 ; IP address to use if peer has not registered
1520 ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
1521 ;permit=192.168.0.60/255.255.255.0
1522 ;permit=192.168.0.60/24 ; we can also use CIDR notation for subnet masks
1523 ;permit=2001:db8::/32 ; IPv6 ACLs can be specified if desired. IPv6 ACLs
1524 ; apply only to IPv6 addresses, and IPv4 ACLs apply
1525 ; only to IPv4 addresses.
1526 ;acl=named_acl_example ; Use named ACLs defined in acl.conf
1531 ;qualify=200 ; Qualify peer is no more than 200ms away
1532 ;host=dynamic ; This device registers with us
1533 ;directmedia=no ; Asterisk by default tries to redirect the
1534 ; RTP media stream (audio) to go directly from
1535 ; the caller to the callee. Some devices do not
1536 ; support this (especially if one of them is
1538 ;defaultip=192.168.0.4 ; IP address to use until registration
1539 ;defaultuser=goran ; Username to use when calling this device before registration
1540 ; Normally you do NOT need to set this parameter
1541 ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from or to this device
1542 ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
1543 ; cause the given audio file to
1544 ; be played upon completion of
1545 ; an attended transfer to the
1546 ; target of the transfer.
1552 ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
1553 ; You must have this turned on or DTMF reception will work improperly.
1554 ;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
1555 ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
1556 ; external IP address of the remote device. If port forwarding is done at the client side
1557 ; then UDPTL will flow to the remote device.
projects / asterisk/asterisk.git / blob