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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. ; And to alter the To: or the From: header, you can additionally append
  28. ; the following to any of the above strings:
  29. ; [![touser[@todomain]][![fromuser][@fromdomain]]]
  30. ;
  31. ;
  32. ; Devicename
  33. ; devicename is defined as a peer in a section below.
  34. ;
  35. ; username@domain
  36. ; Call any SIP user on the Internet
  37. ; (Don't forget to enable DNS SRV records if you want to use this)
  38. ;
  39. ; devicename/extension
  40. ; If you define a SIP proxy as a peer below, you may call
  41. ; SIP/proxyhostname/user or SIP/user@proxyhostname
  42. ; where the proxyhostname is defined in a section below
  43. ; This syntax also works with ATA's with FXO ports
  44. ;
  45. ; SIP/username[:password[:md5secret[:authname]]]@host[:port]
  46. ; This form allows you to specify password or md5secret and authname
  47. ; without altering any authentication data in config.
  48. ; Examples:
  49. ;
  50. ; SIP/*98@mysipproxy
  51. ; SIP/sales:topsecret::account02@domain.com:5062
  52. ; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
  53. ;
  54. ; IPorHost
  55. ; The next server for this call regardless of domain/peer
  56. ;
  57. ; All of these dial strings specify the SIP request URI.
  58. ; In addition, you can specify a specific To: header by adding an
  59. ; exclamation mark after the dial string, like
  60. ;
  61. ; SIP/sales@mysipproxy!sales@edvina.net
  62. ;
  63. ; (Specifying only @todomain without touser will create an invalid SIP
  64. ; request.)
  65. ;
  66. ; Similarly, you can specify the From header as well, after a second
  67. ; exclamation mark:
  68. ;
  69. ; SIP/customer@mysipproxy!!customersupport@wearespindle.com
  70. ;
  71. ; A new feature for 1.8 allows one to specify a host or IP address to use
  72. ; when routing the call. This is typically used in tandem with func_srv if
  73. ; multiple methods of reaching the same domain exist. The host or IP address
  74. ; is specified after the third slash in the dialstring. Examples:
  75. ;
  76. ; SIP/devicename/extension/IPorHost
  77. ; SIP/username@domain//IPorHost
  78. ;
  79. ; CLI Commands
  80. ; -------------------------------------------------------------
  81. ; Useful CLI commands to check peers/users:
  82. ; sip show peers Show all SIP peers (including friends)
  83. ; sip show registry Show status of hosts we register with
  84. ;
  85. ; sip set debug on Show all SIP messages
  86. ;
  87. ; sip reload Reload configuration file
  88. ; sip show settings Show the current channel configuration
  89. ;
  90. ; ------ Naming devices ------------------------------------------------------
  91. ;
  92. ; When naming devices, make sure you understand how Asterisk matches calls
  93. ; that come in.
  94. ; 1. Asterisk checks the SIP From: address username and matches against
  95. ; names of devices with type=user
  96. ; The name is the text between square brackets [name]
  97. ; 2. Asterisk checks the From: addres and matches the list of devices
  98. ; with a type=peer
  99. ; 3. Asterisk checks the IP address (and port number) that the INVITE
  100. ; was sent from and matches against any devices with type=peer
  101. ;
  102. ; Don't mix extensions with the names of the devices. Devices need a unique
  103. ; name. The device name is *not* used as phone numbers. Phone numbers are
  104. ; anything you declare as an extension in the dialplan (extensions.conf).
  105. ;
  106. ; When setting up trunks, make sure there's no risk that any From: username
  107. ; (caller ID) will match any of your device names, because then Asterisk
  108. ; might match the wrong device.
  109. ;
  110. ; Note: The parameter "username" is not the username and in most cases is
  111. ; not needed at all. Check below. In later releases, it's renamed
  112. ; to "defaultuser" which is a better name, since it is used in
  113. ; combination with the "defaultip" setting.
  114. ; ----------------------------------------------------------------------------
  115.  
  116. ; ** Old configuration options **
  117. ; The "call-limit" configuation option is considered old is replaced
  118. ; by new functionality. To enable callcounters, you use the new
  119. ; "callcounter" setting (for extension states in queue and subscriptions)
  120. ; You are encouraged to use the dialplan groupcount functionality
  121. ; to enforce call limits instead of using this channel-specific method.
  122. ; You can still set limits per device in sip.conf or in a database by using
  123. ; "setvar" to set variables that can be used in the dialplan for various limits.
  124.  
  125. [general]
  126. context=public ; Default context for incoming calls. Defaults to 'default'
  127. ;allowguest=no ; Allow or reject guest calls (default is yes)
  128. ; If your Asterisk is connected to the Internet
  129. ; and you have allowguest=yes
  130. ; you want to check which services you offer everyone
  131. ; out there, by enabling them in the default context (see below).
  132. ;match_auth_username=yes ; if available, match user entry using the
  133. ; 'username' field from the authentication line
  134. ; instead of the From: field.
  135. allowoverlap=no ; Disable overlap dialing support. (Default is yes)
  136. ;allowoverlap=yes ; Enable RFC3578 overlap dialing support.
  137. ; Can use the Incomplete application to collect the
  138. ; needed digits from an ambiguous dialplan match.
  139. ;allowoverlap=dtmf ; Enable overlap dialing support using DTMF delivery
  140. ; methods (inband, RFC2833, SIP INFO) in the early
  141. ; media phase. Uses the Incomplete application to
  142. ; collect the needed digits.
  143. ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
  144. ; Default is enabled. The Dial() options 't' and 'T' are not
  145. ; related as to whether SIP transfers are allowed or not.
  146. ;realm=mydomain.tld ; Realm for digest authentication
  147. ; defaults to "asterisk". If you set a system name in
  148. ; asterisk.conf, it defaults to that system name
  149. ; Realms MUST be globally unique according to RFC 3261
  150. ; Set this to your host name or domain name
  151. ;domainsasrealm=no ; Use domains list as realms
  152. ; You can serve multiple Realms specifying several
  153. ; 'domain=...' directives (see below).
  154. ; In this case Realm will be based on request 'From'/'To' header
  155. ; and should match one of domain names.
  156. ; Otherwise default 'realm=...' will be used.
  157. ;recordonfeature=automixmon ; Default feature to use when receiving 'Record: on' header
  158. ; from an INFO message. Defaults to 'automon'. Works with
  159. ; dynamic features. Feature must be usable on requesting
  160. ; channel for it to work. Setting this value to a blank
  161. ; will disable it.
  162. ;recordofffeature=automixmon ; Default feature to use when receiving 'Record: off' header
  163. ; from an INFO message. Defaults to 'automon'. Works with
  164. ; dynamic features. Feature must be usable on requesting
  165. ; channel for it to work. Setting this value to a blank
  166. ; will disable it.
  167.  
  168. ; With the current situation, you can do one of four things:
  169. ; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1
  170. ; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1
  171. ; c) Listen on the IPv4 wildcard. Example: bindaddr=0.0.0.0
  172. ; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=::
  173. ; (You can choose independently for UDP, TCP, and TLS, by specifying different values for
  174. ; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".)
  175. ; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat.
  176. ; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)
  177. ;
  178. ; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061
  179. ; for TLS).
  180. ; IPv4 example: bindaddr=0.0.0.0:5062
  181. ; IPv6 example: bindaddr=[::]:5062
  182. ;
  183. ; The address family of the bound UDP address is used to determine how Asterisk performs
  184. ; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only
  185. ; AAAA records are considered. In case d), both A and AAAA records are considered. Note,
  186. ; however, that Asterisk ignores all records except the first one. In case d), when both A
  187. ; and AAAA records are available, either an A or AAAA record will be first, and which one
  188. ; depends on the operating system. On systems using glibc, AAAA records are given
  189. ; priority.
  190.  
  191. udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
  192. ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
  193.  
  194. ;rtpbindaddr=172.16.42.1 ; IP address to bind RTP listen sock to (default is disabled). When
  195. ; disabled the udpbindaddr is used.
  196.  
  197. ; When a dialog is started with another SIP endpoint, the other endpoint
  198. ; should include an Allow header telling us what SIP methods the endpoint
  199. ; implements. However, some endpoints either do not include an Allow header
  200. ; or lie about what methods they implement. In the former case, Asterisk
  201. ; makes the assumption that the endpoint supports all known SIP methods.
  202. ; If you know that your SIP endpoint does not provide support for a specific
  203. ; method, then you may provide a comma-separated list of methods that your
  204. ; endpoint does not implement in the disallowed_methods option. Note that
  205. ; if your endpoint is truthful with its Allow header, then there is no need
  206. ; to set this option. This option may be set in the general section or may
  207. ; be set per endpoint. If this option is set both in the general section and
  208. ; in a peer section, then the peer setting completely overrides the general
  209. ; setting (i.e. the result is *not* the union of the two options).
  210. ;
  211. ; Note also that while Asterisk currently will parse an Allow header to learn
  212. ; what methods an endpoint supports, the only actual use for this currently
  213. ; is for determining if Asterisk may send connected line UPDATE requests and
  214. ; MESSAGE requests. Its use may be expanded in the future.
  215. ;
  216. ; disallowed_methods = UPDATE
  217.  
  218. ;
  219. ; Note that the TCP and TLS support for chan_sip is currently considered
  220. ; experimental. Since it is new, all of the related configuration options are
  221. ; subject to change in any release. If they are changed, the changes will
  222. ; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
  223. ;
  224. tcpenable=no ; Enable server for incoming TCP connections (default is no)
  225. tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
  226. ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
  227.  
  228. ;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
  229. ;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
  230. ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
  231. ; Remember that the IP address must match the common name (hostname) in the
  232. ; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
  233. ; For details how to construct a certificate for SIP see
  234. ; http://tools.ietf.org/html/draft-ietf-sip-domain-certs
  235.  
  236. ;tcpauthtimeout = 30 ; tcpauthtimeout specifies the maximum number
  237. ; of seconds a client has to authenticate. If
  238. ; the client does not authenticate beofre this
  239. ; timeout expires, the client will be
  240. ; disconnected. (default: 30 seconds)
  241.  
  242. ;tcpauthlimit = 100 ; tcpauthlimit specifies the maximum number of
  243. ; unauthenticated sessions that will be allowed
  244. ; to connect at any given time. (default: 100)
  245.  
  246. ;websocket_enabled = true ; Set to false to prevent chan_sip from listening to websockets. This
  247. ; is neeeded when using chan_sip and res_pjsip_transport_websockets on
  248. ; the same system.
  249.  
  250. ;websocket_write_timeout = 100 ; Default write timeout to set on websocket transports.
  251. ; This value may need to be adjusted for connections where
  252. ; Asterisk must write a substantial amount of data and the
  253. ; receiving clients are slow to process the received information.
  254. ; Value is in milliseconds; default is 100 ms.
  255.  
  256. transport=udp ; Set the default transports. The order determines the primary default transport.
  257. ; If tcpenable=no and the transport set is tcp, we will fallback to UDP.
  258.  
  259. srvlookup=yes ; Enable DNS SRV lookups on outbound calls
  260. ; Note: Asterisk only uses the first host
  261. ; in SRV records
  262. ; Disabling DNS SRV lookups disables the
  263. ; ability to place SIP calls based on domain
  264. ; names to some other SIP users on the Internet
  265. ; Specifying a port in a SIP peer definition or
  266. ; when dialing outbound calls will supress SRV
  267. ; lookups for that peer or call.
  268.  
  269. ;pedantic=yes ; Enable checking of tags in headers,
  270. ; international character conversions in URIs
  271. ; and multiline formatted headers for strict
  272. ; SIP compatibility (defaults to "yes")
  273.  
  274. ; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
  275. ;tos_sip=cs3 ; Sets TOS for SIP packets.
  276. ;tos_audio=ef ; Sets TOS for RTP audio packets.
  277. ;tos_video=af41 ; Sets TOS for RTP video packets.
  278. ;tos_text=af41 ; Sets TOS for RTP text packets.
  279.  
  280. ;cos_sip=3 ; Sets 802.1p priority for SIP packets.
  281. ;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
  282. ;cos_video=4 ; Sets 802.1p priority for RTP video packets.
  283. ;cos_text=3 ; Sets 802.1p priority for RTP text packets.
  284.  
  285. ;maxexpiry=3600 ; Maximum allowed time of incoming registrations (seconds)
  286. ;minexpiry=60 ; Minimum length of registrations (default 60)
  287. ;defaultexpiry=120 ; Default length of incoming/outgoing registration
  288. ;submaxexpiry=3600 ; Maximum allowed time of incoming subscriptions (seconds), default: maxexpiry
  289. ;subminexpiry=60 ; Minimum length of subscriptions, default: minexpiry
  290. ;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
  291. ;maxforwards=70 ; Setting for the SIP Max-Forwards: header (loop prevention)
  292. ; Default value is 70
  293. ;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds
  294. ; and reported in milliseconds with sip show settings.
  295. ; Set to low value if you use low timeout for NAT of UDP sessions
  296. ; Default: 60
  297. ;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
  298. ; Default: 100
  299. ;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
  300. ; Default: 1
  301. ;keepalive=60 ; Interval at which keepalive packets should be sent to a peer
  302. ; Valid options are yes (60 seconds), no, or the number of seconds.
  303. ; Default: 0
  304. ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
  305. ;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
  306. ; fully. Enable this option to not get error messages
  307. ; when sending MWI to phones with this bug.
  308. ;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in
  309. ; the From: header as the "name" portion. Also fill the
  310. ; "user" portion of the URI in the From: header with this
  311. ; value if no fromuser is set
  312. ; Default: empty
  313. ;vmexten=voicemail ; dialplan extension to reach mailbox sets the
  314. ; Message-Account in the MWI notify message
  315. ; defaults to "asterisk"
  316.  
  317. ; Codec negotiation
  318. ;
  319. ; When Asterisk is receiving a call, the codec will initially be set to the
  320. ; first codec in the allowed codecs defined for the user receiving the call
  321. ; that the caller also indicates that it supports. But, after the caller
  322. ; starts sending RTP, Asterisk will switch to using whatever codec the caller
  323. ; is sending.
  324. ;
  325. ; When Asterisk is placing a call, the codec used will be the first codec in
  326. ; the allowed codecs that the callee indicates that it supports. Asterisk will
  327. ; *not* switch to whatever codec the callee is sending.
  328. ;
  329. ;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
  330. ; rather than advertising all joint codec capabilities. This
  331. ; limits the other side's codec choice to exactly what we prefer.
  332.  
  333. ;disallow=all ; First disallow all codecs
  334. ;allow=ulaw ; Allow codecs in order of preference
  335. ;allow=ilbc ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
  336. ; for framing options
  337. ;autoframing=yes ; Set packetization based on the remote endpoint's (ptime)
  338. ; preferences. Defaults to no.
  339. ;
  340. ; This option specifies a preference for which music on hold class this channel
  341. ; should listen to when put on hold if the music class has not been set on the
  342. ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
  343. ; channel putting this one on hold did not suggest a music class.
  344. ;
  345. ; This option may be specified globally, or on a per-user or per-peer basis.
  346. ;
  347. ;mohinterpret=default
  348. ;
  349. ; This option specifies which music on hold class to suggest to the peer channel
  350. ; when this channel places the peer on hold. It may be specified globally or on
  351. ; a per-user or per-peer basis.
  352. ;
  353. ;mohsuggest=default
  354. ;
  355. ;parkinglot=plaza ; Sets the default parking lot for call parking
  356. ; This may also be set for individual users/peers
  357. ; Parkinglots are configured in features.conf
  358. ;language=en ; Default language setting for all users/peers
  359. ; This may also be set for individual users/peers
  360. ;tonezone=se ; Default tonezone for all users/peers
  361. ; This may also be set for individual users/peers
  362.  
  363. ;relaxdtmf=yes ; Relax dtmf handling
  364. ;trustrpid = no ; If Remote-Party-ID should be trusted
  365. ;sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no)
  366. ;sendrpid = rpid ; Use the "Remote-Party-ID" header
  367. ; to send the identity of the remote party
  368. ; This is identical to sendrpid=yes
  369. ;sendrpid = pai ; Use the "P-Asserted-Identity" header
  370. ; to send the identity of the remote party
  371. ;rpid_update = no ; In certain cases, the only method by which a connected line
  372. ; change may be immediately transmitted is with a SIP UPDATE request.
  373. ; If communicating with another Asterisk server, and you wish to be able
  374. ; transmit such UPDATE messages to it, then you must enable this option.
  375. ; Otherwise, we will have to wait until we can send a reinvite to
  376. ; transmit the information.
  377. ;trust_id_outbound = no ; Controls whether or not we trust this peer with private identity
  378. ; information (when the remote party has callingpres=prohib or equivalent).
  379. ; no - RPID/PAI headers will not be included for private peer information
  380. ; yes - RPID/PAI headers will include the private peer information. Privacy
  381. ; requirements will be indicated in a Privacy header for sendrpid=pai
  382. ; legacy - RPID/PAI will be included for private peer information. In the
  383. ; case of sendrpid=pai, private data that would be included in them
  384. ; will be anonymized. For sendrpid=rpid, private data may be included
  385. ; but the remote party's domain will be anonymized. The way legacy
  386. ; behaves may violate RFC-3325, but it follows historic behavior.
  387. ; This option is set to 'legacy' by default
  388. ;prematuremedia=no ; Some ISDN links send empty media frames before
  389. ; the call is in ringing or progress state. The SIP
  390. ; channel will then send 183 indicating early media
  391. ; which will be empty - thus users get no ring signal.
  392. ; Setting this to "yes" will stop any media before we have
  393. ; call progress (meaning the SIP channel will not send 183 Session
  394. ; Progress for early media). Default is "yes". Also make sure that
  395. ; the SIP peer is configured with progressinband=never.
  396. ;
  397. ; In order for "noanswer" applications to work, you need to run
  398. ; the progress() application in the priority before the app.
  399.  
  400. ;progressinband=no ; If we should generate in-band ringing. Always
  401. ; use 'never' to never use in-band signalling, even in cases
  402. ; where some buggy devices might not render it
  403. ; Valid values: yes, no, never Default: no
  404. ;useragent=Asterisk PBX ; Allows you to change the user agent string
  405. ; The default user agent string also contains the Asterisk
  406. ; version. If you don't want to expose this, change the
  407. ; useragent string.
  408. ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
  409. ; Note that promiscredir when redirects are made to the
  410. ; local system will cause loops since Asterisk is incapable
  411. ; of performing a "hairpin" call.
  412. ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
  413. ; a valid phone number
  414. ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
  415. ; Other options:
  416. ; info : SIP INFO messages (application/dtmf-relay)
  417. ; shortinfo : SIP INFO messages (application/dtmf)
  418. ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
  419. ; auto : Use rfc2833 if offered, inband otherwise
  420.  
  421. ;compactheaders = yes ; send compact sip headers.
  422. ;
  423. ;videosupport=yes ; Turn on support for SIP video. You need to turn this
  424. ; on in this section to get any video support at all.
  425. ; You can turn it off on a per peer basis if the general
  426. ; video support is enabled, but you can't enable it for
  427. ; one peer only without enabling in the general section.
  428. ; If you set videosupport to "always", then RTP ports will
  429. ; always be set up for video, even on clients that don't
  430. ; support it. This assists callfile-derived calls and
  431. ; certain transferred calls to use always use video when
  432. ; available. [yes|NO|always]
  433.  
  434. ;textsupport=no ; Support for ITU-T T.140 realtime text.
  435. ; The default value is "no".
  436.  
  437. ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
  438. ; Videosupport and maxcallbitrate is settable
  439. ; for peers and users as well
  440. ;authfailureevents=no ; generate manager "peerstatus" events when peer can't
  441. ; authenticate with Asterisk. Peerstatus will be "rejected".
  442. ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
  443. ; for any reason, always reject with an identical response
  444. ; equivalent to valid username and invalid password/hash
  445. ; instead of letting the requester know whether there was
  446. ; a matching user or peer for their request. This reduces
  447. ; the ability of an attacker to scan for valid SIP usernames.
  448. ; This option is set to "yes" by default.
  449.  
  450. ;auth_options_requests = yes ; Enabling this option will authenticate OPTIONS requests just like
  451. ; INVITE requests are. By default this option is disabled.
  452.  
  453. ;accept_outofcall_message = no ; Disable this option to reject all MESSAGE requests outside of a
  454. ; call. By default, this option is enabled. When enabled, MESSAGE
  455. ; requests are passed in to the dialplan.
  456.  
  457. ;outofcall_message_context = messages ; Context all out of dialog msgs are sent to. When this
  458. ; option is not set, the context used during peer matching
  459. ; is used. This option can be defined at both the peer and
  460. ; global level.
  461.  
  462. ;auth_message_requests = yes ; Enabling this option will authenticate MESSAGE requests.
  463. ; By default this option is enabled. However, it can be disabled
  464. ; should an application desire to not load the Asterisk server with
  465. ; doing authentication and implement end to end security in the
  466. ; message body.
  467.  
  468. ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
  469. ; order instead of RFC3551 packing order (this is required
  470. ; for Sipura and Grandstream ATAs, among others). This is
  471. ; contrary to the RFC3551 specification, the peer _should_
  472. ; be negotiating AAL2-G726-32 instead :-(
  473. ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
  474. ;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
  475. ;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
  476. ;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls
  477. ;outboundproxy=192.0.2.1 ; IPv4 address literal (default port is 5060)
  478. ;outboundproxy=2001:db8::1 ; IPv6 address literal (default port is 5060)
  479. ;outboundproxy=192.168.0.2.1:5062 ; IPv4 address literal with explicit port
  480. ;outboundproxy=[2001:db8::1]:5062 ; IPv6 address literal with explicit port
  481. ; ; (could also be tcp,udp) - defining transports on the proxy line only
  482. ; ; applies for the global proxy, otherwise use the transport= option
  483.  
  484. ;supportpath=yes ; This activates parsing and handling of Path header as defined in RFC 3327. This enables
  485. ; Asterisk to route outgoing out-of-dialog requests via a set of proxies by using a pre-loaded
  486. ; route-set defined by the Path headers in the REGISTER request.
  487. ; NOTE: There are multiple things to consider with this setting:
  488. ; * As this influences routing of SIP requests make sure to not trust Path headers provided
  489. ; by the user's SIP client (the proxy in front of Asterisk should remove existing user
  490. ; provided Path headers).
  491. ; * When a peer has both a path and outboundproxy set, the path will be added to Route: header
  492. ; but routing to next hop is done using the outboundproxy.
  493. ; * If set globally, not only will all peers use the Path header, but outbound REGISTER
  494. ; requests from Asterisk will add path to the Supported header.
  495.  
  496. ;rtsavepath=yes ; If using dynamic realtime, store the path headers
  497.  
  498. ;matchexternaddrlocally = yes ; Only substitute the externaddr or externhost setting if it matches
  499. ; your localnet setting. Unless you have some sort of strange network
  500. ; setup you will not need to enable this.
  501.  
  502. ;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering
  503. ; as any IP address used for staticly defined
  504. ; hosts. This helps avoid the configuration
  505. ; error of allowing your users to register at
  506. ; the same address as a SIP provider.
  507.  
  508. ;contactdeny=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to
  509. ;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may
  510. ; register their phones.
  511. ;contactacl=named_acl_example ; Use named ACLs defined in acl.conf
  512.  
  513. ;rtp_engine=asterisk ; RTP engine to use when communicating with the device
  514.  
  515. ;
  516. ; If regcontext is specified, Asterisk will dynamically create and destroy a
  517. ; NoOp priority 1 extension for a given peer who registers or unregisters with
  518. ; us and have a "regexten=" configuration item.
  519. ; Multiple contexts may be specified by separating them with '&'. The
  520. ; actual extension is the 'regexten' parameter of the registering peer or its
  521. ; name if 'regexten' is not provided. If more than one context is provided,
  522. ; the context must be specified within regexten by appending the desired
  523. ; context after '@'. More than one regexten may be supplied if they are
  524. ; separated by '&'. Patterns may be used in regexten.
  525. ;
  526. ;regcontext=sipregistrations
  527. ;regextenonqualify=yes ; Default "no"
  528. ; If you have qualify on and the peer becomes unreachable
  529. ; this setting will enforce inactivation of the regexten
  530. ; extension for the peer
  531. ;legacy_useroption_parsing=yes ; Default "no" ; If you have this option enabled and there are semicolons
  532. ; in the user field of a sip URI, the field be truncated
  533. ; at the first semicolon seen. This effectively makes
  534. ; semicolon a non-usable character for peer names, extensions,
  535. ; and maybe other, less tested things. This can be useful
  536. ; for improving compatability with devices that like to use
  537. ; user options for whatever reason. The behavior is similar to
  538. ; how SIP URI's were typically handled in 1.6.2, hence the name.
  539.  
  540. ;send_diversion=no ; Default "yes" ; Asterisk normally sends Diversion headers with certain SIP
  541. ; invites to relay data about forwarded calls. If this option
  542. ; is disabled, Asterisk won't send Diversion headers unless
  543. ; they are added manually.
  544.  
  545. ; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
  546. ; in square brackets. For example, the caller id value 555.5555 becomes 5555555
  547. ; when this option is enabled. Disabling this option results in no modification
  548. ; of the caller id value, which is necessary when the caller id represents something
  549. ; that must be preserved. This option can only be used in the [general] section.
  550. ; By default this option is on.
  551. ;
  552. ;shrinkcallerid=yes ; on by default
  553.  
  554.  
  555. ;use_q850_reason = no ; Default "no"
  556. ; Set to yes add Reason header and use Reason header if it is available.
  557.  
  558. ; When the Transfer() application sends a REFER SIP message, extra headers specified in
  559. ; the dialplan by way of SIPAddHeader are sent out with that message. 1.8 and earlier did not
  560. ; add the extra headers. To revert to 1.8- behavior, call SIPRemoveHeader with no arguments
  561. ; before calling Transfer() to remove all additional headers from the channel. The setting
  562. ; below is for transitional compatibility only.
  563. ;
  564. ;refer_addheaders=yes ; on by default
  565.  
  566. ;autocreatepeer=no ; Allow any UAC not explicitly defined to register
  567. ; WITHOUT AUTHENTICATION. Enabling this options poses a high
  568. ; potential security risk and should be avoided unless the
  569. ; server is behind a trusted firewall.
  570. ; If set to "yes", then peers created in this fashion
  571. ; are purged during SIP reloads.
  572. ; When set to "persist", the peers created in this fashion
  573. ; are not purged during SIP reloads.
  574.  
  575. ;
  576. ; ----------------------- TLS settings ------------------------------------------------------------
  577. ;tlscertfile=</path/to/certificate.pem> ; Certificate chain (*.pem format only) to use for TLS connections
  578. ; The certificates must be sorted starting with the subject's certificate
  579. ; and followed by intermediate CA certificates if applicable. If the
  580. ; file name ends in _rsa, for example "asterisk_rsa.pem", the files
  581. ; "asterisk_dsa.pem" and/or "asterisk_ecc.pem" are loaded
  582. ; (certificate, intermediates, private key), to support multiple
  583. ; algorithms for server authentication (RSA, DSA, ECDSA). If the chains
  584. ; are different, at least OpenSSL 1.0.2 is required.
  585. ; Default is to look for "asterisk.pem" in current directory
  586.  
  587. ;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem format only) for TLS connections.
  588. ; If no tlsprivatekey is specified, tlscertfile is searched for
  589. ; for both public and private key.
  590.  
  591. ;tlscafile=</path/to/certificate>
  592. ; If the server your connecting to uses a self signed certificate
  593. ; you should have their certificate installed here so the code can
  594. ; verify the authenticity of their certificate.
  595.  
  596. ;tlscapath=</path/to/ca/dir>
  597. ; A directory full of CA certificates. The files must be named with
  598. ; the CA subject name hash value.
  599. ; (see man SSL_CTX_load_verify_locations for more info)
  600.  
  601. ;tlsdontverifyserver=[yes|no]
  602. ; If set to yes, don't verify the servers certificate when acting as
  603. ; a client. If you don't have the server's CA certificate you can
  604. ; set this and it will connect without requiring tlscafile to be set.
  605. ; Default is no.
  606.  
  607. ;tlscipher=<SSL cipher string>
  608. ; A string specifying which SSL ciphers to use or not use
  609. ; A list of valid SSL cipher strings can be found at:
  610. ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
  611. ;
  612. ;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2.
  613. ; Specify protocol for outbound client connections.
  614. ; If left unspecified, the default is the general-
  615. ; purpose version-flexible SSL/TLS method (sslv23).
  616. ; With that, the actual protocol version used will
  617. ; be negotiated to the highest version mutually
  618. ; supported by Asterisk and the remote server, i.e.
  619. ; TLSv1.2. The supported protocols are listed at
  620. ; http://www.openssl.org/docs/ssl/SSL_CTX_new.html
  621. ; SSLv2 and SSLv3 are disabled within Asterisk.
  622. ; Your distribution might have changed that list
  623. ; further.
  624. ;
  625. ; -------------------------- SIP timers ----------------------------------------------------
  626. ; These timers are used primarily in INVITE transactions.
  627. ; The default for Timer T1 is 500 ms or the measured run-trip time between
  628. ; Asterisk and the device if you have qualify=yes for the device.
  629. ;
  630. ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
  631. ; Defaults to 100 ms
  632. ;timert1=500 ; Default T1 timer
  633. ; Defaults to 500 ms or the measured round-trip
  634. ; time to a peer (qualify=yes).
  635. ;timerb=32000 ; Call setup timer. If a provisional response is not received
  636. ; in this amount of time, the call will autocongest
  637. ; Defaults to 64*timert1
  638.  
  639. ; -------------------------- RTP timers ----------------------------------------------------
  640. ; These timers are currently used for both audio and video streams. The RTP timeouts
  641. ; are only applied to the audio channel.
  642. ; The settings are settable in the global section as well as per device
  643. ;
  644. ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
  645. ; on the audio channel
  646. ; when we're not on hold. This is to be able to hangup
  647. ; a call in the case of a phone disappearing from the net,
  648. ; like a powerloss or grandma tripping over a cable.
  649. ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
  650. ; on the audio channel
  651. ; when we're on hold (must be > rtptimeout)
  652. ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
  653. ; (default is off - zero)
  654.  
  655. ; -------------------------- SIP Session-Timers (RFC 4028)------------------------------------
  656. ; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
  657. ; This mechanism can detect and reclaim SIP channels that do not terminate through normal
  658. ; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
  659. ; The operation of Session-Timers is driven by the following configuration parameters:
  660. ;
  661. ; * session-timers - Session-Timers feature operates in the following three modes:
  662. ; originate : Request and run session-timers always
  663. ; accept : Run session-timers only when requested by other UA
  664. ; refuse : Do not run session timers in any case
  665. ; The default mode of operation is 'accept'.
  666. ; * session-expires - Maximum session refresh interval in seconds. Defaults to 1800 secs.
  667. ; * session-minse - Minimum session refresh interval in seconds. Defualts to 90 secs.
  668. ; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
  669. ; uac - Default to the caller initially refreshing when possible
  670. ; uas - Default to the callee initially refreshing when possible
  671. ;
  672. ; Note that, due to recommendations in RFC 4028, Asterisk will always honor the other
  673. ; endpoint's preference for who will handle refreshes. Asterisk will never override the
  674. ; preferences of the other endpoint. Doing so could result in Asterisk and the endpoint
  675. ; fighting over who sends the refreshes. This holds true for the initiation of session
  676. ; timers and subsequent re-INVITE requests whether Asterisk is the caller or callee, or
  677. ; whether Asterisk is currently the refresher or not.
  678. ;
  679. ;session-timers=originate
  680. ;session-expires=600
  681. ;session-minse=90
  682. ;session-refresher=uac
  683. ;
  684. ; -------------------------- SIP DEBUGGING ---------------------------------------------------
  685. ;sipdebug = yes ; Turn on SIP debugging by default, from
  686. ; the moment the channel loads this configuration.
  687. ; NOTE: You cannot use the CLI to turn it off. You'll
  688. ; need to edit this and reload the config.
  689. ;recordhistory=yes ; Record SIP history by default
  690. ; (see sip history / sip no history)
  691. ;dumphistory=yes ; Dump SIP history at end of SIP dialogue
  692. ; SIP history is output to the DEBUG logging channel
  693.  
  694.  
  695. ; -------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
  696. ; You can subscribe to the status of extensions with a "hint" priority
  697. ; (See extensions.conf.sample for examples)
  698. ; chan_sip support two major formats for notifications: dialog-info and SIMPLE
  699. ;
  700. ; You will get more detailed reports (busy etc) if you have a call counter enabled
  701. ; for a device.
  702. ;
  703. ; If you set the busylevel, we will indicate busy when we have a number of calls that
  704. ; matches the busylevel treshold.
  705. ;
  706. ; For queues, you will need this level of detail in status reporting, regardless
  707. ; if you use SIP subscriptions. Queues and manager use the same internal interface
  708. ; for reading status information.
  709. ;
  710. ; Note: Subscriptions does not work if you have a realtime dialplan and use the
  711. ; realtime switch.
  712. ;
  713. ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
  714. ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
  715. ; Useful to limit subscriptions to local extensions
  716. ; Settable per peer/user also
  717. ;notifyringing = no ; Control when subscriptions get notified of ringing state.
  718. ; Specify 'no' to not send any ringing notifications.
  719. ; Specify 'yes' to always send ringing notifications (default).
  720. ; Specify 'notinuse' to only send ringing notifications for
  721. ; extensions that are not currently in use. This is useful as a
  722. ; visual indication of who is available to pick up an incoming call
  723. ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
  724. ; Turning on notifyringing and notifyhold will add a lot
  725. ; more database transactions if you are using realtime.
  726. ;notifycid = yes ; Control whether caller ID information is sent along with
  727. ; dialog-info+xml notifications (supported by snom phones).
  728. ; Note that this feature will only work properly when the
  729. ; incoming call is using the same extension and context that
  730. ; is being used as the hint for the called extension. This means
  731. ; that it won't work when using subscribecontext for your sip
  732. ; user or peer (if subscribecontext is different than context).
  733. ; This is also limited to a single caller, meaning that if an
  734. ; extension is ringing because multiple calls are incoming,
  735. ; only one will be used as the source of caller ID. Specify
  736. ; 'ignore-context' to ignore the called context when looking
  737. ; for the caller's channel. The default value is 'no.' Setting
  738. ; notifycid to 'ignore-context' also causes call-pickups attempted
  739. ; via SNOM's NOTIFY mechanism to set the context for the call pickup
  740. ; to PICKUPMARK.
  741. ;callcounter = yes ; Enable call counters on devices. This can be set per
  742. ; device too.
  743.  
  744. ; ---------------------------------------- T.38 FAX SUPPORT ----------------------------------
  745. ;
  746. ; This setting is available in the [general] section as well as in device configurations.
  747. ; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off.
  748. ;
  749. ; t38pt_udptl = yes ; Enables T.38 with FEC error correction.
  750. ; t38pt_udptl = yes,fec ; Enables T.38 with FEC error correction.
  751. ; t38pt_udptl = yes,redundancy ; Enables T.38 with redundancy error correction.
  752. ; t38pt_udptl = yes,none ; Enables T.38 with no error correction.
  753. ;
  754. ; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that
  755. ; is based on an incorrect interpretation of the T.38 recommendation, and results in failures
  756. ; because Asterisk does not believe it can send T.38 packets of a reasonable size to that
  757. ; endpoint (Cisco media gateways are one example of this situation). In these cases, during a
  758. ; T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL
  759. ; stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you
  760. ; can set an override (globally, or on a per-device basis) to make Asterisk ignore the
  761. ; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead.
  762. ; This can be done by appending 'maxdatagram=<value>' to the t38pt_udptl configuration option,
  763. ; like this:
  764. ;
  765. ; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error correction and overrides
  766. ; ; the other endpoint's provided value to assume we can
  767. ; ; send 400 byte T.38 FAX packets to it.
  768. ;
  769. ; FAX detection will cause the SIP channel to jump to the 'fax' extension (if it exists)
  770. ; based one or more events being detected. The events that can be detected are an incoming
  771. ; CNG tone or an incoming T.38 re-INVITE request.
  772. ;
  773. ; faxdetect = yes ; Default 'no', 'yes' enables both CNG and T.38 detection
  774. ; faxdetect = cng ; Enables only CNG detection
  775. ; faxdetect = t38 ; Enables only T.38 detection
  776. ;
  777. ; ---------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
  778. ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
  779. ; Format for the register statement is:
  780. ; register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
  781. ;
  782. ;
  783. ;
  784. ; domain is either
  785. ; - domain in DNS
  786. ; - host name in DNS
  787. ; - the name of a peer defined below or in realtime
  788. ; The domain is where you register your username, so your SIP uri you are registering to
  789. ; is username@domain
  790. ;
  791. ; If no extension is given, the 's' extension is used. The extension needs to
  792. ; be defined in extensions.conf to be able to accept calls from this SIP proxy
  793. ; (provider).
  794. ;
  795. ; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
  796. ; this is equivalent to having the following line in the general section:
  797. ;
  798. ; register => fromuser:secret:username@host/callbackextension
  799. ;
  800. ; and more readable because you don't have to write the parameters in two places
  801. ; (note that the "port" is ignored - this is a bug that should be fixed).
  802. ;
  803. ; Note that a register= line doesn't mean that we will match the incoming call in any
  804. ; other way than described above. If you want to control where the call enters your
  805. ; dialplan, which context, you want to define a peer with the hostname of the provider's
  806. ; server. If the provider has multiple servers to place calls to your system, you need
  807. ; a peer for each server.
  808. ;
  809. ; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may
  810. ; contain a port number. Since the logical separator between a host and port number is a
  811. ; ':' character, and this character is already used to separate between the optional "secret"
  812. ; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish
  813. ; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
  814. ; they are blank. See the third example below for an illustration.
  815. ;
  816. ;
  817. ; Examples:
  818. ;
  819. ;register => 1234:password@mysipprovider.com
  820. ;
  821. ; This will pass incoming calls to the 's' extension
  822. ;
  823. ;
  824. ;register => 2345:password@sip_proxy/1234
  825. ;
  826. ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
  827. ; connect to local extension 1234 in extensions.conf, default context,
  828. ; unless you configure a [sip_proxy] section below, and configure a
  829. ; context.
  830. ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
  831. ; Tip 2: Use separate inbound and outbound sections for SIP providers
  832. ; (instead of type=friend) if you have calls in both directions
  833. ;
  834. ;register => 3456@mydomain:5082::@mysipprovider.com
  835. ;
  836. ; Note that in this example, the optional authuser and secret portions have
  837. ; been left blank because we have specified a port in the user section
  838. ;
  839. ;register => tls://username:xxxxxx@sip-tls-proxy.example.org
  840. ;
  841. ; The 'transport' part defaults to 'udp' but may also be 'tcp' or 'tls'.
  842. ; Using 'udp://' explicitly is also useful in case the username part
  843. ; contains a '/' ('user/name').
  844.  
  845. ;registertimeout=20 ; retry registration calls every 20 seconds (default)
  846. ;registerattempts=10 ; Number of registration attempts before we give up
  847. ; 0 = continue forever, hammering the other server
  848. ; until it accepts the registration
  849. ; Default is 0 tries, continue forever
  850. ;register_retry_403=yes ; Treat 403 responses to registrations as if they were
  851. ; 401 responses and continue retrying according to normal
  852. ; retry rules.
  853.  
  854. ; ---------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
  855. ; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
  856. ; by other phones. At this time, you can only subscribe using UDP as the transport.
  857. ; Format for the mwi register statement is:
  858. ; mwi => user[:secret[:authuser]]@host[:port]/mailbox
  859. ;
  860. ; Examples:
  861. ;mwi => 1234:password@mysipprovider.com/1234
  862. ;mwi => 1234:password@myportprovider.com:6969/1234
  863. ;mwi => 1234:password:authuser@myauthprovider.com/1234
  864. ;mwi => 1234:password:authuser@myauthportprovider.com:6969/1234
  865. ;
  866. ; MWI received will be stored in the 1234 mailbox of the SIP_Remote context.
  867. ; It can be used by other phones by following the below:
  868. ; mailbox=1234@SIP_Remote
  869. ; ---------------------------------------- NAT SUPPORT ------------------------
  870. ;
  871. ; WARNING: SIP operation behind a NAT is tricky and you really need
  872. ; to read and understand well the following section.
  873. ;
  874. ; When Asterisk is behind a NAT device, the "local" address (and port) that
  875. ; a socket is bound to has different values when seen from the inside or
  876. ; from the outside of the NATted network. Unfortunately this address must
  877. ; be communicated to the outside (e.g. in SIP and SDP messages), and in
  878. ; order to determine the correct value Asterisk needs to know:
  879. ;
  880. ; + whether it is talking to someone "inside" or "outside" of the NATted network.
  881. ; This is configured by assigning the "localnet" parameter with a list
  882. ; of network addresses that are considered "inside" of the NATted network.
  883. ; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
  884. ; Multiple entries are allowed, e.g. a reasonable set is the following:
  885. ;
  886. ; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
  887. ; localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
  888. ; localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
  889. ; localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
  890. ;
  891. ; + the "externally visible" address and port number to be used when talking
  892. ; to a host outside the NAT. This information is derived by one of the
  893. ; following (mutually exclusive) config file parameters:
  894. ;
  895. ; a. "externaddr = hostname[:port]" specifies a static address[:port] to
  896. ; be used in SIP and SDP messages.
  897. ; The hostname is looked up only once, when [re]loading sip.conf .
  898. ; If a port number is not present, use the port specified in the "udpbindaddr"
  899. ; (which is not guaranteed to work correctly, because a NAT box might remap the
  900. ; port number as well as the address).
  901. ; This approach can be useful if you have a NAT device where you can
  902. ; configure the mapping statically. Examples:
  903. ;
  904. ; externaddr = 12.34.56.78 ; use this address.
  905. ; externaddr = 12.34.56.78:9900 ; use this address and port.
  906. ; externaddr = mynat.my.org:12600 ; Public address of my nat box.
  907. ; externtcpport = 9900 ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT.
  908. ; ; externtcpport will default to the externaddr or externhost port if either one is set.
  909. ; externtlsport = 12600 ; The externally mapped tls port, when Asterisk is behind a static NAT or PAT.
  910. ; ; externtlsport port will default to the RFC designated port of 5061.
  911. ;
  912. ; b. "externhost = hostname[:port]" is similar to "externaddr" except
  913. ; that the hostname is looked up every "externrefresh" seconds
  914. ; (default 10s). This can be useful when your NAT device lets you choose
  915. ; the port mapping, but the IP address is dynamic.
  916. ; Beware, you might suffer from service disruption when the name server
  917. ; resolution fails. Examples:
  918. ;
  919. ; externhost=foo.dyndns.net ; refreshed periodically
  920. ; externrefresh=180 ; change the refresh interval
  921. ;
  922. ; Note that at the moment all these mechanism work only for the SIP socket.
  923. ; The IP address discovered with externaddr/externhost is reused for
  924. ; media sessions as well, but the port numbers are not remapped so you
  925. ; may still experience problems.
  926. ;
  927. ; NOTE 1: in some cases, NAT boxes will use different port numbers in
  928. ; the internal<->external mapping. In these cases, the "externaddr" and
  929. ; "externhost" might not help you configure addresses properly.
  930. ;
  931. ; NOTE 2: when using "externaddr" or "externhost", the address part is
  932. ; also used as the external address for media sessions. Thus, the port
  933. ; information in the SDP may be wrong!
  934. ;
  935. ; In addition to the above, Asterisk has an additional "nat" parameter to
  936. ; address NAT-related issues in incoming SIP or media sessions.
  937. ; In particular, depending on the 'nat= ' settings described below, Asterisk
  938. ; may override the address/port information specified in the SIP/SDP messages,
  939. ; and use the information (sender address) supplied by the network stack instead.
  940. ; However, this is only useful if the external traffic can reach us.
  941. ; The following settings are allowed (both globally and in individual sections):
  942. ;
  943. ; nat = no ; Do no special NAT handling other than RFC3581
  944. ; nat = force_rport ; Pretend there was an rport parameter even if there wasn't
  945. ; nat = comedia ; Send media to the port Asterisk received it from regardless
  946. ; ; of where the SDP says to send it.
  947. ; nat = auto_force_rport ; Set the force_rport option if Asterisk detects NAT (default)
  948. ; nat = auto_comedia ; Set the comedia option if Asterisk detects NAT
  949. ;
  950. ; The nat settings can be combined. For example, to set both force_rport and comedia
  951. ; one would set nat=force_rport,comedia. If any of the comma-separated options is 'no',
  952. ; Asterisk will ignore any other settings and set nat=no. If one of the "auto" settings
  953. ; is used in conjunction with its non-auto counterpart (nat=comedia,auto_comedia), then
  954. ; the non-auto option will be ignored.
  955. ;
  956. ; The RFC 3581-defined 'rport' parameter allows a client to request that Asterisk send
  957. ; SIP responses to it via the source IP and port from which the request originated
  958. ; instead of the address/port listed in the top-most Via header. This is useful if a
  959. ; client knows that it is behind a NAT and therefore cannot guess from what address/port
  960. ; its request will be sent. Asterisk will always honor the 'rport' parameter if it is
  961. ; sent. The force_rport setting causes Asterisk to always send responses back to the
  962. ; address/port from which it received requests; even if the other side doesn't support
  963. ; adding the 'rport' parameter.
  964. ;
  965. ; 'comedia RTP handling' refers to the technique of sending RTP to the port that the
  966. ; the other endpoint's RTP arrived from, and means 'connection-oriented media'. This is
  967. ; only partially related to RFC 4145 which was referred to as COMEDIA while it was in
  968. ; draft form. This method is used to accomodate endpoints that may be located behind
  969. ; NAT devices, and as such the address/port they tell Asterisk to send RTP packets to
  970. ; for their media streams is not the actual address/port that will be used on the nearer
  971. ; side of the NAT.
  972. ;
  973. ; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
  974. ; the nat setting in a peer definition, then the peer username will be discoverable
  975. ; by outside parties as Asterisk will respond to different ports for defined and
  976. ; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
  977. ; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the
  978. ; other, then valid peers with settings differing from those in the general section will
  979. ; be discoverable.
  980. ;
  981. ; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by
  982. ; RFC 4961; Asterisk will always send RTP packets from the same port number it expects
  983. ; to receive them on.
  984. ;
  985. ; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using
  986. ; the media_address configuration option. This is only applicable to the general section and
  987. ; can not be set per-user or per-peer.
  988. ;
  989. ; Note that this does not change the listen address for RTP, it only changes the
  990. ; advertised address in the SDP. The Asterisk RTP engine will still listen on
  991. ; the standard IP address.
  992. ;
  993. ; media_address = 172.16.42.1
  994. ;
  995. ; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the
  996. ; perceived external network address has changed. When the stun_monitor is installed and
  997. ; configured, chan_sip will renew all outbound registrations when the monitor detects any sort
  998. ; of network change has occurred. By default this option is enabled, but only takes effect once
  999. ; res_stun_monitor is configured. If res_stun_monitor is enabled and you wish to not
  1000. ; generate all outbound registrations on a network change, use the option below to disable
  1001. ; this feature.
  1002. ;
  1003. ; subscribe_network_change_event = yes ; on by default
  1004. ;
  1005. ; ICE/STUN/TURN usage can be enabled globally or on a per-peer basis using the icesupport
  1006. ; configuration option. When set to yes ICE support is enabled. When set to no it is disabled.
  1007. ; It is disabled by default.
  1008. ;
  1009. ; icesupport = yes
  1010.  
  1011. ; ---------------------------------- MEDIA HANDLING --------------------------------
  1012. ; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
  1013. ; no reason for Asterisk to stay in the media path, the media will be redirected.
  1014. ; This does not really work well in the case where Asterisk is outside and the
  1015. ; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
  1016. ;
  1017. ;directmedia=yes ; Asterisk by default tries to redirect the
  1018. ; RTP media stream to go directly from
  1019. ; the caller to the callee. Some devices do not
  1020. ; support this (especially if one of them is behind a NAT).
  1021. ; The default setting is YES. If you have all clients
  1022. ; behind a NAT, or for some other reason want Asterisk to
  1023. ; stay in the audio path, you may want to turn this off.
  1024.  
  1025. ; This setting also affect direct RTP
  1026. ; at call setup (a new feature in 1.4 - setting up the
  1027. ; call directly between the endpoints instead of sending
  1028. ; a re-INVITE).
  1029.  
  1030. ; Additionally this option does not disable all reINVITE operations.
  1031. ; It only controls Asterisk generating reINVITEs for the specific
  1032. ; purpose of setting up a direct media path. If a reINVITE is
  1033. ; needed to switch a media stream to inactive (when placed on
  1034. ; hold) or to T.38, it will still be done, regardless of this
  1035. ; setting. Note that direct T.38 is not supported.
  1036.  
  1037. ;directmedia=nonat ; An additional option is to allow media path redirection
  1038. ; (reinvite) but only when the peer where the media is being
  1039. ; sent is known to not be behind a NAT (as the RTP core can
  1040. ; determine it based on the apparent IP address the media
  1041. ; arrives from).
  1042.  
  1043. ;directmedia=update ; Yet a third option... use UPDATE for media path redirection,
  1044. ; instead of INVITE. This can be combined with 'nonat', as
  1045. ; 'directmedia=update,nonat'. It implies 'yes'.
  1046.  
  1047. ;directmedia=outgoing ; When sending directmedia reinvites, do not send an immediate
  1048. ; reinvite on an incoming call leg. This option is useful when
  1049. ; peered with another SIP user agent that is known to send
  1050. ; immediate direct media reinvites upon call establishment. Setting
  1051. ; the option in this situation helps to prevent potential glares.
  1052. ; Setting this option implies 'yes'.
  1053.  
  1054. ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
  1055. ; the call directly with media peer-2-peer without re-invites.
  1056. ; Will not work for video and cases where the callee sends
  1057. ; RTP payloads and fmtp headers in the 200 OK that does not match the
  1058. ; callers INVITE. This will also fail if directmedia is enabled when
  1059. ; the device is actually behind NAT.
  1060.  
  1061. ;directmediadeny=0.0.0.0/0 ; Use directmediapermit and directmediadeny to restrict
  1062. ;directmediapermit=172.16.0.0/16; which peers should be able to pass directmedia to each other
  1063. ; (There is no default setting, this is just an example)
  1064. ; Use this if some of your phones are on IP addresses that
  1065. ; can not reach each other directly. This way you can force
  1066. ; RTP to always flow through asterisk in such cases.
  1067. ;directmediaacl=acl_example ; Use named ACLs defined in acl.conf
  1068.  
  1069. ;ignoresdpversion=yes ; By default, Asterisk will honor the session version
  1070. ; number in SDP packets and will only modify the SDP
  1071. ; session if the version number changes. This option will
  1072. ; force asterisk to ignore the SDP session version number
  1073. ; and treat all SDP data as new data. This is required
  1074. ; for devices that send us non standard SDP packets
  1075. ; (observed with Microsoft OCS). By default this option is
  1076. ; off.
  1077.  
  1078. ;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
  1079. ; Like the useragent parameter, the default user agent string
  1080. ; also contains the Asterisk version.
  1081. ;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
  1082. ; This field MUST NOT contain spaces
  1083. ;encryption=no ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
  1084. ; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
  1085. ; the peer does not support SRTP. Defaults to no.
  1086. ;encryption_taglen=80 ; Set the auth tag length offered in the INVITE either 32/80 default 80
  1087. ;
  1088. ;avpf=yes ; Enable inter-operability with media streams using the AVPF RTP profile.
  1089. ; This will cause all offers and answers to use AVPF (or SAVPF). This
  1090. ; option may be specified at the global or peer scope.
  1091. ;force_avp=yes ; Force 'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', and 'RTP/SAVPF' to be used for
  1092. ; media streams when appropriate, even if a DTLS stream is present.
  1093. ;rtcp_mux=yes ; Enable support for RFC 5761 RTCP multiplexing which is required for
  1094. ; WebRTC support
  1095. ; ---------------------------------------- REALTIME SUPPORT ------------------------
  1096. ; For additional information on ARA, the Asterisk Realtime Architecture,
  1097. ; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
  1098. ;
  1099. ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
  1100. ; just like friends added from the config file only on a
  1101. ; as-needed basis? (yes|no)
  1102.  
  1103. ;rtsavesysname=yes ; Save systemname in realtime database at registration
  1104. ; Default= no
  1105.  
  1106. ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
  1107. ; If set to yes, when a SIP UA registers successfully, the ip address,
  1108. ; the origination port, the registration period, and the username of
  1109. ; the UA will be set to database via realtime.
  1110. ; If not present, defaults to 'yes'. Note: realtime peers will
  1111. ; probably not function across reloads in the way that you expect, if
  1112. ; you turn this option off.
  1113. ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
  1114. ; as if it had just registered? (yes|no|<seconds>)
  1115. ; If set to yes, when the registration expires, the friend will
  1116. ; vanish from the configuration until requested again. If set
  1117. ; to an integer, friends expire within this number of seconds
  1118. ; instead of the registration interval.
  1119.  
  1120. ;ignoreregexpire=yes ; Enabling this setting has two functions:
  1121. ;
  1122. ; For non-realtime peers, when their registration expires, the
  1123. ; information will _not_ be removed from memory or the Asterisk database
  1124. ; if you attempt to place a call to the peer, the existing information
  1125. ; will be used in spite of it having expired
  1126. ;
  1127. ; For realtime peers, when the peer is retrieved from realtime storage,
  1128. ; the registration information will be used regardless of whether
  1129. ; it has expired or not; if it expires while the realtime peer
  1130. ; is still in memory (due to caching or other reasons), the
  1131. ; information will not be removed from realtime storage
  1132.  
  1133. ; ---------------------------------------- SIP DOMAIN SUPPORT ------------------------
  1134. ; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
  1135. ; domains, each of which can direct the call to a specific context if desired.
  1136. ; By default, all domains are accepted and sent to the default context or the
  1137. ; context associated with the user/peer placing the call.
  1138. ; REGISTER to non-local domains will be automatically denied if a domain
  1139. ; list is configured.
  1140. ;
  1141. ; Domains can be specified using:
  1142. ; domain=<domain>[,<context>]
  1143. ; Examples:
  1144. ; domain=myasterisk.dom
  1145. ; domain=customer.com,customer-context
  1146. ;
  1147. ; In addition, all the 'default' domains associated with a server should be
  1148. ; added if incoming request filtering is desired.
  1149. ; autodomain=yes
  1150. ;
  1151. ; To disallow requests for domains not serviced by this server:
  1152. ; allowexternaldomains=no
  1153.  
  1154. ;domain=mydomain.tld,mydomain-incoming
  1155. ; Add domain and configure incoming context
  1156. ; for external calls to this domain
  1157. ;domain=1.2.3.4 ; Add IP address as local domain
  1158. ; You can have several "domain" settings
  1159. ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
  1160. ; Default is yes
  1161. ;autodomain=yes ; Turn this on to have Asterisk add local host
  1162. ; name and local IP to domain list.
  1163.  
  1164. ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
  1165. ; non-peers, use your primary domain "identity"
  1166. ; for From: headers instead of just your IP
  1167. ; address. This is to be polite and
  1168. ; it may be a mandatory requirement for some
  1169. ; destinations which do not have a prior
  1170. ; account relationship with your server.
  1171.  
  1172. ; ----------------------------- Advice of Charge CONFIGURATION --------------------------
  1173. ; snom_aoc_enabled = yes; ; This options turns on and off support for sending AOC-D and
  1174. ; AOC-E to snom endpoints. This option can be used both in the
  1175. ; peer and global scope. The default for this option is off.
  1176.  
  1177.  
  1178. ; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
  1179. ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
  1180. ; SIP channel. Defaults to "no". An enabled jitterbuffer will
  1181. ; be used only if the sending side can create and the receiving
  1182. ; side can not accept jitter. The SIP channel can accept jitter,
  1183. ; thus a jitterbuffer on the receive SIP side will be used only
  1184. ; if it is forced and enabled.
  1185.  
  1186. ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
  1187. ; channel. Defaults to "no".
  1188.  
  1189. ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
  1190.  
  1191. ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
  1192. ; resynchronized. Useful to improve the quality of the voice, with
  1193. ; big jumps in/broken timestamps, usually sent from exotic devices
  1194. ; and programs. Defaults to 1000.
  1195.  
  1196. ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
  1197. ; channel. Two implementations are currently available - "fixed"
  1198. ; (with size always equals to jbmaxsize) and "adaptive" (with
  1199. ; variable size, actually the new jb of IAX2). Defaults to fixed.
  1200.  
  1201. ; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
  1202. ; The option represents the number of milliseconds by which the new jitter buffer
  1203. ; will pad its size. the default is 40, so without modification, the new
  1204. ; jitter buffer will set its size to the jitter value plus 40 milliseconds.
  1205. ; increasing this value may help if your network normally has low jitter,
  1206. ; but occasionally has spikes.
  1207.  
  1208. ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
  1209.  
  1210. ; ----------------------------------------------------------------------------------
  1211.  
  1212. [authentication]
  1213. ; Global credentials for outbound calls, i.e. when a proxy challenges your
  1214. ; Asterisk server for authentication. These credentials override
  1215. ; any credentials in peer/register definition if realm is matched.
  1216. ;
  1217. ; This way, Asterisk can authenticate for outbound calls to other
  1218. ; realms. We match realm on the proxy challenge and pick an set of
  1219. ; credentials from this list
  1220. ; Syntax:
  1221. ; auth = <user>:<secret>@<realm>
  1222. ; auth = <user>#<md5secret>@<realm>
  1223. ; Example:
  1224. ;auth=mark:topsecret@digium.com
  1225. ;
  1226. ; You may also add auth= statements to [peer] definitions
  1227. ; Peer auth= override all other authentication settings if we match on realm
  1228.  
  1229. ; -----------------------------------------------------------------------------
  1230. ; DEVICE CONFIGURATION
  1231. ;
  1232. ; SIP entities have a 'type' which determines their roles within Asterisk.
  1233. ; * For entities with 'type=peer':
  1234. ; Peers handle both inbound and outbound calls and are matched by ip/port, so for
  1235. ; The case of incoming calls from the peer, the IP address must match in order for
  1236. ; The invitation to work. This means calls made from either direction won't work if
  1237. ; The peer is unregistered while host=dynamic or if the host is otherise not set to
  1238. ; the correct IP of the sender.
  1239. ; * For entities with 'type=user':
  1240. ; Asterisk users handle inbound calls only (meaning they call Asterisk, Asterisk can't
  1241. ; call them) and are matched by their authorization information (authname and secret).
  1242. ; Asterisk doesn't rely on their IP and will accept calls regardless of the host setting
  1243. ; as long as the incoming SIP invite authorizes successfully.
  1244. ; * For entities with 'type=friend':
  1245. ; Asterisk will create the entity as both a friend and a peer. Asterisk will accept
  1246. ; calls from friends like it would for users, requiring only that the authorization
  1247. ; matches rather than the IP address. Since it is also a peer, a friend entity can
  1248. ; be called as long as its IP is known to Asterisk. In the case of host=dynamic,
  1249. ; this means it is necessary for the entity to register before Asterisk can call it.
  1250. ;
  1251. ; Use remotesecret for outbound authentication, and secret for authenticating
  1252. ; inbound requests. For historical reasons, if no remotesecret is supplied for an
  1253. ; outbound registration or call, the secret will be used.
  1254. ;
  1255. ; For device names, we recommend using only a-z, numerics (0-9) and underscore
  1256. ;
  1257. ; For local phones, type=friend works most of the time
  1258. ;
  1259. ; If you have one-way audio, you probably have NAT problems.
  1260. ; If Asterisk is on a public IP, and the phone is inside of a NAT device
  1261. ; you will need to configure nat option for those phones.
  1262. ; Also, turn on qualify=yes to keep the nat session open
  1263. ;
  1264. ; Configuration options available
  1265. ; --------------------
  1266. ; context
  1267. ; callingpres
  1268. ; permit
  1269. ; deny
  1270. ; secret
  1271. ; md5secret
  1272. ; remotesecret
  1273. ; transport
  1274. ; dtmfmode
  1275. ; directmedia
  1276. ; nat
  1277. ; callgroup
  1278. ; pickupgroup
  1279. ; language
  1280. ; allow
  1281. ; disallow
  1282. ; autoframing
  1283. ; insecure
  1284. ; trustrpid
  1285. ; trust_id_outbound
  1286. ; progressinband
  1287. ; promiscredir
  1288. ; useclientcode
  1289. ; accountcode
  1290. ; setvar
  1291. ; callerid
  1292. ; amaflags
  1293. ; callcounter
  1294. ; busylevel
  1295. ; allowoverlap
  1296. ; allowsubscribe
  1297. ; allowtransfer
  1298. ; ignoresdpversion
  1299. ; subscribecontext
  1300. ; template
  1301. ; videosupport
  1302. ; maxcallbitrate
  1303. ; rfc2833compensate
  1304. ; Note: app_voicemail mailboxes must be in the form of mailbox@context.
  1305. ; mailbox
  1306. ; session-timers
  1307. ; session-expires
  1308. ; session-minse
  1309. ; session-refresher
  1310. ; t38pt_usertpsource
  1311. ; regexten
  1312. ; fromdomain
  1313. ; fromuser
  1314. ; host
  1315. ; port
  1316. ; qualify
  1317. ; keepalive
  1318. ; defaultip
  1319. ; defaultuser
  1320. ; rtptimeout
  1321. ; rtpholdtimeout
  1322. ; sendrpid
  1323. ; outboundproxy
  1324. ; rfc2833compensate
  1325. ; callbackextension
  1326. ; timert1
  1327. ; timerb
  1328. ; qualifyfreq
  1329. ; t38pt_usertpsource
  1330. ; contactpermit ; Limit what a host may register as (a neat trick
  1331. ; contactdeny ; is to register at the same IP as a SIP provider,
  1332. ; contactacl ; then call oneself, and get redirected to that
  1333. ; ; same location).
  1334. ; directmediapermit
  1335. ; directmediadeny
  1336. ; directmediaacl
  1337. ; unsolicited_mailbox
  1338. ; use_q850_reason
  1339. ; maxforwards
  1340. ; encryption
  1341. ; description ; Used to provide a description of the peer in console output
  1342. ; dtlsenable
  1343. ; dtlsautogeneratecert
  1344. ; dtlsverify
  1345. ; dtlsrekey
  1346. ; dtlscertfile
  1347. ; dtlsprivatekey
  1348. ; dtlscipher
  1349. ; dtlscafile
  1350. ; dtlscapath
  1351. ; dtlssetup
  1352. ; dtlsfingerprint
  1353. ; ignore_requested_pref ; Ignore the requested codec and determine the preferred codec
  1354. ; ; from the peer's configuration.
  1355. ;
  1356.  
  1357. ; -----------------------------------------------------------------------------
  1358. ; DTLS-SRTP CONFIGURATION
  1359. ;
  1360. ; DTLS-SRTP support is available if the underlying RTP engine in use supports it.
  1361. ;
  1362. ; Note that all configuration options except dtlsenable can be set at the general level.
  1363. ; If set they will be present on the user or peer unless overridden with a different value.
  1364. ;
  1365. ; dtlsenable = yes ; Enable or disable DTLS-SRTP support
  1366. ; dtlsverify = yes ; Verify that provided peer certificate and fingerprint are valid
  1367. ; ; A value of 'yes' will perform both certificate and fingerprint verification
  1368. ; ; A value of 'no' will perform no certificate or fingerprint verification
  1369. ; ; A value of 'fingerprint' will perform ONLY fingerprint verification
  1370. ; ; A value of 'certificate' will perform ONLY certficiate verification
  1371. ; dtlsrekey = 60 ; Interval at which to renegotiate the TLS session and rekey the SRTP session
  1372. ; ; If this is not set or the value provided is 0 rekeying will be disabled
  1373. ; dtlsautogeneratecert = yes ; Enable ephemeral DTLS certificate generation. The default is 'no.'
  1374. ; dtlscertfile = file ; Path to certificate file to present
  1375. ; dtlsprivatekey = file ; Path to private key for certificate file
  1376. ; dtlscipher = <SSL cipher string> ; Cipher to use for TLS negotiation
  1377. ; ; A list of valid SSL cipher strings can be found at:
  1378. ; ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
  1379. ; dtlscafile = file ; Path to certificate authority certificate
  1380. ; dtlscapath = path ; Path to a directory containing certificate authority certificates
  1381. ; dtlssetup = actpass ; Whether we are willing to accept connections, connect to the other party, or both.
  1382. ; ; Valid options are active (we want to connect to the other party), passive (we want to
  1383. ; ; accept connections only), and actpass (we will do both). This value will be used in
  1384. ; ; the outgoing SDP when offering and for incoming SDP offers when the remote party sends
  1385. ; ; actpass
  1386. ; dtlsfingerprint = sha-1 ; The hash to use for the fingerprint in SDP (valid options are sha-1 and sha-256)
  1387.  
  1388. ;[sip_proxy]
  1389. ; For incoming calls only. Example: FWD (Free World Dialup)
  1390. ; We match on IP address of the proxy for incoming calls
  1391. ; since we can not match on username (caller id)
  1392. ;type=peer
  1393. ;context=from-fwd
  1394. ;host=fwd.pulver.com
  1395.  
  1396. ;[sip_proxy-out]
  1397. ;type=peer ; we only want to call out, not be called
  1398. ;remotesecret=guessit ; Our password to their service
  1399. ;defaultuser=yourusername ; Authentication user for outbound proxies
  1400. ;fromuser=yourusername ; Many SIP providers require this!
  1401. ;fromdomain=provider.sip.domain
  1402. ;host=box.provider.com
  1403. ;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will
  1404. ; ; accept both tcp and udp. The default transport type is only used for
  1405. ; ; outbound messages until a Registration takes place. During the
  1406. ; ; peer Registration the transport type may change to another supported
  1407. ; ; type if the peer requests so.
  1408.  
  1409. ;usereqphone=yes ; This provider requires ";user=phone" on URI
  1410. ;callcounter=yes ; Enable call counter
  1411. ;busylevel=2 ; Signal busy at 2 or more calls
  1412. ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
  1413. ;port=80 ; The port number we want to connect to on the remote side
  1414. ; Also used as "defaultport" in combination with "defaultip" settings
  1415.  
  1416. ; -- sample definition for a provider
  1417. ;[provider1]
  1418. ;type=peer
  1419. ;host=sip.provider1.com
  1420. ;fromuser=4015552299 ; how your provider knows you
  1421. ;remotesecret=youwillneverguessit ; The password we use to authenticate to them
  1422. ;secret=gissadetdu ; The password they use to contact us
  1423. ;callbackextension=123 ; Register with this server and require calls coming back to this extension
  1424. ;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
  1425. ; ; accept both tcp and udp. Default is udp. The first transport
  1426. ; ; listed will always be used for outgoing connections.
  1427. ;unsolicited_mailbox=4015552299 ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old
  1428. ; ; message count will be stored in the configured virtual mailbox. It can be used
  1429. ; ; by any device supporting MWI by specifying <configured value>@SIP_Remote as the
  1430. ; ; mailbox.
  1431.  
  1432. ;
  1433. ; Because you might have a large number of similar sections, it is generally
  1434. ; convenient to use templates for the common parameters, and add them
  1435. ; the the various sections. Examples are below, and we can even leave
  1436. ; the templates uncommented as they will not harm:
  1437.  
  1438. [basic-options](!) ; a template
  1439. dtmfmode=rfc2833
  1440. context=from-office
  1441. type=friend
  1442.  
  1443. [natted-phone](!,basic-options) ; another template inheriting basic-options
  1444. directmedia=no
  1445. host=dynamic
  1446.  
  1447. [public-phone](!,basic-options) ; another template inheriting basic-options
  1448. directmedia=yes
  1449.  
  1450. [my-codecs](!) ; a template for my preferred codecs
  1451. disallow=all
  1452. allow=ilbc
  1453. allow=g729
  1454. allow=gsm
  1455. allow=g723
  1456. allow=ulaw
  1457. ; Or, more simply:
  1458. ;allow=!all,ilbc,g729,gsm,g723,ulaw
  1459.  
  1460. [ulaw-phone](!) ; and another one for ulaw-only
  1461. disallow=all
  1462. allow=ulaw
  1463. ; Again, more simply:
  1464. ;allow=!all,ulaw
  1465.  
  1466. ; and finally instantiate a few phones
  1467. ;
  1468. ; [2133](natted-phone,my-codecs)
  1469. ; secret = peekaboo
  1470. ; [2134](natted-phone,ulaw-phone)
  1471. ; secret = not_very_secret
  1472. ; [2136](public-phone,ulaw-phone)
  1473. ; secret = not_very_secret_either
  1474. ; ...
  1475. ;
  1476.  
  1477. ; Standard configurations not using templates look like this:
  1478. ;
  1479. ;[grandstream1]
  1480. ;type=friend
  1481. ;context=from-sip ; Where to start in the dialplan when this phone calls
  1482. ;recordonfeature=dynamicfeature1 ; Feature to use when INFO with Record: on is received.
  1483. ;recordofffeature=dynamicfeature2 ; Feature to use when INFO with Record: off is received.
  1484. ;callerid=John Doe <1234> ; Full caller ID, to override the phones config
  1485. ; on incoming calls to Asterisk
  1486. ;description=Courtesy Phone ; Description of the peer. Shown when doing 'sip show peers'.
  1487. ;host=192.168.0.23 ; we have a static but private IP address
  1488. ; No registration allowed
  1489. ;directmedia=yes ; allow RTP voice traffic to bypass Asterisk
  1490. ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
  1491. ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
  1492. ; from the phone to asterisk (deprecated)
  1493. ; 1 for the explicit peer, 1 for the explicit user,
  1494. ; remember that a friend equals 1 peer and 1 user in
  1495. ; memory
  1496. ; There is no combined call counter for a "friend"
  1497. ; so there's currently no way in sip.conf to limit
  1498. ; to one inbound or outbound call per phone. Use
  1499. ; the group counters in the dial plan for that.
  1500. ;
  1501. ;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
  1502. ;disallow=all ; need to disallow=all before we can use allow=
  1503. ;allow=ulaw ; Note: In user sections the order of codecs
  1504. ; listed with allow= does NOT matter!
  1505. ;allow=alaw
  1506. ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
  1507. ;allow=g729 ; Pass-thru only unless g729 license obtained
  1508. ;callingpres=allowed_passed_screen ; Set caller ID presentation
  1509. ; See function CALLERPRES documentation for possible
  1510. ; values.
  1511.  
  1512. ;[xlite1]
  1513. ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
  1514. ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
  1515. ;type=friend
  1516. ;regexten=1234 ; When they register, create extension 1234
  1517. ;callerid="Jane Smith" <5678>
  1518. ;host=dynamic ; This device needs to register
  1519. ;directmedia=no ; Typically set to NO if behind NAT
  1520. ;disallow=all
  1521. ;allow=gsm ; GSM consumes far less bandwidth than ulaw
  1522. ;allow=ulaw
  1523. ;allow=alaw
  1524. ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
  1525.  
  1526. ;[snom]
  1527. ;type=friend ; Friends place calls and receive calls
  1528. ;context=from-sip ; Context for incoming calls from this user
  1529. ;secret=blah
  1530. ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
  1531. ;language=de ; Use German prompts for this user
  1532. ;host=dynamic ; This peer register with us
  1533. ;dtmfmode=inband ; Choices are inband, rfc2833, or info
  1534. ;defaultip=192.168.0.59 ; IP used until peer registers
  1535. ;mailbox=1234@context,2345@context ; Mailbox(-es) for message waiting indicator
  1536. ;subscribemwi=yes ; Only send notifications if this phone
  1537. ; subscribes for mailbox notification
  1538. ;vmexten=voicemail ; dialplan extension to reach mailbox
  1539. ; sets the Message-Account in the MWI notify message
  1540. ; defaults to global vmexten which defaults to "asterisk"
  1541. ;disallow=all
  1542. ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
  1543.  
  1544.  
  1545. ;[polycom]
  1546. ;type=friend ; Friends place calls and receive calls
  1547. ;context=from-sip ; Context for incoming calls from this user
  1548. ;secret=blahpoly
  1549. ;host=dynamic ; This peer register with us
  1550. ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
  1551. ;defaultuser=polly ; Username to use in INVITE until peer registers
  1552. ;defaultip=192.168.40.123
  1553. ; Normally you do NOT need to set this parameter
  1554. ;disallow=all
  1555. ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
  1556. ;progressinband=no ; Polycom phones don't work properly with "never"
  1557.  
  1558.  
  1559. ;[pingtel]
  1560. ;type=friend
  1561. ;secret=blah
  1562. ;host=dynamic
  1563. ;insecure=port ; Allow matching of peer by IP address without
  1564. ; matching port number
  1565. ;insecure=invite ; Do not require authentication of incoming INVITEs
  1566. ;insecure=port,invite ; (both)
  1567. ;qualify=1000 ; Consider it down if it's 1 second to reply
  1568. ; Helps with NAT session
  1569. ; qualify=yes uses default value
  1570. ;qualifyfreq=60 ; Qualification: How often to check for the
  1571. ; host to be up in seconds
  1572. ; Set to low value if you use low timeout for
  1573. ; NAT of UDP sessions
  1574. ;
  1575. ; Call group and Pickup group should be in the range from 0 to 63
  1576. ;
  1577. ;callgroup=1,3-4 ; We are in caller groups 1,3,4
  1578. ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
  1579. ;namedcallgroup=engineering,sales,netgroup,protgroup ; We are in named call groups engineering,sales,netgroup,protgroup
  1580. ;namedpickupgroup=sales ; We can do call pick-p for named call group sales
  1581. ;defaultip=192.168.0.60 ; IP address to use if peer has not registered
  1582. ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
  1583. ;permit=192.168.0.60/255.255.255.0
  1584. ;permit=192.168.0.60/24 ; we can also use CIDR notation for subnet masks
  1585. ;permit=2001:db8::/32 ; IPv6 ACLs can be specified if desired. IPv6 ACLs
  1586. ; apply only to IPv6 addresses, and IPv4 ACLs apply
  1587. ; only to IPv4 addresses.
  1588. ;acl=named_acl_example ; Use named ACLs defined in acl.conf
  1589.  
  1590. ;[cisco1]
  1591. ;type=friend
  1592. ;secret=blah
  1593. ;qualify=200 ; Qualify peer is no more than 200ms away
  1594. ;host=dynamic ; This device registers with us
  1595. ;directmedia=no ; Asterisk by default tries to redirect the
  1596. ; RTP media stream (audio) to go directly from
  1597. ; the caller to the callee. Some devices do not
  1598. ; support this (especially if one of them is
  1599. ; behind a NAT).
  1600. ;defaultip=192.168.0.4 ; IP address to use until registration
  1601. ;defaultuser=goran ; Username to use when calling this device before registration
  1602. ; Normally you do NOT need to set this parameter
  1603. ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from or to this device
  1604. ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
  1605. ; cause the given audio file to
  1606. ; be played upon completion of
  1607. ; an attended transfer to the
  1608. ; target of the transfer.
  1609.  
  1610. ;[pre14-asterisk]
  1611. ;type=friend
  1612. ;secret=digium
  1613. ;host=dynamic
  1614. ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
  1615. ; You must have this turned on or DTMF reception will work improperly.
  1616. ;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
  1617. ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
  1618. ; external IP address of the remote device. If port forwarding is done at the client side
  1619. ; then UDPTL will flow to the remote device.
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