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  1. ;
  2. ; SIP Configuration example for Asterisk
  3. ;
  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
  9. ; Asterisk.
  10. ;
  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
  16. ;
  17. ; SIP dial strings
  18. ;-----------------------------------------------------------
  19. ; In the dialplan (extensions.conf) you can use several
  20. ; syntaxes for dialing SIP devices.
  21. ; SIP/devicename
  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
  27. ;
  28. ;
  29. ; Devicename
  30. ; devicename is defined as a peer in a section below.
  31. ;
  32. ; username@domain
  33. ; Call any SIP user on the Internet
  34. ; (Don't forget to enable DNS SRV records if you want to use this)
  35. ;
  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
  41. ;
  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.
  45. ; Examples:
  46. ;
  47. ; SIP/*98@mysipproxy
  48. ; SIP/sales:topsecret::account02@domain.com:5062
  49. ; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
  50. ;
  51. ; IPorHost
  52. ; The next server for this call regardless of domain/peer
  53. ;
  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
  57. ;
  58. ; SIP/sales@mysipproxy!sales@edvina.net
  59. ;
  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:
  64. ;
  65. ; SIP/devicename/extension/IPorHost
  66. ; SIP/username@domain//IPorHost
  67. ;
  68. ; CLI Commands
  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
  73. ;
  74. ; sip set debug on Show all SIP messages
  75. ;
  76. ; sip reload Reload configuration file
  77. ; sip show settings Show the current channel configuration
  78. ;
  79. ;------- Naming devices ------------------------------------------------------
  80. ;
  81. ; When naming devices, make sure you understand how Asterisk matches calls
  82. ; that come in.
  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
  87. ; with a type=peer
  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
  90. ;
  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).
  94. ;
  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.
  98. ;
  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. ;-----------------------------------------------------------------------------
  104.  
  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.
  113.  
  114. [general]
  115. context=local ; 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
  150. ; will disable it.
  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
  155. ; will disable it.
  156.  
  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.)
  166. ;
  167. ; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061
  168. ; for TLS).
  169. ; IPv4 example: bindaddr=0.0.0.0:5062
  170. ; IPv6 example: bindaddr=[::]:5062
  171. ;
  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
  178. ; priority.
  179. bindport=5060
  180. bindaddr=0.0.0.0
  181. udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
  182. ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
  183.  
  184. ; When a dialog is started with another SIP endpoint, the other endpoint
  185. ; should include an Allow header telling us what SIP methods the endpoint
  186. ; implements. However, some endpoints either do not include an Allow header
  187. ; or lie about what methods they implement. In the former case, Asterisk
  188. ; makes the assumption that the endpoint supports all known SIP methods.
  189. ; If you know that your SIP endpoint does not provide support for a specific
  190. ; method, then you may provide a comma-separated list of methods that your
  191. ; endpoint does not implement in the disallowed_methods option. Note that
  192. ; if your endpoint is truthful with its Allow header, then there is no need
  193. ; to set this option. This option may be set in the general section or may
  194. ; be set per endpoint. If this option is set both in the general section and
  195. ; in a peer section, then the peer setting completely overrides the general
  196. ; setting (i.e. the result is *not* the union of the two options).
  197. ;
  198. ; Note also that while Asterisk currently will parse an Allow header to learn
  199. ; what methods an endpoint supports, the only actual use for this currently
  200. ; is for determining if Asterisk may send connected line UPDATE requests and
  201. ; MESSAGE requests. Its use may be expanded in the future.
  202. ;
  203. ; disallowed_methods = UPDATE
  204.  
  205. ;
  206. ; Note that the TCP and TLS support for chan_sip is currently considered
  207. ; experimental. Since it is new, all of the related configuration options are
  208. ; subject to change in any release. If they are changed, the changes will
  209. ; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
  210. ;
  211. tcpenable=no ; Enable server for incoming TCP connections (default is no)
  212. tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
  213. ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
  214.  
  215. ;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
  216. ;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
  217. ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
  218. ; Remember that the IP address must match the common name (hostname) in the
  219. ; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
  220. ; For details how to construct a certificate for SIP see
  221. ; http://tools.ietf.org/html/draft-ietf-sip-domain-certs
  222.  
  223. ;tcpauthtimeout = 30 ; tcpauthtimeout specifies the maximum number
  224. ; of seconds a client has to authenticate. If
  225. ; the client does not authenticate beofre this
  226. ; timeout expires, the client will be
  227. ; disconnected. (default: 30 seconds)
  228.  
  229. ;tcpauthlimit = 100 ; tcpauthlimit specifies the maximum number of
  230. ; unauthenticated sessions that will be allowed
  231. ; to connect at any given time. (default: 100)
  232.  
  233. ;websocket_write_timeout = 100 ; Default write timeout to set on websocket transports.
  234. ; This value may need to be adjusted for connections where
  235. ; Asterisk must write a substantial amount of data and the
  236. ; receiving clients are slow to process the received information.
  237. ; Value is in milliseconds; default is 100 ms.
  238.  
  239. transport=udp ; Set the default transports. The order determines the primary default transport.
  240. ; If tcpenable=no and the transport set is tcp, we will fallback to UDP.
  241.  
  242. srvlookup=yes ; Enable DNS SRV lookups on outbound calls
  243. ; Note: Asterisk only uses the first host
  244. ; in SRV records
  245. ; Disabling DNS SRV lookups disables the
  246. ; ability to place SIP calls based on domain
  247. ; names to some other SIP users on the Internet
  248. ; Specifying a port in a SIP peer definition or
  249. ; when dialing outbound calls will supress SRV
  250. ; lookups for that peer or call.
  251.  
  252. ;pedantic=yes ; Enable checking of tags in headers,
  253. ; international character conversions in URIs
  254. ; and multiline formatted headers for strict
  255. ; SIP compatibility (defaults to "yes")
  256.  
  257. ; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
  258. ;tos_sip=cs3 ; Sets TOS for SIP packets.
  259. ;tos_audio=ef ; Sets TOS for RTP audio packets.
  260. ;tos_video=af41 ; Sets TOS for RTP video packets.
  261. ;tos_text=af41 ; Sets TOS for RTP text packets.
  262.  
  263. ;cos_sip=3 ; Sets 802.1p priority for SIP packets.
  264. ;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
  265. ;cos_video=4 ; Sets 802.1p priority for RTP video packets.
  266. ;cos_text=3 ; Sets 802.1p priority for RTP text packets.
  267.  
  268. ;maxexpiry=3600 ; Maximum allowed time of incoming registrations (seconds)
  269. ;minexpiry=60 ; Minimum length of registrations (default 60)
  270. ;defaultexpiry=120 ; Default length of incoming/outgoing registration
  271. ;submaxexpiry=3600 ; Maximum allowed time of incoming subscriptions (seconds), default: maxexpiry
  272. ;subminexpiry=60 ; Minimum length of subscriptions, default: minexpiry
  273. ;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
  274. ;maxforwards=70 ; Setting for the SIP Max-Forwards: header (loop prevention)
  275. ; Default value is 70
  276. ;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds
  277. ; and reported in milliseconds with sip show settings.
  278. ; Set to low value if you use low timeout for NAT of UDP sessions
  279. ; Default: 60
  280. ;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
  281. ; Default: 100
  282. ;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
  283. ; Default: 1
  284. ;keepalive=60 ; Interval at which keepalive packets should be sent to a peer
  285. ; Valid options are yes (60 seconds), no, or the number of seconds.
  286. ; Default: 0
  287. ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
  288. ;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
  289. ; fully. Enable this option to not get error messages
  290. ; when sending MWI to phones with this bug.
  291. ;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in
  292. ; the From: header as the "name" portion. Also fill the
  293. ; "user" portion of the URI in the From: header with this
  294. ; value if no fromuser is set
  295. ; Default: empty
  296. ;vmexten=voicemail ; dialplan extension to reach mailbox sets the
  297. ; Message-Account in the MWI notify message
  298. ; defaults to "asterisk"
  299.  
  300. ; Codec negotiation
  301. ;
  302. ; When Asterisk is receiving a call, the codec will initially be set to the
  303. ; first codec in the allowed codecs defined for the user receiving the call
  304. ; that the caller also indicates that it supports. But, after the caller
  305. ; starts sending RTP, Asterisk will switch to using whatever codec the caller
  306. ; is sending.
  307. ;
  308. ; When Asterisk is placing a call, the codec used will be the first codec in
  309. ; the allowed codecs that the callee indicates that it supports. Asterisk will
  310. ; *not* switch to whatever codec the callee is sending.
  311. ;
  312. ;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
  313. ; rather than advertising all joint codec capabilities. This
  314. ; limits the other side's codec choice to exactly what we prefer.
  315.  
  316. ;disallow=all ; First disallow all codecs
  317. ;allow=ulaw ; Allow codecs in order of preference
  318. ;allow=ilbc ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
  319. ; for framing options
  320. ;autoframing=yes ; Set packetization based on the remote endpoint's (ptime)
  321. ; preferences. Defaults to no.
  322. ;
  323. ; This option specifies a preference for which music on hold class this channel
  324. ; should listen to when put on hold if the music class has not been set on the
  325. ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
  326. ; channel putting this one on hold did not suggest a music class.
  327. ;
  328. ; This option may be specified globally, or on a per-user or per-peer basis.
  329. ;
  330. ;mohinterpret=default
  331. ;
  332. ; This option specifies which music on hold class to suggest to the peer channel
  333. ; when this channel places the peer on hold. It may be specified globally or on
  334. ; a per-user or per-peer basis.
  335. ;
  336. ;mohsuggest=default
  337. ;
  338. ;parkinglot=plaza ; Sets the default parking lot for call parking
  339. ; This may also be set for individual users/peers
  340. ; Parkinglots are configured in features.conf
  341. language=fr ; Default language setting for all users/peers
  342. ; This may also be set for individual users/peers
  343. ;tonezone=se ; Default tonezone for all users/peers
  344. ; This may also be set for individual users/peers
  345.  
  346. ;relaxdtmf=yes ; Relax dtmf handling
  347. ;trustrpid = no ; If Remote-Party-ID should be trusted
  348. ;sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no)
  349. ;sendrpid = rpid ; Use the "Remote-Party-ID" header
  350. ; to send the identity of the remote party
  351. ; This is identical to sendrpid=yes
  352. ;sendrpid = pai ; Use the "P-Asserted-Identity" header
  353. ; to send the identity of the remote party
  354. ;rpid_update = no ; In certain cases, the only method by which a connected line
  355. ; change may be immediately transmitted is with a SIP UPDATE request.
  356. ; If communicating with another Asterisk server, and you wish to be able
  357. ; transmit such UPDATE messages to it, then you must enable this option.
  358. ; Otherwise, we will have to wait until we can send a reinvite to
  359. ; transmit the information.
  360. ;trust_id_outbound = no ; Controls whether or not we trust this peer with private identity
  361. ; information (when the remote party has callingpres=prohib or equivalent).
  362. ; no - RPID/PAI headers will not be included for private peer information
  363. ; yes - RPID/PAI headers will include the private peer information. Privacy
  364. ; requirements will be indicated in a Privacy header for sendrpid=pai
  365. ; legacy - RPID/PAI will be included for private peer information. In the
  366. ; case of sendrpid=pai, private data that would be included in them
  367. ; will be anonymized. For sendrpid=rpid, private data may be included
  368. ; but the remote party's domain will be anonymized. The way legacy
  369. ; behaves may violate RFC-3325, but it follows historic behavior.
  370. ; This option is set to 'legacy' by default
  371. ;prematuremedia=no ; Some ISDN links send empty media frames before
  372. ; the call is in ringing or progress state. The SIP
  373. ; channel will then send 183 indicating early media
  374. ; which will be empty - thus users get no ring signal.
  375. ; Setting this to "yes" will stop any media before we have
  376. ; call progress (meaning the SIP channel will not send 183 Session
  377. ; Progress for early media). Default is "yes". Also make sure that
  378. ; the SIP peer is configured with progressinband=never.
  379. ;
  380. ; In order for "noanswer" applications to work, you need to run
  381. ; the progress() application in the priority before the app.
  382.  
  383. ;progressinband=never ; If we should generate in-band ringing always
  384. ; use 'never' to never use in-band signalling, even in cases
  385. ; where some buggy devices might not render it
  386. ; Valid values: yes, no, never Default: never
  387. ;useragent=Asterisk PBX ; Allows you to change the user agent string
  388. ; The default user agent string also contains the Asterisk
  389. ; version. If you don't want to expose this, change the
  390. ; useragent string.
  391. ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
  392. ; Note that promiscredir when redirects are made to the
  393. ; local system will cause loops since Asterisk is incapable
  394. ; of performing a "hairpin" call.
  395. ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
  396. ; a valid phone number
  397. ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
  398. ; Other options:
  399. ; info : SIP INFO messages (application/dtmf-relay)
  400. ; shortinfo : SIP INFO messages (application/dtmf)
  401. ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
  402. ; auto : Use rfc2833 if offered, inband otherwise
  403.  
  404. ;compactheaders = yes ; send compact sip headers.
  405. ;
  406. ;videosupport=yes ; Turn on support for SIP video. You need to turn this
  407. ; on in this section to get any video support at all.
  408. ; You can turn it off on a per peer basis if the general
  409. ; video support is enabled, but you can't enable it for
  410. ; one peer only without enabling in the general section.
  411. ; If you set videosupport to "always", then RTP ports will
  412. ; always be set up for video, even on clients that don't
  413. ; support it. This assists callfile-derived calls and
  414. ; certain transferred calls to use always use video when
  415. ; available. [yes|NO|always]
  416.  
  417. ;textsupport=no ; Support for ITU-T T.140 realtime text.
  418. ; The default value is "no".
  419.  
  420. ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
  421. ; Videosupport and maxcallbitrate is settable
  422. ; for peers and users as well
  423. ;callevents=no ; generate manager events when sip ua
  424. ; performs events (e.g. hold)
  425. ;authfailureevents=no ; generate manager "peerstatus" events when peer can't
  426. ; authenticate with Asterisk. Peerstatus will be "rejected".
  427. ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
  428. ; for any reason, always reject with an identical response
  429. ; equivalent to valid username and invalid password/hash
  430. ; instead of letting the requester know whether there was
  431. ; a matching user or peer for their request. This reduces
  432. ; the ability of an attacker to scan for valid SIP usernames.
  433. ; This option is set to "yes" by default.
  434.  
  435. ;auth_options_requests = yes ; Enabling this option will authenticate OPTIONS requests just like
  436. ; INVITE requests are. By default this option is disabled.
  437.  
  438. ;accept_outofcall_message = no ; Disable this option to reject all MESSAGE requests outside of a
  439. ; call. By default, this option is enabled. When enabled, MESSAGE
  440. ; requests are passed in to the dialplan.
  441.  
  442. ;outofcall_message_context = messages ; Context all out of dialog msgs are sent to. When this
  443. ; option is not set, the context used during peer matching
  444. ; is used. This option can be defined at both the peer and
  445. ; global level.
  446.  
  447. ;auth_message_requests = yes ; Enabling this option will authenticate MESSAGE requests.
  448. ; By default this option is enabled. However, it can be disabled
  449. ; should an application desire to not load the Asterisk server with
  450. ; doing authentication and implement end to end security in the
  451. ; message body.
  452.  
  453. ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
  454. ; order instead of RFC3551 packing order (this is required
  455. ; for Sipura and Grandstream ATAs, among others). This is
  456. ; contrary to the RFC3551 specification, the peer _should_
  457. ; be negotiating AAL2-G726-32 instead :-(
  458. ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
  459. ;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
  460. ;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
  461. ;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls
  462. ;outboundproxy=192.0.2.1 ; IPv4 address literal (default port is 5060)
  463. ;outboundproxy=2001:db8::1 ; IPv6 address literal (default port is 5060)
  464. ;outboundproxy=192.168.0.2.1:5062 ; IPv4 address literal with explicit port
  465. ;outboundproxy=[2001:db8::1]:5062 ; IPv6 address literal with explicit port
  466. ; ; (could also be tcp,udp) - defining transports on the proxy line only
  467. ; ; applies for the global proxy, otherwise use the transport= option
  468. ;matchexternaddrlocally = yes ; Only substitute the externaddr or externhost setting if it matches
  469. ; your localnet setting. Unless you have some sort of strange network
  470. ; setup you will not need to enable this.
  471.  
  472. ;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering
  473. ; as any IP address used for staticly defined
  474. ; hosts. This helps avoid the configuration
  475. ; error of allowing your users to register at
  476. ; the same address as a SIP provider.
  477.  
  478. ;contactdeny=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to
  479. ;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may
  480. ; register their phones.
  481. ;contactacl=named_acl_example ; Use named ACLs defined in acl.conf
  482.  
  483. ;rtp_engine=asterisk ; RTP engine to use when communicating with the device
  484.  
  485. ;
  486. ; If regcontext is specified, Asterisk will dynamically create and destroy a
  487. ; NoOp priority 1 extension for a given peer who registers or unregisters with
  488. ; us and have a "regexten=" configuration item.
  489. ; Multiple contexts may be specified by separating them with '&'. The
  490. ; actual extension is the 'regexten' parameter of the registering peer or its
  491. ; name if 'regexten' is not provided. If more than one context is provided,
  492. ; the context must be specified within regexten by appending the desired
  493. ; context after '@'. More than one regexten may be supplied if they are
  494. ; separated by '&'. Patterns may be used in regexten.
  495. ;
  496. ;regcontext=sipregistrations
  497. ;regextenonqualify=yes ; Default "no"
  498. ; If you have qualify on and the peer becomes unreachable
  499. ; this setting will enforce inactivation of the regexten
  500. ; extension for the peer
  501. ;legacy_useroption_parsing=yes ; Default "no" ; If you have this option enabled and there are semicolons
  502. ; in the user field of a sip URI, the field be truncated
  503. ; at the first semicolon seen. This effectively makes
  504. ; semicolon a non-usable character for peer names, extensions,
  505. ; and maybe other, less tested things. This can be useful
  506. ; for improving compatability with devices that like to use
  507. ; user options for whatever reason. The behavior is similar to
  508. ; how SIP URI's were typically handled in 1.6.2, hence the name.
  509.  
  510. ;send_diversion=no ; Default "yes" ; Asterisk normally sends Diversion headers with certain SIP
  511. ; invites to relay data about forwarded calls. If this option
  512. ; is disabled, Asterisk won't send Diversion headers unless
  513. ; they are added manually.
  514.  
  515. ; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
  516. ; in square brackets. For example, the caller id value 555.5555 becomes 5555555
  517. ; when this option is enabled. Disabling this option results in no modification
  518. ; of the caller id value, which is necessary when the caller id represents something
  519. ; that must be preserved. This option can only be used in the [general] section.
  520. ; By default this option is on.
  521. ;
  522. ;shrinkcallerid=yes ; on by default
  523.  
  524.  
  525. ;use_q850_reason = no ; Default "no"
  526. ; Set to yes add Reason header and use Reason header if it is available.
  527.  
  528. ; When the Transfer() application sends a REFER SIP message, extra headers specified in
  529. ; the dialplan by way of SIPAddHeader are sent out with that message. 1.8 and earlier did not
  530. ; add the extra headers. To revert to 1.8- behavior, call SIPRemoveHeader with no arguments
  531. ; before calling Transfer() to remove all additional headers from the channel. The setting
  532. ; below is for transitional compatibility only.
  533. ;
  534. ;refer_addheaders=yes ; on by default
  535.  
  536. ;autocreatepeer=no ; Allow any UAC not explicitly defined to register
  537. ; WITHOUT AUTHENTICATION. Enabling this options poses a high
  538. ; potential security risk and should be avoided unless the
  539. ; server is behind a trusted firewall.
  540. ; If set to "yes", then peers created in this fashion
  541. ; are purged during SIP reloads.
  542. ; When set to "persist", the peers created in this fashion
  543. ; are not purged during SIP reloads.
  544.  
  545. ;
  546. ;------------------------ TLS settings ------------------------------------------------------------
  547. ;tlscertfile=</path/to/certificate.pem> ; Certificate chain (*.pem format only) to use for TLS connections
  548. ; The certificates must be sorted starting with the subject's certificate
  549. ; and followed by intermediate CA certificates if applicable.
  550. ; Default is to look for "asterisk.pem" in current directory
  551.  
  552. ;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem format only) for TLS connections.
  553. ; If no tlsprivatekey is specified, tlscertfile is searched for
  554. ; for both public and private key.
  555.  
  556. ;tlscafile=</path/to/certificate>
  557. ; If the server your connecting to uses a self signed certificate
  558. ; you should have their certificate installed here so the code can
  559. ; verify the authenticity of their certificate.
  560.  
  561. ;tlscapath=</path/to/ca/dir>
  562. ; A directory full of CA certificates. The files must be named with
  563. ; the CA subject name hash value.
  564. ; (see man SSL_CTX_load_verify_locations for more info)
  565.  
  566. ;tlsdontverifyserver=[yes|no]
  567. ; If set to yes, don't verify the servers certificate when acting as
  568. ; a client. If you don't have the server's CA certificate you can
  569. ; set this and it will connect without requiring tlscafile to be set.
  570. ; Default is no.
  571.  
  572. ;tlscipher=<SSL cipher string>
  573. ; A string specifying which SSL ciphers to use or not use
  574. ; A list of valid SSL cipher strings can be found at:
  575. ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
  576. ;
  577. ;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2.
  578. ; Specify protocol for outbound client connections.
  579. ; If left unspecified, the default is sslv2.
  580. ;
  581. ;--------------------------- SIP timers ----------------------------------------------------
  582. ; These timers are used primarily in INVITE transactions.
  583. ; The default for Timer T1 is 500 ms or the measured run-trip time between
  584. ; Asterisk and the device if you have qualify=yes for the device.
  585. ;
  586. ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
  587. ; Defaults to 100 ms
  588. ;timert1=500 ; Default T1 timer
  589. ; Defaults to 500 ms or the measured round-trip
  590. ; time to a peer (qualify=yes).
  591. ;timerb=32000 ; Call setup timer. If a provisional response is not received
  592. ; in this amount of time, the call will autocongest
  593. ; Defaults to 64*timert1
  594.  
  595. ;--------------------------- RTP timers ----------------------------------------------------
  596. ; These timers are currently used for both audio and video streams. The RTP timeouts
  597. ; are only applied to the audio channel.
  598. ; The settings are settable in the global section as well as per device
  599. ;
  600. ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
  601. ; on the audio channel
  602. ; when we're not on hold. This is to be able to hangup
  603. ; a call in the case of a phone disappearing from the net,
  604. ; like a powerloss or grandma tripping over a cable.
  605. ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
  606. ; on the audio channel
  607. ; when we're on hold (must be > rtptimeout)
  608. ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
  609. ; (default is off - zero)
  610.  
  611. ;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
  612. ; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
  613. ; This mechanism can detect and reclaim SIP channels that do not terminate through normal
  614. ; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
  615. ; The operation of Session-Timers is driven by the following configuration parameters:
  616. ;
  617. ; * session-timers - Session-Timers feature operates in the following three modes:
  618. ; originate : Request and run session-timers always
  619. ; accept : Run session-timers only when requested by other UA
  620. ; refuse : Do not run session timers in any case
  621. ; The default mode of operation is 'accept'.
  622. ; * session-expires - Maximum session refresh interval in seconds. Defaults to 1800 secs.
  623. ; * session-minse - Minimum session refresh interval in seconds. Defualts to 90 secs.
  624. ; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
  625. ; uac - Default to the caller initially refreshing when possible
  626. ; uas - Default to the callee initially refreshing when possible
  627. ;
  628. ; Note that, due to recommendations in RFC 4028, Asterisk will always honor the other
  629. ; endpoint's preference for who will handle refreshes. Asterisk will never override the
  630. ; preferences of the other endpoint. Doing so could result in Asterisk and the endpoint
  631. ; fighting over who sends the refreshes. This holds true for the initiation of session
  632. ; timers and subsequent re-INVITE requests whether Asterisk is the caller or callee, or
  633. ; whether Asterisk is currently the refresher or not.
  634. ;
  635. ;session-timers=originate
  636. ;session-expires=600
  637. ;session-minse=90
  638. ;session-refresher=uac
  639. ;
  640. ;--------------------------- SIP DEBUGGING ---------------------------------------------------
  641. ;sipdebug = yes ; Turn on SIP debugging by default, from
  642. ; the moment the channel loads this configuration
  643. ;recordhistory=yes ; Record SIP history by default
  644. ; (see sip history / sip no history)
  645. ;dumphistory=yes ; Dump SIP history at end of SIP dialogue
  646. ; SIP history is output to the DEBUG logging channel
  647.  
  648.  
  649. ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
  650. ; You can subscribe to the status of extensions with a "hint" priority
  651. ; (See extensions.conf.sample for examples)
  652. ; chan_sip support two major formats for notifications: dialog-info and SIMPLE
  653. ;
  654. ; You will get more detailed reports (busy etc) if you have a call counter enabled
  655. ; for a device.
  656. ;
  657. ; If you set the busylevel, we will indicate busy when we have a number of calls that
  658. ; matches the busylevel treshold.
  659. ;
  660. ; For queues, you will need this level of detail in status reporting, regardless
  661. ; if you use SIP subscriptions. Queues and manager use the same internal interface
  662. ; for reading status information.
  663. ;
  664. ; Note: Subscriptions does not work if you have a realtime dialplan and use the
  665. ; realtime switch.
  666. ;
  667. ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
  668. ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
  669. ; Useful to limit subscriptions to local extensions
  670. ; Settable per peer/user also
  671. ;notifyringing = no ; Control whether subscriptions already INUSE get sent
  672. ; RINGING when another call is sent (default: yes)
  673. ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
  674. ; Turning on notifyringing and notifyhold will add a lot
  675. ; more database transactions if you are using realtime.
  676. ;notifycid = yes ; Control whether caller ID information is sent along with
  677. ; dialog-info+xml notifications (supported by snom phones).
  678. ; Note that this feature will only work properly when the
  679. ; incoming call is using the same extension and context that
  680. ; is being used as the hint for the called extension. This means
  681. ; that it won't work when using subscribecontext for your sip
  682. ; user or peer (if subscribecontext is different than context).
  683. ; This is also limited to a single caller, meaning that if an
  684. ; extension is ringing because multiple calls are incoming,
  685. ; only one will be used as the source of caller ID. Specify
  686. ; 'ignore-context' to ignore the called context when looking
  687. ; for the caller's channel. The default value is 'no.' Setting
  688. ; notifycid to 'ignore-context' also causes call-pickups attempted
  689. ; via SNOM's NOTIFY mechanism to set the context for the call pickup
  690. ; to PICKUPMARK.
  691. ;callcounter = yes ; Enable call counters on devices. This can be set per
  692. ; device too.
  693.  
  694. ;----------------------------------------- T.38 FAX SUPPORT ----------------------------------
  695. ;
  696. ; This setting is available in the [general] section as well as in device configurations.
  697. ; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off.
  698. ;
  699. ; t38pt_udptl = yes ; Enables T.38 with FEC error correction.
  700. ; t38pt_udptl = yes,fec ; Enables T.38 with FEC error correction.
  701. ; t38pt_udptl = yes,redundancy ; Enables T.38 with redundancy error correction.
  702. ; t38pt_udptl = yes,none ; Enables T.38 with no error correction.
  703. ;
  704. ; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that
  705. ; is based on an incorrect interpretation of the T.38 recommendation, and results in failures
  706. ; because Asterisk does not believe it can send T.38 packets of a reasonable size to that
  707. ; endpoint (Cisco media gateways are one example of this situation). In these cases, during a
  708. ; T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL
  709. ; stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you
  710. ; can set an override (globally, or on a per-device basis) to make Asterisk ignore the
  711. ; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead.
  712. ; This can be done by appending 'maxdatagram=<value>' to the t38pt_udptl configuration option,
  713. ; like this:
  714. ;
  715. ; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error correction and overrides
  716. ; ; the other endpoint's provided value to assume we can
  717. ; ; send 400 byte T.38 FAX packets to it.
  718. ;
  719. ; FAX detection will cause the SIP channel to jump to the 'fax' extension (if it exists)
  720. ; based one or more events being detected. The events that can be detected are an incoming
  721. ; CNG tone or an incoming T.38 re-INVITE request.
  722. ;
  723. ; faxdetect = yes ; Default 'no', 'yes' enables both CNG and T.38 detection
  724. ; faxdetect = cng ; Enables only CNG detection
  725. ; faxdetect = t38 ; Enables only T.38 detection
  726. ;
  727. ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
  728. ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
  729. ; Format for the register statement is:
  730. ; register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
  731. ;
  732. ;
  733. ;
  734. ; domain is either
  735. ; - domain in DNS
  736. ; - host name in DNS
  737. ; - the name of a peer defined below or in realtime
  738. ; The domain is where you register your username, so your SIP uri you are registering to
  739. ; is username@domain
  740. ;
  741. ; If no extension is given, the 's' extension is used. The extension needs to
  742. ; be defined in extensions.conf to be able to accept calls from this SIP proxy
  743. ; (provider).
  744. ;
  745. ; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
  746. ; this is equivalent to having the following line in the general section:
  747. ;
  748. ; register => username:secret@host/callbackextension
  749. ;
  750. ; and more readable because you don't have to write the parameters in two places
  751. ; (note that the "port" is ignored - this is a bug that should be fixed).
  752. ;
  753. ; Note that a register= line doesn't mean that we will match the incoming call in any
  754. ; other way than described above. If you want to control where the call enters your
  755. ; dialplan, which context, you want to define a peer with the hostname of the provider's
  756. ; server. If the provider has multiple servers to place calls to your system, you need
  757. ; a peer for each server.
  758. ;
  759. ; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may
  760. ; contain a port number. Since the logical separator between a host and port number is a
  761. ; ':' character, and this character is already used to separate between the optional "secret"
  762. ; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish
  763. ; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
  764. ; they are blank. See the third example below for an illustration.
  765. ;
  766. ;
  767. ; Examples:
  768. ;
  769. ;register => 1234:password@mysipprovider.com
  770. ;
  771. ; This will pass incoming calls to the 's' extension
  772. ;
  773. ;
  774. ;register => 2345:password@sip_proxy/1234
  775. ;
  776. ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
  777. ; connect to local extension 1234 in extensions.conf, default context,
  778. ; unless you configure a [sip_proxy] section below, and configure a
  779. ; context.
  780. ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
  781. ; Tip 2: Use separate inbound and outbound sections for SIP providers
  782. ; (instead of type=friend) if you have calls in both directions
  783. ;
  784. ;register => 3456@mydomain:5082::@mysipprovider.com
  785. ;
  786. ; Note that in this example, the optional authuser and secret portions have
  787. ; been left blank because we have specified a port in the user section
  788. ;
  789. ;register => tls://username:xxxxxx@sip-tls-proxy.example.org
  790. ;
  791. ; The 'transport' part defaults to 'udp' but may also be 'tcp', 'tls', 'ws', or 'wss'.
  792. ; Using 'udp://' explicitly is also useful in case the username part
  793. ; contains a '/' ('user/name').
  794.  
  795. ;registertimeout=20 ; retry registration calls every 20 seconds (default)
  796. ;registerattempts=10 ; Number of registration attempts before we give up
  797. ; 0 = continue forever, hammering the other server
  798. ; until it accepts the registration
  799. ; Default is 0 tries, continue forever
  800. ;register_retry_403=yes ; Treat 403 responses to registrations as if they were
  801. ; 401 responses and continue retrying according to normal
  802. ; retry rules.
  803.  
  804. ;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
  805. ; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
  806. ; by other phones. At this time, you can only subscribe using UDP as the transport.
  807. ; Format for the mwi register statement is:
  808. ; mwi => user[:secret[:authuser]]@host[:port]/mailbox
  809. ;
  810. ; Examples:
  811. ;mwi => 1234:password@mysipprovider.com/1234
  812. ;mwi => 1234:password@myportprovider.com:6969/1234
  813. ;mwi => 1234:password:authuser@myauthprovider.com/1234
  814. ;mwi => 1234:password:authuser@myauthportprovider.com:6969/1234
  815. ;
  816. ; 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:
  817. ; mailbox=1234@SIP_Remote
  818. ;----------------------------------------- NAT SUPPORT ------------------------
  819. ;
  820. ; WARNING: SIP operation behind a NAT is tricky and you really need
  821. ; to read and understand well the following section.
  822. ;
  823. ; When Asterisk is behind a NAT device, the "local" address (and port) that
  824. ; a socket is bound to has different values when seen from the inside or
  825. ; from the outside of the NATted network. Unfortunately this address must
  826. ; be communicated to the outside (e.g. in SIP and SDP messages), and in
  827. ; order to determine the correct value Asterisk needs to know:
  828. ;
  829. ; + whether it is talking to someone "inside" or "outside" of the NATted network.
  830. ; This is configured by assigning the "localnet" parameter with a list
  831. ; of network addresses that are considered "inside" of the NATted network.
  832. ; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
  833. ; Multiple entries are allowed, e.g. a reasonable set is the following:
  834. ;
  835. ; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
  836. ; localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
  837. ; localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
  838. ; localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
  839. ;
  840. ; + the "externally visible" address and port number to be used when talking
  841. ; to a host outside the NAT. This information is derived by one of the
  842. ; following (mutually exclusive) config file parameters:
  843. ;
  844. ; a. "externaddr = hostname[:port]" specifies a static address[:port] to
  845. ; be used in SIP and SDP messages.
  846. ; The hostname is looked up only once, when [re]loading sip.conf .
  847. ; If a port number is not present, use the port specified in the "udpbindaddr"
  848. ; (which is not guaranteed to work correctly, because a NAT box might remap the
  849. ; port number as well as the address).
  850. ; This approach can be useful if you have a NAT device where you can
  851. ; configure the mapping statically. Examples:
  852. ;
  853. ; externaddr = 12.34.56.78 ; use this address.
  854. ; externaddr = 12.34.56.78:9900 ; use this address and port.
  855. ; externaddr = mynat.my.org:12600 ; Public address of my nat box.
  856. ; externtcpport = 9900 ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT.
  857. ; ; externtcpport will default to the externaddr or externhost port if either one is set.
  858. ; externtlsport = 12600 ; The externally mapped tls port, when Asterisk is behind a static NAT or PAT.
  859. ; ; externtlsport port will default to the RFC designated port of 5061.
  860. ;
  861. ; b. "externhost = hostname[:port]" is similar to "externaddr" except
  862. ; that the hostname is looked up every "externrefresh" seconds
  863. ; (default 10s). This can be useful when your NAT device lets you choose
  864. ; the port mapping, but the IP address is dynamic.
  865. ; Beware, you might suffer from service disruption when the name server
  866. ; resolution fails. Examples:
  867. ;
  868. ; externhost=foo.dyndns.net ; refreshed periodically
  869. ; externrefresh=180 ; change the refresh interval
  870. ;
  871. ; Note that at the moment all these mechanism work only for the SIP socket.
  872. ; The IP address discovered with externaddr/externhost is reused for
  873. ; media sessions as well, but the port numbers are not remapped so you
  874. ; may still experience problems.
  875. ;
  876. ; NOTE 1: in some cases, NAT boxes will use different port numbers in
  877. ; the internal<->external mapping. In these cases, the "externaddr" and
  878. ; "externhost" might not help you configure addresses properly.
  879. ;
  880. ; NOTE 2: when using "externaddr" or "externhost", the address part is
  881. ; also used as the external address for media sessions. Thus, the port
  882. ; information in the SDP may be wrong!
  883. ;
  884. ; In addition to the above, Asterisk has an additional "nat" parameter to
  885. ; address NAT-related issues in incoming SIP or media sessions.
  886. ; In particular, depending on the 'nat= ' settings described below, Asterisk
  887. ; may override the address/port information specified in the SIP/SDP messages,
  888. ; and use the information (sender address) supplied by the network stack instead.
  889. ; However, this is only useful if the external traffic can reach us.
  890. ; The following settings are allowed (both globally and in individual sections):
  891. ;
  892. ; nat = no ; Do no special NAT handling other than RFC3581
  893. ; nat = force_rport ; Pretend there was an rport parameter even if there wasn't
  894. ; nat = comedia ; Send media to the port Asterisk received it from regardless
  895. ; ; of where the SDP says to send it.
  896. ; nat = auto_force_rport ; Set the force_rport option if Asterisk detects NAT (default)
  897. ; nat = auto_comedia ; Set the comedia option if Asterisk detects NAT
  898. ;
  899. ; The nat settings can be combined. For example, to set both force_rport and comedia
  900. ; one would set nat=force_rport,comedia. If any of the comma-separated options is 'no',
  901. ; Asterisk will ignore any other settings and set nat=no. If one of the "auto" settings
  902. ; is used in conjunction with its non-auto counterpart (nat=comedia,auto_comedia), then
  903. ; the non-auto option will be ignored.
  904. ;
  905. ; The RFC 3581-defined 'rport' parameter allows a client to request that Asterisk send
  906. ; SIP responses to it via the source IP and port from which the request originated
  907. ; instead of the address/port listed in the top-most Via header. This is useful if a
  908. ; client knows that it is behind a NAT and therefore cannot guess from what address/port
  909. ; its request will be sent. Asterisk will always honor the 'rport' parameter if it is
  910. ; sent. The force_rport setting causes Asterisk to always send responses back to the
  911. ; address/port from which it received requests; even if the other side doesn't support
  912. ; adding the 'rport' parameter.
  913. ;
  914. ; 'comedia RTP handling' refers to the technique of sending RTP to the port that the
  915. ; the other endpoint's RTP arrived from, and means 'connection-oriented media'. This is
  916. ; only partially related to RFC 4145 which was referred to as COMEDIA while it was in
  917. ; draft form. This method is used to accomodate endpoints that may be located behind
  918. ; NAT devices, and as such the address/port they tell Asterisk to send RTP packets to
  919. ; for their media streams is not the actual address/port that will be used on the nearer
  920. ; side of the NAT.
  921. ;
  922. ; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
  923. ; the nat setting in a peer definition, then the peer username will be discoverable
  924. ; by outside parties as Asterisk will respond to different ports for defined and
  925. ; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
  926. ; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the
  927. ; other, then valid peers with settings differing from those in the general section will
  928. ; be discoverable.
  929. ;
  930. ; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by
  931. ; RFC 4961; Asterisk will always send RTP packets from the same port number it expects
  932. ; to receive them on.
  933. ;
  934. ; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using
  935. ; the media_address configuration option. This is only applicable to the general section and
  936. ; can not be set per-user or per-peer.
  937. ;
  938. ; media_address = 172.16.42.1
  939. ;
  940. ; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the
  941. ; perceived external network address has changed. When the stun_monitor is installed and
  942. ; configured, chan_sip will renew all outbound registrations when the monitor detects any sort
  943. ; of network change has occurred. By default this option is enabled, but only takes effect once
  944. ; res_stun_monitor is configured. If res_stun_monitor is enabled and you wish to not
  945. ; generate all outbound registrations on a network change, use the option below to disable
  946. ; this feature.
  947. ;
  948. ; subscribe_network_change_event = yes ; on by default
  949. ;
  950. ; ICE/STUN/TURN usage can be enabled globally or on a per-peer basis using the icesupport
  951. ; configuration option. When set to yes ICE support is enabled. When set to no it is disabled.
  952. ; It is disabled by default.
  953. ;
  954. ; icesupport = yes
  955.  
  956. ;----------------------------------- MEDIA HANDLING --------------------------------
  957. ; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
  958. ; no reason for Asterisk to stay in the media path, the media will be redirected.
  959. ; This does not really work well in the case where Asterisk is outside and the
  960. ; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
  961. ;
  962. ;directmedia=yes ; Asterisk by default tries to redirect the
  963. ; RTP media stream to go directly from
  964. ; the caller to the callee. Some devices do not
  965. ; support this (especially if one of them is behind a NAT).
  966. ; The default setting is YES. If you have all clients
  967. ; behind a NAT, or for some other reason want Asterisk to
  968. ; stay in the audio path, you may want to turn this off.
  969.  
  970. ; This setting also affect direct RTP
  971. ; at call setup (a new feature in 1.4 - setting up the
  972. ; call directly between the endpoints instead of sending
  973. ; a re-INVITE).
  974.  
  975. ; Additionally this option does not disable all reINVITE operations.
  976. ; It only controls Asterisk generating reINVITEs for the specific
  977. ; purpose of setting up a direct media path. If a reINVITE is
  978. ; needed to switch a media stream to inactive (when placed on
  979. ; hold) or to T.38, it will still be done, regardless of this
  980. ; setting. Note that direct T.38 is not supported.
  981.  
  982. ;directmedia=nonat ; An additional option is to allow media path redirection
  983. ; (reinvite) but only when the peer where the media is being
  984. ; sent is known to not be behind a NAT (as the RTP core can
  985. ; determine it based on the apparent IP address the media
  986. ; arrives from).
  987.  
  988. ;directmedia=update ; Yet a third option... use UPDATE for media path redirection,
  989. ; instead of INVITE. This can be combined with 'nonat', as
  990. ; 'directmedia=update,nonat'. It implies 'yes'.
  991.  
  992. ;directmedia=outgoing ; When sending directmedia reinvites, do not send an immediate
  993. ; reinvite on an incoming call leg. This option is useful when
  994. ; peered with another SIP user agent that is known to send
  995. ; immediate direct media reinvites upon call establishment. Setting
  996. ; the option in this situation helps to prevent potential glares.
  997. ; Setting this option implies 'yes'.
  998.  
  999. ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
  1000. ; the call directly with media peer-2-peer without re-invites.
  1001. ; Will not work for video and cases where the callee sends
  1002. ; RTP payloads and fmtp headers in the 200 OK that does not match the
  1003. ; callers INVITE. This will also fail if directmedia is enabled when
  1004. ; the device is actually behind NAT.
  1005.  
  1006. ;directmediadeny=0.0.0.0/0 ; Use directmediapermit and directmediadeny to restrict
  1007. ;directmediapermit=172.16.0.0/16; which peers should be able to pass directmedia to each other
  1008. ; (There is no default setting, this is just an example)
  1009. ; Use this if some of your phones are on IP addresses that
  1010. ; can not reach each other directly. This way you can force
  1011. ; RTP to always flow through asterisk in such cases.
  1012. ;directmediaacl=acl_example ; Use named ACLs defined in acl.conf
  1013.  
  1014. ;ignoresdpversion=yes ; By default, Asterisk will honor the session version
  1015. ; number in SDP packets and will only modify the SDP
  1016. ; session if the version number changes. This option will
  1017. ; force asterisk to ignore the SDP session version number
  1018. ; and treat all SDP data as new data. This is required
  1019. ; for devices that send us non standard SDP packets
  1020. ; (observed with Microsoft OCS). By default this option is
  1021. ; off.
  1022.  
  1023. ;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
  1024. ; Like the useragent parameter, the default user agent string
  1025. ; also contains the Asterisk version.
  1026. ;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
  1027. ; This field MUST NOT contain spaces
  1028. ;encryption=no ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
  1029. ; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
  1030. ; the peer does not support SRTP. Defaults to no.
  1031. ;encryption_taglen=80 ; Set the auth tag length offered in the INVITE either 32/80 default 80
  1032. ;
  1033. ;avpf=yes ; Enable inter-operability with media streams using the AVPF RTP profile.
  1034. ; This will cause all offers and answers to use AVPF (or SAVPF). This
  1035. ; option may be specified at the global or peer scope.
  1036. ;force_avp=yes ; Force 'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', and 'RTP/SAVPF' to be used for
  1037. ; media streams when appropriate, even if a DTLS stream is present.
  1038. ;----------------------------------------- REALTIME SUPPORT ------------------------
  1039. ; For additional information on ARA, the Asterisk Realtime Architecture,
  1040. ; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
  1041. ;
  1042. ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
  1043. ; just like friends added from the config file only on a
  1044. ; as-needed basis? (yes|no)
  1045.  
  1046. ;rtsavesysname=yes ; Save systemname in realtime database at registration
  1047. ; Default= no
  1048.  
  1049. ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
  1050. ; If set to yes, when a SIP UA registers successfully, the ip address,
  1051. ; the origination port, the registration period, and the username of
  1052. ; the UA will be set to database via realtime.
  1053. ; If not present, defaults to 'yes'. Note: realtime peers will
  1054. ; probably not function across reloads in the way that you expect, if
  1055. ; you turn this option off.
  1056. ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
  1057. ; as if it had just registered? (yes|no|<seconds>)
  1058. ; If set to yes, when the registration expires, the friend will
  1059. ; vanish from the configuration until requested again. If set
  1060. ; to an integer, friends expire within this number of seconds
  1061. ; instead of the registration interval.
  1062.  
  1063. ;ignoreregexpire=yes ; Enabling this setting has two functions:
  1064. ;
  1065. ; For non-realtime peers, when their registration expires, the
  1066. ; information will _not_ be removed from memory or the Asterisk database
  1067. ; if you attempt to place a call to the peer, the existing information
  1068. ; will be used in spite of it having expired
  1069. ;
  1070. ; For realtime peers, when the peer is retrieved from realtime storage,
  1071. ; the registration information will be used regardless of whether
  1072. ; it has expired or not; if it expires while the realtime peer
  1073. ; is still in memory (due to caching or other reasons), the
  1074. ; information will not be removed from realtime storage
  1075.  
  1076. ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
  1077. ; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
  1078. ; domains, each of which can direct the call to a specific context if desired.
  1079. ; By default, all domains are accepted and sent to the default context or the
  1080. ; context associated with the user/peer placing the call.
  1081. ; REGISTER to non-local domains will be automatically denied if a domain
  1082. ; list is configured.
  1083. ;
  1084. ; Domains can be specified using:
  1085. ; domain=<domain>[,<context>]
  1086. ; Examples:
  1087. ; domain=myasterisk.dom
  1088. ; domain=customer.com,customer-context
  1089. ;
  1090. ; In addition, all the 'default' domains associated with a server should be
  1091. ; added if incoming request filtering is desired.
  1092. ; autodomain=yes
  1093. ;
  1094. ; To disallow requests for domains not serviced by this server:
  1095. ; allowexternaldomains=no
  1096.  
  1097. ;domain=mydomain.tld,mydomain-incoming
  1098. ; Add domain and configure incoming context
  1099. ; for external calls to this domain
  1100. ;domain=1.2.3.4 ; Add IP address as local domain
  1101. ; You can have several "domain" settings
  1102. ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
  1103. ; Default is yes
  1104. ;autodomain=yes ; Turn this on to have Asterisk add local host
  1105. ; name and local IP to domain list.
  1106.  
  1107. ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
  1108. ; non-peers, use your primary domain "identity"
  1109. ; for From: headers instead of just your IP
  1110. ; address. This is to be polite and
  1111. ; it may be a mandatory requirement for some
  1112. ; destinations which do not have a prior
  1113. ; account relationship with your server.
  1114.  
  1115. ;------------------------------ Advice of Charge CONFIGURATION --------------------------
  1116. ; snom_aoc_enabled = yes; ; This options turns on and off support for sending AOC-D and
  1117. ; AOC-E to snom endpoints. This option can be used both in the
  1118. ; peer and global scope. The default for this option is off.
  1119.  
  1120.  
  1121. ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
  1122. ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
  1123. ; SIP channel. Defaults to "no". An enabled jitterbuffer will
  1124. ; be used only if the sending side can create and the receiving
  1125. ; side can not accept jitter. The SIP channel can accept jitter,
  1126. ; thus a jitterbuffer on the receive SIP side will be used only
  1127. ; if it is forced and enabled.
  1128.  
  1129. ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
  1130. ; channel. Defaults to "no".
  1131.  
  1132. ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
  1133.  
  1134. ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
  1135. ; resynchronized. Useful to improve the quality of the voice, with
  1136. ; big jumps in/broken timestamps, usually sent from exotic devices
  1137. ; and programs. Defaults to 1000.
  1138.  
  1139. ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
  1140. ; channel. Two implementations are currently available - "fixed"
  1141. ; (with size always equals to jbmaxsize) and "adaptive" (with
  1142. ; variable size, actually the new jb of IAX2). Defaults to fixed.
  1143.  
  1144. ; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
  1145. ; The option represents the number of milliseconds by which the new jitter buffer
  1146. ; will pad its size. the default is 40, so without modification, the new
  1147. ; jitter buffer will set its size to the jitter value plus 40 milliseconds.
  1148. ; increasing this value may help if your network normally has low jitter,
  1149. ; but occasionally has spikes.
  1150.  
  1151. ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
  1152.  
  1153. ;-----------------------------------------------------------------------------------
  1154.  
  1155. [authentication]
  1156. ; Global credentials for outbound calls, i.e. when a proxy challenges your
  1157. ; Asterisk server for authentication. These credentials override
  1158. ; any credentials in peer/register definition if realm is matched.
  1159. ;
  1160. ; This way, Asterisk can authenticate for outbound calls to other
  1161. ; realms. We match realm on the proxy challenge and pick an set of
  1162. ; credentials from this list
  1163. ; Syntax:
  1164. ; auth = <user>:<secret>@<realm>
  1165. ; auth = <user>#<md5secret>@<realm>
  1166. ; Example:
  1167. ;auth=mark:topsecret@digium.com
  1168. ;
  1169. ; You may also add auth= statements to [peer] definitions
  1170. ; Peer auth= override all other authentication settings if we match on realm
  1171.  
  1172. ;------------------------------------------------------------------------------
  1173. ; DEVICE CONFIGURATION
  1174. ;
  1175. ; SIP entities have a 'type' which determines their roles within Asterisk.
  1176. ; * For entities with 'type=peer':
  1177. ; Peers handle both inbound and outbound calls and are matched by ip/port, so for
  1178. ; The case of incoming calls from the peer, the IP address must match in order for
  1179. ; The invitation to work. This means calls made from either direction won't work if
  1180. ; The peer is unregistered while host=dynamic or if the host is otherise not set to
  1181. ; the correct IP of the sender.
  1182. ; * For entities with 'type=user':
  1183. ; Asterisk users handle inbound calls only (meaning they call Asterisk, Asterisk can't
  1184. ; call them) and are matched by their authorization information (authname and secret).
  1185. ; Asterisk doesn't rely on their IP and will accept calls regardless of the host setting
  1186. ; as long as the incoming SIP invite authorizes successfully.
  1187. ; * For entities with 'type=friend':
  1188. ; Asterisk will create the entity as both a friend and a peer. Asterisk will accept
  1189. ; calls from friends like it would for users, requiring only that the authorization
  1190. ; matches rather than the IP address. Since it is also a peer, a friend entity can
  1191. ; be called as long as its IP is known to Asterisk. In the case of host=dynamic,
  1192. ; this means it is necessary for the entity to register before Asterisk can call it.
  1193. ;
  1194. ; Use remotesecret for outbound authentication, and secret for authenticating
  1195. ; inbound requests. For historical reasons, if no remotesecret is supplied for an
  1196. ; outbound registration or call, the secret will be used.
  1197. ;
  1198. ; For device names, we recommend using only a-z, numerics (0-9) and underscore
  1199. ;
  1200. ; For local phones, type=friend works most of the time
  1201. ;
  1202. ; If you have one-way audio, you probably have NAT problems.
  1203. ; If Asterisk is on a public IP, and the phone is inside of a NAT device
  1204. ; you will need to configure nat option for those phones.
  1205. ; Also, turn on qualify=yes to keep the nat session open
  1206. ;
  1207. ; Configuration options available
  1208. ; --------------------
  1209. ; context
  1210. ; callingpres
  1211. ; permit
  1212. ; deny
  1213. ; secret
  1214. ; md5secret
  1215. ; remotesecret
  1216. ; transport
  1217. ; dtmfmode
  1218. ; directmedia
  1219. ; nat
  1220. ; callgroup
  1221. ; pickupgroup
  1222. ; language
  1223. ; allow
  1224. ; disallow
  1225. ; autoframing
  1226. ; insecure
  1227. ; trustrpid
  1228. ; trust_id_outbound
  1229. ; progressinband
  1230. ; promiscredir
  1231. ; useclientcode
  1232. ; accountcode
  1233. ; setvar
  1234. ; callerid
  1235. ; amaflags
  1236. ; callcounter
  1237. ; busylevel
  1238. ; allowoverlap
  1239. ; allowsubscribe
  1240. ; allowtransfer
  1241. ; ignoresdpversion
  1242. ; subscribecontext
  1243. ; template
  1244. ; videosupport
  1245. ; maxcallbitrate
  1246. ; rfc2833compensate
  1247. ; mailbox
  1248. ; session-timers
  1249. ; session-expires
  1250. ; session-minse
  1251. ; session-refresher
  1252. ; t38pt_usertpsource
  1253. ; regexten
  1254. ; fromdomain
  1255. ; fromuser
  1256. ; host
  1257. ; port
  1258. ; qualify
  1259. ; keepalive
  1260. ; defaultip
  1261. ; defaultuser
  1262. ; rtptimeout
  1263. ; rtpholdtimeout
  1264. ; sendrpid
  1265. ; outboundproxy
  1266. ; rfc2833compensate
  1267. ; callbackextension
  1268. ; timert1
  1269. ; timerb
  1270. ; qualifyfreq
  1271. ; t38pt_usertpsource
  1272. ; contactpermit ; Limit what a host may register as (a neat trick
  1273. ; contactdeny ; is to register at the same IP as a SIP provider,
  1274. ; contactacl ; then call oneself, and get redirected to that
  1275. ; ; same location).
  1276. ; directmediapermit
  1277. ; directmediadeny
  1278. ; directmediaacl
  1279. ; unsolicited_mailbox
  1280. ; use_q850_reason
  1281. ; maxforwards
  1282. ; encryption
  1283. ; description ; Used to provide a description of the peer in console output
  1284. ; dtlsenable
  1285. ; dtlsverify
  1286. ; dtlsrekey
  1287. ; dtlscertfile
  1288. ; dtlsprivatekey
  1289. ; dtlscipher
  1290. ; dtlscafile
  1291. ; dtlscapath
  1292. ; dtlssetup
  1293. ; dtlsfingerprint
  1294. ;
  1295.  
  1296. ;------------------------------------------------------------------------------
  1297. ; DTLS-SRTP CONFIGURATION
  1298. ;
  1299. ; DTLS-SRTP support is available if the underlying RTP engine in use supports it.
  1300. ;
  1301. ; dtlsenable = yes ; Enable or disable DTLS-SRTP support
  1302. ; dtlsverify = yes ; Verify that provided peer certificate and fingerprint are valid
  1303. ; ; A value of 'yes' will perform both certificate and fingerprint verification
  1304. ; ; A value of 'no' will perform no certificate or fingerprint verification
  1305. ; ; A value of 'fingerprint' will perform ONLY fingerprint verification
  1306. ; ; A value of 'certificate' will perform ONLY certficiate verification
  1307. ; dtlsrekey = 60 ; Interval at which to renegotiate the TLS session and rekey the SRTP session
  1308. ; ; If this is not set or the value provided is 0 rekeying will be disabled
  1309. ; dtlscertfile = file ; Path to certificate file to present
  1310. ; dtlsprivatekey = file ; Path to private key for certificate file
  1311. ; dtlscipher = <SSL cipher string> ; Cipher to use for TLS negotiation
  1312. ; ; A list of valid SSL cipher strings can be found at:
  1313. ; ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
  1314. ; dtlscafile = file ; Path to certificate authority certificate
  1315. ; dtlscapath = path ; Path to a directory containing certificate authority certificates
  1316. ; dtlssetup = actpass ; Whether we are willing to accept connections, connect to the other party, or both.
  1317. ; ; Valid options are active (we want to connect to the other party), passive (we want to
  1318. ; ; accept connections only), and actpass (we will do both). This value will be used in
  1319. ; ; the outgoing SDP when offering and for incoming SDP offers when the remote party sends
  1320. ; ; actpass
  1321. ; dtlsfingerprint = sha-1 ; The hash to use for the fingerprint in SDP (valid options are sha-1 and sha-256)
  1322.  
  1323. ;[sip_proxy]
  1324. ; For incoming calls only. Example: FWD (Free World Dialup)
  1325. ; We match on IP address of the proxy for incoming calls
  1326. ; since we can not match on username (caller id)
  1327. ;type=peer
  1328. ;context=from-fwd
  1329. ;host=fwd.pulver.com
  1330.  
  1331. ;[sip_proxy-out]
  1332. ;type=peer ; we only want to call out, not be called
  1333. ;remotesecret=guessit ; Our password to their service
  1334. ;defaultuser=yourusername ; Authentication user for outbound proxies
  1335. ;fromuser=yourusername ; Many SIP providers require this!
  1336. ;fromdomain=provider.sip.domain
  1337. ;host=box.provider.com
  1338. ;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will
  1339. ; ; accept both tcp and udp. The default transport type is only used for
  1340. ; ; outbound messages until a Registration takes place. During the
  1341. ; ; peer Registration the transport type may change to another supported
  1342. ; ; type if the peer requests so.
  1343.  
  1344. ;usereqphone=yes ; This provider requires ";user=phone" on URI
  1345. ;callcounter=yes ; Enable call counter
  1346. ;busylevel=2 ; Signal busy at 2 or more calls
  1347. ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
  1348. ;port=80 ; The port number we want to connect to on the remote side
  1349. ; Also used as "defaultport" in combination with "defaultip" settings
  1350.  
  1351. ;--- sample definition for a provider
  1352. ;[provider1]
  1353. ;type=peer
  1354. ;host=sip.provider1.com
  1355. ;fromuser=4015552299 ; how your provider knows you
  1356. ;remotesecret=youwillneverguessit ; The password we use to authenticate to them
  1357. ;secret=gissadetdu ; The password they use to contact us
  1358. ;callbackextension=123 ; Register with this server and require calls coming back to this extension
  1359. ;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
  1360. ; ; accept both tcp and udp. Default is udp. The first transport
  1361. ; ; listed will always be used for outgoing connections.
  1362. ;unsolicited_mailbox=4015552299 ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old
  1363. ; ; message count will be stored in the configured virtual mailbox. It can be used
  1364. ; ; by any device supporting MWI by specifying <configured value>@SIP_Remote as the
  1365. ; ; mailbox.
  1366.  
  1367. ;
  1368. ; Because you might have a large number of similar sections, it is generally
  1369. ; convenient to use templates for the common parameters, and add them
  1370. ; the the various sections. Examples are below, and we can even leave
  1371. ; the templates uncommented as they will not harm:
  1372.  
  1373. [basic-options](!) ; a template
  1374. dtmfmode=rfc2833
  1375. context=from-office
  1376. type=friend
  1377.  
  1378. [natted-phone](!,basic-options) ; another template inheriting basic-options
  1379. directmedia=no
  1380. host=dynamic
  1381.  
  1382. [public-phone](!,basic-options) ; another template inheriting basic-options
  1383. directmedia=yes
  1384.  
  1385. [my-codecs](!) ; a template for my preferred codecs
  1386. disallow=all
  1387. allow=ilbc
  1388. allow=g729
  1389. allow=gsm
  1390. allow=g723
  1391. allow=ulaw
  1392. ; Or, more simply:
  1393. ;allow=!all,ilbc,g729,gsm,g723,ulaw
  1394.  
  1395. [ulaw-phone](!) ; and another one for ulaw-only
  1396. disallow=all
  1397. allow=ulaw
  1398. ; Again, more simply:
  1399. ;allow=!all,ulaw
  1400.  
  1401. ; and finally instantiate a few phones
  1402. ;
  1403. ; [2133](natted-phone,my-codecs)
  1404. ; secret = peekaboo
  1405. ; [2134](natted-phone,ulaw-phone)
  1406. ; secret = not_very_secret
  1407. ; [2136](public-phone,ulaw-phone)
  1408. ; secret = not_very_secret_either
  1409. ; ...
  1410. ;
  1411.  
  1412. ; Standard configurations not using templates look like this:
  1413. ;
  1414. ;[grandstream1]
  1415. ;type=friend
  1416. ;context=from-sip ; Where to start in the dialplan when this phone calls
  1417. ;recordonfeature=dynamicfeature1 ; Feature to use when INFO with Record: on is received.
  1418. ;recordofffeature=dynamicfeature2 ; Feature to use when INFO with Record: off is received.
  1419. ;callerid=John Doe <1234> ; Full caller ID, to override the phones config
  1420. ; on incoming calls to Asterisk
  1421. ;description=Courtesy Phone ; Description of the peer. Shown when doing 'sip show peers'.
  1422. ;host=192.168.0.23 ; we have a static but private IP address
  1423. ; No registration allowed
  1424. ;directmedia=yes ; allow RTP voice traffic to bypass Asterisk
  1425. ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
  1426. ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
  1427. ; from the phone to asterisk (deprecated)
  1428. ; 1 for the explicit peer, 1 for the explicit user,
  1429. ; remember that a friend equals 1 peer and 1 user in
  1430. ; memory
  1431. ; There is no combined call counter for a "friend"
  1432. ; so there's currently no way in sip.conf to limit
  1433. ; to one inbound or outbound call per phone. Use
  1434. ; the group counters in the dial plan for that.
  1435. ;
  1436. ;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
  1437. ;disallow=all ; need to disallow=all before we can use allow=
  1438. ;allow=ulaw ; Note: In user sections the order of codecs
  1439. ; listed with allow= does NOT matter!
  1440. ;allow=alaw
  1441. ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
  1442. ;allow=g729 ; Pass-thru only unless g729 license obtained
  1443. ;callingpres=allowed_passed_screen ; Set caller ID presentation
  1444. ; See function CALLERPRES documentation for possible
  1445. ; values.
  1446.  
  1447. ;[xlite1]
  1448. ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
  1449. ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
  1450. ;type=friend
  1451. ;regexten=1234 ; When they register, create extension 1234
  1452. ;callerid="Jane Smith" <5678>
  1453. ;host=dynamic ; This device needs to register
  1454. ;directmedia=no ; Typically set to NO if behind NAT
  1455. ;disallow=all
  1456. ;allow=gsm ; GSM consumes far less bandwidth than ulaw
  1457. ;allow=ulaw
  1458. ;allow=alaw
  1459. ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
  1460. ;registertrying=yes ; Send a 100 Trying when the device registers.
  1461.  
  1462. ;[snom]
  1463. ;type=friend ; Friends place calls and receive calls
  1464. ;context=from-sip ; Context for incoming calls from this user
  1465. ;secret=blah
  1466. ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
  1467. ;language=de ; Use German prompts for this user
  1468. ;host=dynamic ; This peer register with us
  1469. ;dtmfmode=inband ; Choices are inband, rfc2833, or info
  1470. ;defaultip=192.168.0.59 ; IP used until peer registers
  1471. ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
  1472. ;subscribemwi=yes ; Only send notifications if this phone
  1473. ; subscribes for mailbox notification
  1474. ;vmexten=voicemail ; dialplan extension to reach mailbox
  1475. ; sets the Message-Account in the MWI notify message
  1476. ; defaults to global vmexten which defaults to "asterisk"
  1477. ;disallow=all
  1478. ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
  1479.  
  1480.  
  1481. ;[polycom]
  1482. ;type=friend ; Friends place calls and receive calls
  1483. ;context=from-sip ; Context for incoming calls from this user
  1484. ;secret=blahpoly
  1485. ;host=dynamic ; This peer register with us
  1486. ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
  1487. ;defaultuser=polly ; Username to use in INVITE until peer registers
  1488. ;defaultip=192.168.40.123
  1489. ; Normally you do NOT need to set this parameter
  1490. ;disallow=all
  1491. ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
  1492. ;progressinband=no ; Polycom phones don't work properly with "never"
  1493.  
  1494.  
  1495. ;[pingtel]
  1496. ;type=friend
  1497. ;secret=blah
  1498. ;host=dynamic
  1499. ;insecure=port ; Allow matching of peer by IP address without
  1500. ; matching port number
  1501. ;insecure=invite ; Do not require authentication of incoming INVITEs
  1502. ;insecure=port,invite ; (both)
  1503. ;qualify=1000 ; Consider it down if it's 1 second to reply
  1504. ; Helps with NAT session
  1505. ; qualify=yes uses default value
  1506. ;qualifyfreq=60 ; Qualification: How often to check for the
  1507. ; host to be up in seconds
  1508. ; Set to low value if you use low timeout for
  1509. ; NAT of UDP sessions
  1510. ;
  1511. ; Call group and Pickup group should be in the range from 0 to 63
  1512. ;
  1513. ;callgroup=1,3-4 ; We are in caller groups 1,3,4
  1514. ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
  1515. ;namedcallgroup=engineering,sales,netgroup,protgroup ; We are in named call groups engineering,sales,netgroup,protgroup
  1516. ;namedpickupgroup=sales ; We can do call pick-p for named call group sales
  1517. ;defaultip=192.168.0.60 ; IP address to use if peer has not registered
  1518. ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
  1519. ;permit=192.168.0.60/255.255.255.0
  1520. ;permit=192.168.0.60/24 ; we can also use CIDR notation for subnet masks
  1521. ;permit=2001:db8::/32 ; IPv6 ACLs can be specified if desired. IPv6 ACLs
  1522. ; apply only to IPv6 addresses, and IPv4 ACLs apply
  1523. ; only to IPv4 addresses.
  1524. ;acl=named_acl_example ; Use named ACLs defined in acl.conf
  1525.  
  1526. ;[cisco1]
  1527. ;type=friend
  1528. ;secret=blah
  1529. ;qualify=200 ; Qualify peer is no more than 200ms away
  1530. ;host=dynamic ; This device registers with us
  1531. ;directmedia=no ; Asterisk by default tries to redirect the
  1532. ; RTP media stream (audio) to go directly from
  1533. ; the caller to the callee. Some devices do not
  1534. ; support this (especially if one of them is
  1535. ; behind a NAT).
  1536. ;defaultip=192.168.0.4 ; IP address to use until registration
  1537. ;defaultuser=goran ; Username to use when calling this device before registration
  1538. ; Normally you do NOT need to set this parameter
  1539. ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from or to this device
  1540. ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
  1541. ; cause the given audio file to
  1542. ; be played upon completion of
  1543. ; an attended transfer.
  1544.  
  1545. ;[pre14-asterisk]
  1546. ;type=friend
  1547. ;secret=digium
  1548. ;host=dynamic
  1549. ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
  1550. ; You must have this turned on or DTMF reception will work improperly.
  1551. ;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
  1552. ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
  1553. ; external IP address of the remote device. If port forwarding is done at the client side
  1554. ; then UDPTL will flow to the remote device.
  1555.  
  1556. [201]
  1557. secret=azerty
  1558. callerid="Jacky" <201>
  1559. context=local
  1560. mailbox=201@default
  1561. type=friend
  1562. host=dynamic
  1563. nat=yes
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