API tools faq
paste
Advertisement
Guest User

klaus-sip

a guest
May 8th, 2013
189
0
Never
Add comment
Not a member of Pastebin yet? Sign Up, it unlocks many cool features!
text 77.13 KB | None | 0 0
raw download clone embed print report
  1. ;!
  2. ;! Automatically generated configuration file
  3. ;! Filename: sip.conf (/etc/asterisk/sip.conf)
  4. ;! Generator: Manager
  5. ;! Creation Date: Fri May 3 17:34:18 2013
  6. ;!
  7. ;
  8. ; SIP Configuration example for Asterisk
  9. ;
  10. ; Note: Please read the security documentation for Asterisk in order to
  11. ; understand the risks of installing Asterisk with the sample
  12. ; configuration. If your Asterisk is installed on a public
  13. ; IP address connected to the Internet, you will want to learn
  14. ; about the various security settings BEFORE you start
  15. ; Asterisk.
  16. ;
  17. ; Especially note the following settings:
  18. ; - allowguest (default enabled)
  19. ; - permit/deny/acl - IP address filters
  20. ; - contactpermit/contactdeny/contactacl - IP address filters for registrations
  21. ; - context - Which set of services you offer various users
  22. ;
  23. ; SIP dial strings
  24. ;-----------------------------------------------------------
  25. ; In the dialplan (extensions.conf) you can use several
  26. ; syntaxes for dialing SIP devices.
  27. ; SIP/devicename
  28. ; SIP/username@domain (SIP uri)
  29. ; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
  30. ; SIP/devicename/extension
  31. ; SIP/devicename/extension/IPorHost
  32. ; SIP/username@domain//IPorHost
  33. ;
  34. ;
  35. ; Devicename
  36. ; devicename is defined as a peer in a section below.
  37. ;
  38. ; username@domain
  39. ; Call any SIP user on the Internet
  40. ; (Don't forget to enable DNS SRV records if you want to use this)
  41. ;
  42. ; devicename/extension
  43. ; If you define a SIP proxy as a peer below, you may call
  44. ; SIP/proxyhostname/user or SIP/user@proxyhostname
  45. ; where the proxyhostname is defined in a section below
  46. ; This syntax also works with ATA's with FXO ports
  47. ;
  48. ; SIP/username[:password[:md5secret[:authname]]]@host[:port]
  49. ; This form allows you to specify password or md5secret and authname
  50. ; without altering any authentication data in config.
  51. ; Examples:
  52. ;
  53. ; SIP/*98@mysipproxy
  54. ; SIP/sales:topsecret::account02@domain.com:5062
  55. ; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
  56. ;
  57. ; IPorHost
  58. ; The next server for this call regardless of domain/peer
  59. ;
  60. ; All of these dial strings specify the SIP request URI.
  61. ; In addition, you can specify a specific To: header by adding an
  62. ; exclamation mark after the dial string, like
  63. ;
  64. ; SIP/sales@mysipproxy!sales@edvina.net
  65. ;
  66. ; A new feature for 1.8 allows one to specify a host or IP address to use
  67. ; when routing the call. This is typically used in tandem with func_srv if
  68. ; multiple methods of reaching the same domain exist. The host or IP address
  69. ; is specified after the third slash in the dialstring. Examples:
  70. ;
  71. ; SIP/devicename/extension/IPorHost
  72. ; SIP/username@domain//IPorHost
  73. ;
  74. ; CLI Commands
  75. ; -------------------------------------------------------------
  76. ; Useful CLI commands to check peers/users:
  77. ; sip show peers Show all SIP peers (including friends)
  78. ; sip show registry Show status of hosts we register with
  79. ;
  80. ; sip set debug on Show all SIP messages
  81. ;
  82. ; sip reload Reload configuration file
  83. ; sip show settings Show the current channel configuration
  84. ;
  85. ;------- Naming devices ------------------------------------------------------
  86. ;
  87. ; When naming devices, make sure you understand how Asterisk matches calls
  88. ; that come in.
  89. ; 1. Asterisk checks the SIP From: address username and matches against
  90. ; names of devices with type=user
  91. ; The name is the text between square brackets [name]
  92. ; 2. Asterisk checks the From: addres and matches the list of devices
  93. ; with a type=peer
  94. ; 3. Asterisk checks the IP address (and port number) that the INVITE
  95. ; was sent from and matches against any devices with type=peer
  96. ;
  97. ; Don't mix extensions with the names of the devices. Devices need a unique
  98. ; name. The device name is *not* used as phone numbers. Phone numbers are
  99. ; anything you declare as an extension in the dialplan (extensions.conf).
  100. ;
  101. ; When setting up trunks, make sure there's no risk that any From: username
  102. ; (caller ID) will match any of your device names, because then Asterisk
  103. ; might match the wrong device.
  104. ;
  105. ; Note: The parameter "username" is not the username and in most cases is
  106. ; not needed at all. Check below. In later releases, it's renamed
  107. ; to "defaultuser" which is a better name, since it is used in
  108. ; combination with the "defaultip" setting.
  109. ;-----------------------------------------------------------------------------
  110.  
  111. ; ** Old configuration options **
  112. ; The "call-limit" configuation option is considered old is replaced
  113. ; by new functionality. To enable callcounters, you use the new
  114. ; "callcounter" setting (for extension states in queue and subscriptions)
  115. ; You are encouraged to use the dialplan groupcount functionality
  116. ; to enforce call limits instead of using this channel-specific method.
  117. ; You can still set limits per device in sip.conf or in a database by using
  118. ; "setvar" to set variables that can be used in the dialplan for various limits.
  119.  
  120. [general]
  121. context = public ; Default context for incoming calls. Defaults to 'default'
  122. ;allowguest=no ; Allow or reject guest calls (default is yes)
  123. ; If your Asterisk is connected to the Internet
  124. ; and you have allowguest=yes
  125. ; you want to check which services you offer everyone
  126. ; out there, by enabling them in the default context (see below).
  127. ;match_auth_username=yes ; if available, match user entry using the
  128. ; 'username' field from the authentication line
  129. ; instead of the From: field.
  130. allowoverlap = no ; Disable overlap dialing support. (Default is yes)
  131. ;allowoverlap=yes ; Enable RFC3578 overlap dialing support.
  132. ; Can use the Incomplete application to collect the
  133. ; needed digits from an ambiguous dialplan match.
  134. ;allowoverlap=dtmf ; Enable overlap dialing support using DTMF delivery
  135. ; methods (inband, RFC2833, SIP INFO) in the early
  136. ; media phase. Uses the Incomplete application to
  137. ; collect the needed digits.
  138. ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
  139. ; Default is enabled. The Dial() options 't' and 'T' are not
  140. ; related as to whether SIP transfers are allowed or not.
  141. ;realm=mydomain.tld ; Realm for digest authentication
  142. ; defaults to "asterisk". If you set a system name in
  143. ; asterisk.conf, it defaults to that system name
  144. ; Realms MUST be globally unique according to RFC 3261
  145. ; Set this to your host name or domain name
  146. ;domainsasrealm=no ; Use domains list as realms
  147. ; You can serve multiple Realms specifying several
  148. ; 'domain=...' directives (see below).
  149. ; In this case Realm will be based on request 'From'/'To' header
  150. ; and should match one of domain names.
  151. ; Otherwise default 'realm=...' will be used.
  152. ;recordonfeature=automixmon ; Default feature to use when receiving 'Record: on' header
  153. ; from an INFO message. Defaults to 'automon'. Works with
  154. ; dynamic features. Feature must be usable on requesting
  155. ; channel for it to work. Setting this value to a blank
  156. ; will disable it.
  157. ;recordofffeature=automixmon ; Default feature to use when receiving 'Record: off' 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.  
  163. ; With the current situation, you can do one of four things:
  164. ; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1
  165. ; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1
  166. ; c) Listen on the IPv4 wildcard. Example: bindaddr=0.0.0.0
  167. ; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=::
  168. ; (You can choose independently for UDP, TCP, and TLS, by specifying different values for
  169. ; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".)
  170. ; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat.
  171. ; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)
  172. ;
  173. ; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061
  174. ; for TLS).
  175. ; IPv4 example: bindaddr=0.0.0.0:5062
  176. ; IPv6 example: bindaddr=[::]:5062
  177. ;
  178. ; The address family of the bound UDP address is used to determine how Asterisk performs
  179. ; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only
  180. ; AAAA records are considered. In case d), both A and AAAA records are considered. Note,
  181. ; however, that Asterisk ignores all records except the first one. In case d), when both A
  182. ; and AAAA records are available, either an A or AAAA record will be first, and which one
  183. ; depends on the operating system. On systems using glibc, AAAA records are given
  184. ; priority.
  185.  
  186. udpbindaddr = 0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
  187. ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
  188.  
  189. ; When a dialog is started with another SIP endpoint, the other endpoint
  190. ; should include an Allow header telling us what SIP methods the endpoint
  191. ; implements. However, some endpoints either do not include an Allow header
  192. ; or lie about what methods they implement. In the former case, Asterisk
  193. ; makes the assumption that the endpoint supports all known SIP methods.
  194. ; If you know that your SIP endpoint does not provide support for a specific
  195. ; method, then you may provide a comma-separated list of methods that your
  196. ; endpoint does not implement in the disallowed_methods option. Note that
  197. ; if your endpoint is truthful with its Allow header, then there is no need
  198. ; to set this option. This option may be set in the general section or may
  199. ; be set per endpoint. If this option is set both in the general section and
  200. ; in a peer section, then the peer setting completely overrides the general
  201. ; setting (i.e. the result is *not* the union of the two options).
  202. ;
  203. ; Note also that while Asterisk currently will parse an Allow header to learn
  204. ; what methods an endpoint supports, the only actual use for this currently
  205. ; is for determining if Asterisk may send connected line UPDATE requests and
  206. ; MESSAGE requests. Its use may be expanded in the future.
  207. ;
  208. ; disallowed_methods = UPDATE
  209.  
  210. ;
  211. ; Note that the TCP and TLS support for chan_sip is currently considered
  212. ; experimental. Since it is new, all of the related configuration options are
  213. ; subject to change in any release. If they are changed, the changes will
  214. ; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
  215. ;
  216. tcpenable = no ; Enable server for incoming TCP connections (default is no)
  217. tcpbindaddr = 0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
  218. ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
  219.  
  220. ;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
  221. ;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
  222. ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
  223. ; Remember that the IP address must match the common name (hostname) in the
  224. ; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
  225. ; For details how to construct a certificate for SIP see
  226. ; http://tools.ietf.org/html/draft-ietf-sip-domain-certs
  227.  
  228. ;tcpauthtimeout = 30 ; tcpauthtimeout specifies the maximum number
  229. ; of seconds a client has to authenticate. If
  230. ; the client does not authenticate beofre this
  231. ; timeout expires, the client will be
  232. ; disconnected. (default: 30 seconds)
  233.  
  234. ;tcpauthlimit = 100 ; tcpauthlimit specifies the maximum number of
  235. ; unauthenticated sessions that will be allowed
  236. ; to connect at any given time. (default: 100)
  237.  
  238. transport = udp ; Set the default transports. The order determines the primary default transport.
  239. ; If tcpenable=no and the transport set is tcp, we will fallback to UDP.
  240.  
  241. srvlookup = yes ; Enable DNS SRV lookups on outbound calls
  242. subscribecontext = default
  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. ;
  321. ; This option specifies a preference for which music on hold class this channel
  322. ; should listen to when put on hold if the music class has not been set on the
  323. ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
  324. ; channel putting this one on hold did not suggest a music class.
  325. ;
  326. ; This option may be specified globally, or on a per-user or per-peer basis.
  327. ;
  328. ;mohinterpret=default
  329. ;
  330. ; This option specifies which music on hold class to suggest to the peer channel
  331. ; when this channel places the peer on hold. It may be specified globally or on
  332. ; a per-user or per-peer basis.
  333. ;
  334. ;mohsuggest=default
  335. ;
  336. ;parkinglot=plaza ; Sets the default parking lot for call parking
  337. ; This may also be set for individual users/peers
  338. ; Parkinglots are configured in features.conf
  339. language=es ; Default language setting for all users/peers
  340. ; This may also be set for individual users/peers
  341. ;tonezone=se ; Default tonezone for all users/peers
  342. ; This may also be set for individual users/peers
  343.  
  344. ;relaxdtmf=yes ; Relax dtmf handling
  345. ;trustrpid = no ; If Remote-Party-ID should be trusted
  346. ;sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no)
  347. ;sendrpid = rpid ; Use the "Remote-Party-ID" header
  348. ; to send the identity of the remote party
  349. ; This is identical to sendrpid=yes
  350. ;sendrpid = pai ; Use the "P-Asserted-Identity" header
  351. ; to send the identity of the remote party
  352. ;rpid_update = no ; In certain cases, the only method by which a connected line
  353. ; change may be immediately transmitted is with a SIP UPDATE request.
  354. ; If communicating with another Asterisk server, and you wish to be able
  355. ; transmit such UPDATE messages to it, then you must enable this option.
  356. ; Otherwise, we will have to wait until we can send a reinvite to
  357. ; transmit the information.
  358. ;prematuremedia=no ; Some ISDN links send empty media frames before
  359. ; the call is in ringing or progress state. The SIP
  360. ; channel will then send 183 indicating early media
  361. ; which will be empty - thus users get no ring signal.
  362. ; Setting this to "yes" will stop any media before we have
  363. ; call progress (meaning the SIP channel will not send 183 Session
  364. ; Progress for early media). Default is "yes". Also make sure that
  365. ; the SIP peer is configured with progressinband=never.
  366. ;
  367. ; In order for "noanswer" applications to work, you need to run
  368. ; the progress() application in the priority before the app.
  369.  
  370. ;progressinband=never ; If we should generate in-band ringing always
  371. ; use 'never' to never use in-band signalling, even in cases
  372. ; where some buggy devices might not render it
  373. ; Valid values: yes, no, never Default: never
  374. ;useragent=Asterisk PBX ; Allows you to change the user agent string
  375. ; The default user agent string also contains the Asterisk
  376. ; version. If you don't want to expose this, change the
  377. ; useragent string.
  378. ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
  379. ; Note that promiscredir when redirects are made to the
  380. ; local system will cause loops since Asterisk is incapable
  381. ; of performing a "hairpin" call.
  382. ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
  383. ; a valid phone number
  384. ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
  385. ; Other options:
  386. ; info : SIP INFO messages (application/dtmf-relay)
  387. ; shortinfo : SIP INFO messages (application/dtmf)
  388. ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
  389. ; auto : Use rfc2833 if offered, inband otherwise
  390.  
  391. ;compactheaders = yes ; send compact sip headers.
  392. ;
  393. ;videosupport=yes ; Turn on support for SIP video. You need to turn this
  394. ; on in this section to get any video support at all.
  395. ; You can turn it off on a per peer basis if the general
  396. ; video support is enabled, but you can't enable it for
  397. ; one peer only without enabling in the general section.
  398. ; If you set videosupport to "always", then RTP ports will
  399. ; always be set up for video, even on clients that don't
  400. ; support it. This assists callfile-derived calls and
  401. ; certain transferred calls to use always use video when
  402. ; available. [yes|NO|always]
  403.  
  404. ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
  405. ; Videosupport and maxcallbitrate is settable
  406. ; for peers and users as well
  407. ;callevents=no ; generate manager events when sip ua
  408. ; performs events (e.g. hold)
  409. ;authfailureevents=no ; generate manager "peerstatus" events when peer can't
  410. ; authenticate with Asterisk. Peerstatus will be "rejected".
  411. ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
  412. ; for any reason, always reject with an identical response
  413. ; equivalent to valid username and invalid password/hash
  414. ; instead of letting the requester know whether there was
  415. ; a matching user or peer for their request. This reduces
  416. ; the ability of an attacker to scan for valid SIP usernames.
  417. ; This option is set to "yes" by default.
  418.  
  419. ;auth_options_requests = yes ; Enabling this option will authenticate OPTIONS requests just like
  420. ; INVITE requests are. By default this option is disabled.
  421.  
  422. ;accept_outofcall_message = no ; Disable this option to reject all MESSAGE requests outside of a
  423. ; call. By default, this option is enabled. When enabled, MESSAGE
  424. ; requests are passed in to the dialplan.
  425.  
  426. ;outofcall_message_context = messages ; Context all out of dialog msgs are sent to. When this
  427. ; option is not set, the context used during peer matching
  428. ; is used. This option can be defined at both the peer and
  429. ; global level.
  430.  
  431. ;auth_message_requests = yes ; Enabling this option will authenticate MESSAGE requests.
  432. ; By default this option is enabled. However, it can be disabled
  433. ; should an application desire to not load the Asterisk server with
  434. ; doing authentication and implement end to end security in the
  435. ; message body.
  436.  
  437. ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
  438. ; order instead of RFC3551 packing order (this is required
  439. ; for Sipura and Grandstream ATAs, among others). This is
  440. ; contrary to the RFC3551 specification, the peer _should_
  441. ; be negotiating AAL2-G726-32 instead :-(
  442. ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
  443. ;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
  444. ;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
  445. ;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls
  446. ;outboundproxy=192.0.2.1 ; IPv4 address literal (default port is 5060)
  447. ;outboundproxy=2001:db8::1 ; IPv6 address literal (default port is 5060)
  448. ;outboundproxy=192.168.0.2.1:5062 ; IPv4 address literal with explicit port
  449. ;outboundproxy=[2001:db8::1]:5062 ; IPv6 address literal with explicit port
  450. ; ; (could also be tcp,udp) - defining transports on the proxy line only
  451. ; ; applies for the global proxy, otherwise use the transport= option
  452. ;matchexternaddrlocally = yes ; Only substitute the externaddr or externhost setting if it matches
  453. ; your localnet setting. Unless you have some sort of strange network
  454. ; setup you will not need to enable this.
  455.  
  456. ;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering
  457. ; as any IP address used for staticly defined
  458. ; hosts. This helps avoid the configuration
  459. ; error of allowing your users to register at
  460. ; the same address as a SIP provider.
  461.  
  462. ;contactdeny=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to
  463. ;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may
  464. ; register their phones.
  465. ;contactacl=named_acl_example ; Use named ACLs defined in acl.conf
  466.  
  467. ;engine=asterisk ; RTP engine to use when communicating with the device
  468.  
  469. ;
  470. ; If regcontext is specified, Asterisk will dynamically create and destroy a
  471. ; NoOp priority 1 extension for a given peer who registers or unregisters with
  472. ; us and have a "regexten=" configuration item.
  473. ; Multiple contexts may be specified by separating them with '&'. The
  474. ; actual extension is the 'regexten' parameter of the registering peer or its
  475. ; name if 'regexten' is not provided. If more than one context is provided,
  476. ; the context must be specified within regexten by appending the desired
  477. ; context after '@'. More than one regexten may be supplied if they are
  478. ; separated by '&'. Patterns may be used in regexten.
  479. ;
  480. ;regcontext=sipregistrations
  481. ;regextenonqualify=yes ; Default "no"
  482. ; If you have qualify on and the peer becomes unreachable
  483. ; this setting will enforce inactivation of the regexten
  484. ; extension for the peer
  485. ;legacy_useroption_parsing=yes ; Default "no" ; If you have this option enabled and there are semicolons
  486. ; in the user field of a sip URI, the field be truncated
  487. ; at the first semicolon seen. This effectively makes
  488. ; semicolon a non-usable character for peer names, extensions,
  489. ; and maybe other, less tested things. This can be useful
  490. ; for improving compatability with devices that like to use
  491. ; user options for whatever reason. The behavior is similar to
  492. ; how SIP URI's were typically handled in 1.6.2, hence the name.
  493.  
  494. ;send_diversion=no ; Default "yes" ; Asterisk normally sends Diversion headers with certain SIP
  495. ; invites to relay data about forwarded calls. If this option
  496. ; is disabled, Asterisk won't send Diversion headers unless
  497. ; they are added manually.
  498.  
  499. ; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
  500. ; in square brackets. For example, the caller id value 555.5555 becomes 5555555
  501. ; when this option is enabled. Disabling this option results in no modification
  502. ; of the caller id value, which is necessary when the caller id represents something
  503. ; that must be preserved. This option can only be used in the [general] section.
  504. ; By default this option is on.
  505. ;
  506. ;shrinkcallerid=yes ; on by default
  507.  
  508.  
  509. ;use_q850_reason = no ; Default "no"
  510. ; Set to yes add Reason header and use Reason header if it is available.
  511.  
  512. ; When the Transfer() application sends a REFER SIP message, extra headers specified in
  513. ; the dialplan by way of SIPAddHeader are sent out with that message. 1.8 and earlier did not
  514. ; add the extra headers. To revert to 1.8- behavior, call SIPRemoveHeader with no arguments
  515. ; before calling Transfer() to remove all additional headers from the channel. The setting
  516. ; below is for transitional compatibility only.
  517. ;
  518. ;refer_addheaders=yes ; on by default
  519.  
  520. ;autocreatepeer=no ; Allow any UAC not explicitly defined to register
  521. ; WITHOUT AUTHENTICATION. Enabling this options poses a high
  522. ; potential security risk and should be avoided unless the
  523. ; server is behind a trusted firewall.
  524. ; If set to "yes", then peers created in this fashion
  525. ; are purged during SIP reloads.
  526. ; When set to "persist", the peers created in this fashion
  527. ; are not purged during SIP reloads.
  528.  
  529. ;
  530. ;------------------------ TLS settings ------------------------------------------------------------
  531. ;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem format only) to use for TLS connections
  532. ; default is to look for "asterisk.pem" in current directory
  533.  
  534. ;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem format only) for TLS connections.
  535. ; If no tlsprivatekey is specified, tlscertfile is searched for
  536. ; for both public and private key.
  537.  
  538. ;tlscafile=</path/to/certificate>
  539. ; If the server your connecting to uses a self signed certificate
  540. ; you should have their certificate installed here so the code can
  541. ; verify the authenticity of their certificate.
  542.  
  543. ;tlscapath=</path/to/ca/dir>
  544. ; A directory full of CA certificates. The files must be named with
  545. ; the CA subject name hash value.
  546. ; (see man SSL_CTX_load_verify_locations for more info)
  547.  
  548. ;tlsdontverifyserver=[yes|no]
  549. ; If set to yes, don't verify the servers certificate when acting as
  550. ; a client. If you don't have the server's CA certificate you can
  551. ; set this and it will connect without requiring tlscafile to be set.
  552. ; Default is no.
  553.  
  554. ;tlscipher=<SSL cipher string>
  555. ; A string specifying which SSL ciphers to use or not use
  556. ; A list of valid SSL cipher strings can be found at:
  557. ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
  558. ;
  559. ;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2.
  560. ; Specify protocol for outbound client connections.
  561. ; If left unspecified, the default is sslv2.
  562. ;
  563. ;--------------------------- SIP timers ----------------------------------------------------
  564. ; These timers are used primarily in INVITE transactions.
  565. ; The default for Timer T1 is 500 ms or the measured run-trip time between
  566. ; Asterisk and the device if you have qualify=yes for the device.
  567. ;
  568. ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
  569. ; Defaults to 100 ms
  570. ;timert1=500 ; Default T1 timer
  571. ; Defaults to 500 ms or the measured round-trip
  572. ; time to a peer (qualify=yes).
  573. ;timerb=32000 ; Call setup timer. If a provisional response is not received
  574. ; in this amount of time, the call will autocongest
  575. ; Defaults to 64*timert1
  576.  
  577. ;--------------------------- RTP timers ----------------------------------------------------
  578. ; These timers are currently used for both audio and video streams. The RTP timeouts
  579. ; are only applied to the audio channel.
  580. ; The settings are settable in the global section as well as per device
  581. ;
  582. ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
  583. ; on the audio channel
  584. ; when we're not on hold. This is to be able to hangup
  585. ; a call in the case of a phone disappearing from the net,
  586. ; like a powerloss or grandma tripping over a cable.
  587. ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
  588. ; on the audio channel
  589. ; when we're on hold (must be > rtptimeout)
  590. ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
  591. ; (default is off - zero)
  592.  
  593. ;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
  594. ; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
  595. ; This mechanism can detect and reclaim SIP channels that do not terminate through normal
  596. ; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
  597. ; The operation of Session-Timers is driven by the following configuration parameters:
  598. ;
  599. ; * session-timers - Session-Timers feature operates in the following three modes:
  600. ; originate : Request and run session-timers always
  601. ; accept : Run session-timers only when requested by other UA
  602. ; refuse : Do not run session timers in any case
  603. ; The default mode of operation is 'accept'.
  604. ; * session-expires - Maximum session refresh interval in seconds. Defaults to 1800 secs.
  605. ; * session-minse - Minimum session refresh interval in seconds. Defualts to 90 secs.
  606. ; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
  607. ; uac - Default to the caller initially refreshing when possible
  608. ; uas - Default to the callee initially refreshing when possible
  609. ;
  610. ; Note that, due to recommendations in RFC 4028, Asterisk will always honor the other
  611. ; endpoint's preference for who will handle refreshes. Asterisk will never override the
  612. ; preferences of the other endpoint. Doing so could result in Asterisk and the endpoint
  613. ; fighting over who sends the refreshes. This holds true for the initiation of session
  614. ; timers and subsequent re-INVITE requests whether Asterisk is the caller or callee, or
  615. ; whether Asterisk is currently the refresher or not.
  616. ;
  617. ;session-timers=originate
  618. ;session-expires=600
  619. ;session-minse=90
  620. ;session-refresher=uac
  621. ;
  622. ;--------------------------- SIP DEBUGGING ---------------------------------------------------
  623. ;sipdebug = yes ; Turn on SIP debugging by default, from
  624. ; the moment the channel loads this configuration
  625. ;recordhistory=yes ; Record SIP history by default
  626. ; (see sip history / sip no history)
  627. ;dumphistory=yes ; Dump SIP history at end of SIP dialogue
  628. ; SIP history is output to the DEBUG logging channel
  629.  
  630.  
  631. ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
  632. ; You can subscribe to the status of extensions with a "hint" priority
  633. ; (See extensions.conf.sample for examples)
  634. ; chan_sip support two major formats for notifications: dialog-info and SIMPLE
  635. ;
  636. ; You will get more detailed reports (busy etc) if you have a call counter enabled
  637. ; for a device.
  638. ;
  639. ; If you set the busylevel, we will indicate busy when we have a number of calls that
  640. ; matches the busylevel treshold.
  641. ;
  642. ; For queues, you will need this level of detail in status reporting, regardless
  643. ; if you use SIP subscriptions. Queues and manager use the same internal interface
  644. ; for reading status information.
  645. ;
  646. ; Note: Subscriptions does not work if you have a realtime dialplan and use the
  647. ; realtime switch.
  648. ;
  649. ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
  650. ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
  651. ; Useful to limit subscriptions to local extensions
  652. ; Settable per peer/user also
  653. ;notifyringing = no ; Control whether subscriptions already INUSE get sent
  654. ; RINGING when another call is sent (default: yes)
  655. ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
  656. ; Turning on notifyringing and notifyhold will add a lot
  657. ; more database transactions if you are using realtime.
  658. ;notifycid = yes ; Control whether caller ID information is sent along with
  659. ; dialog-info+xml notifications (supported by snom phones).
  660. ; Note that this feature will only work properly when the
  661. ; incoming call is using the same extension and context that
  662. ; is being used as the hint for the called extension. This means
  663. ; that it won't work when using subscribecontext for your sip
  664. ; user or peer (if subscribecontext is different than context).
  665. ; This is also limited to a single caller, meaning that if an
  666. ; extension is ringing because multiple calls are incoming,
  667. ; only one will be used as the source of caller ID. Specify
  668. ; 'ignore-context' to ignore the called context when looking
  669. ; for the caller's channel. The default value is 'no.' Setting
  670. ; notifycid to 'ignore-context' also causes call-pickups attempted
  671. ; via SNOM's NOTIFY mechanism to set the context for the call pickup
  672. ; to PICKUPMARK.
  673. ;callcounter = yes ; Enable call counters on devices. This can be set per
  674. ; device too.
  675.  
  676. ;----------------------------------------- T.38 FAX SUPPORT ----------------------------------
  677. ;
  678. ; This setting is available in the [general] section as well as in device configurations.
  679. ; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off.
  680. ;
  681. ; t38pt_udptl = yes ; Enables T.38 with FEC error correction.
  682. ; t38pt_udptl = yes,fec ; Enables T.38 with FEC error correction.
  683. ; t38pt_udptl = yes,redundancy ; Enables T.38 with redundancy error correction.
  684. ; t38pt_udptl = yes,none ; Enables T.38 with no error correction.
  685. ;
  686. ; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that
  687. ; is based on an incorrect interpretation of the T.38 recommendation, and results in failures
  688. ; because Asterisk does not believe it can send T.38 packets of a reasonable size to that
  689. ; endpoint (Cisco media gateways are one example of this situation). In these cases, during a
  690. ; T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL
  691. ; stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you
  692. ; can set an override (globally, or on a per-device basis) to make Asterisk ignore the
  693. ; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead.
  694. ; This can be done by appending 'maxdatagram=<value>' to the t38pt_udptl configuration option,
  695. ; like this:
  696. ;
  697. ; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error correction and overrides
  698. ; ; the other endpoint's provided value to assume we can
  699. ; ; send 400 byte T.38 FAX packets to it.
  700. ;
  701. ; FAX detection will cause the SIP channel to jump to the 'fax' extension (if it exists)
  702. ; based one or more events being detected. The events that can be detected are an incoming
  703. ; CNG tone or an incoming T.38 re-INVITE request.
  704. ;
  705. ; faxdetect = yes ; Default 'no', 'yes' enables both CNG and T.38 detection
  706. ; faxdetect = cng ; Enables only CNG detection
  707. ; faxdetect = t38 ; Enables only T.38 detection
  708. ;
  709. ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
  710. ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
  711. ; Format for the register statement is:
  712. ; register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
  713. ;
  714. ;
  715. ;
  716. ; domain is either
  717. ; - domain in DNS
  718. ; - host name in DNS
  719. ; - the name of a peer defined below or in realtime
  720. ; The domain is where you register your username, so your SIP uri you are registering to
  721. ; is username@domain
  722. ;
  723. ; If no extension is given, the 's' extension is used. The extension needs to
  724. ; be defined in extensions.conf to be able to accept calls from this SIP proxy
  725. ; (provider).
  726. ;
  727. ; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
  728. ; this is equivalent to having the following line in the general section:
  729. ;
  730. ; register => username:secret@host/callbackextension
  731. ;
  732. ; and more readable because you don't have to write the parameters in two places
  733. ; (note that the "port" is ignored - this is a bug that should be fixed).
  734. ;
  735. ; Note that a register= line doesn't mean that we will match the incoming call in any
  736. ; other way than described above. If you want to control where the call enters your
  737. ; dialplan, which context, you want to define a peer with the hostname of the provider's
  738. ; server. If the provider has multiple servers to place calls to your system, you need
  739. ; a peer for each server.
  740. ;
  741. ; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may
  742. ; contain a port number. Since the logical separator between a host and port number is a
  743. ; ':' character, and this character is already used to separate between the optional "secret"
  744. ; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish
  745. ; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
  746. ; they are blank. See the third example below for an illustration.
  747. ;
  748. ;
  749. ; Examples:
  750. ;
  751. ;register => 1234:password@mysipprovider.com
  752. ;
  753. ; This will pass incoming calls to the 's' extension
  754. ;
  755. ;
  756. ;register => 2345:password@sip_proxy/1234
  757. ;
  758. ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
  759. ; connect to local extension 1234 in extensions.conf, default context,
  760. ; unless you configure a [sip_proxy] section below, and configure a
  761. ; context.
  762. ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
  763. ; Tip 2: Use separate inbound and outbound sections for SIP providers
  764. ; (instead of type=friend) if you have calls in both directions
  765. ;
  766. ;register => 3456@mydomain:5082::@mysipprovider.com
  767. ;
  768. ; Note that in this example, the optional authuser and secret portions have
  769. ; been left blank because we have specified a port in the user section
  770. ;
  771. ;register => tls://username:xxxxxx@sip-tls-proxy.example.org
  772. ;
  773. ; The 'transport' part defaults to 'udp' but may also be 'tcp', 'tls', 'ws', or 'wss'.
  774. ; Using 'udp://' explicitly is also useful in case the username part
  775. ; contains a '/' ('user/name').
  776.  
  777. ;registertimeout=20 ; retry registration calls every 20 seconds (default)
  778. ;registerattempts=10 ; Number of registration attempts before we give up
  779. ; 0 = continue forever, hammering the other server
  780. ; until it accepts the registration
  781. ; Default is 0 tries, continue forever
  782.  
  783. ;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
  784. ; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
  785. ; by other phones. At this time, you can only subscribe using UDP as the transport.
  786. ; Format for the mwi register statement is:
  787. ; mwi => user[:secret[:authuser]]@host[:port]/mailbox
  788. ;
  789. ; Examples:
  790. ;mwi => 1234:password@mysipprovider.com/1234
  791. ;mwi => 1234:password@myportprovider.com:6969/1234
  792. ;mwi => 1234:password:authuser@myauthprovider.com/1234
  793. ;mwi => 1234:password:authuser@myauthportprovider.com:6969/1234
  794. ;
  795. ; 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:
  796. ; mailbox=1234@SIP_Remote
  797. ;----------------------------------------- NAT SUPPORT ------------------------
  798. ;
  799. ; WARNING: SIP operation behind a NAT is tricky and you really need
  800. ; to read and understand well the following section.
  801. ;
  802. ; When Asterisk is behind a NAT device, the "local" address (and port) that
  803. ; a socket is bound to has different values when seen from the inside or
  804. ; from the outside of the NATted network. Unfortunately this address must
  805. ; be communicated to the outside (e.g. in SIP and SDP messages), and in
  806. ; order to determine the correct value Asterisk needs to know:
  807. ;
  808. ; + whether it is talking to someone "inside" or "outside" of the NATted network.
  809. ; This is configured by assigning the "localnet" parameter with a list
  810. ; of network addresses that are considered "inside" of the NATted network.
  811. ; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
  812. ; Multiple entries are allowed, e.g. a reasonable set is the following:
  813. ;
  814. ; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
  815. ; localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
  816. ; localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
  817. ; localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
  818. ;
  819. ; + the "externally visible" address and port number to be used when talking
  820. ; to a host outside the NAT. This information is derived by one of the
  821. ; following (mutually exclusive) config file parameters:
  822. ;
  823. ; a. "externaddr = hostname[:port]" specifies a static address[:port] to
  824. ; be used in SIP and SDP messages.
  825. ; The hostname is looked up only once, when [re]loading sip.conf .
  826. ; If a port number is not present, use the port specified in the "udpbindaddr"
  827. ; (which is not guaranteed to work correctly, because a NAT box might remap the
  828. ; port number as well as the address).
  829. ; This approach can be useful if you have a NAT device where you can
  830. ; configure the mapping statically. Examples:
  831. ;
  832. ; externaddr = 12.34.56.78 ; use this address.
  833. ; externaddr = 12.34.56.78:9900 ; use this address and port.
  834. ; externaddr = mynat.my.org:12600 ; Public address of my nat box.
  835. ; externtcpport = 9900 ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT.
  836. ; ; externtcpport will default to the externaddr or externhost port if either one is set.
  837. ; externtlsport = 12600 ; The externally mapped tls port, when Asterisk is behind a static NAT or PAT.
  838. ; ; externtlsport port will default to the RFC designated port of 5061.
  839. ;
  840. ; b. "externhost = hostname[:port]" is similar to "externaddr" except
  841. ; that the hostname is looked up every "externrefresh" seconds
  842. ; (default 10s). This can be useful when your NAT device lets you choose
  843. ; the port mapping, but the IP address is dynamic.
  844. ; Beware, you might suffer from service disruption when the name server
  845. ; resolution fails. Examples:
  846. ;
  847. ; externhost=foo.dyndns.net ; refreshed periodically
  848. ; externrefresh=180 ; change the refresh interval
  849. ;
  850. ; Note that at the moment all these mechanism work only for the SIP socket.
  851. ; The IP address discovered with externaddr/externhost is reused for
  852. ; media sessions as well, but the port numbers are not remapped so you
  853. ; may still experience problems.
  854. ;
  855. ; NOTE 1: in some cases, NAT boxes will use different port numbers in
  856. ; the internal<->external mapping. In these cases, the "externaddr" and
  857. ; "externhost" might not help you configure addresses properly.
  858. ;
  859. ; NOTE 2: when using "externaddr" or "externhost", the address part is
  860. ; also used as the external address for media sessions. Thus, the port
  861. ; information in the SDP may be wrong!
  862. ;
  863. ; In addition to the above, Asterisk has an additional "nat" parameter to
  864. ; address NAT-related issues in incoming SIP or media sessions.
  865. ; In particular, depending on the 'nat= ' settings described below, Asterisk
  866. ; may override the address/port information specified in the SIP/SDP messages,
  867. ; and use the information (sender address) supplied by the network stack instead.
  868. ; However, this is only useful if the external traffic can reach us.
  869. ; The following settings are allowed (both globally and in individual sections):
  870. ;
  871. ; nat = no ; Do no special NAT handling other than RFC3581
  872. ; nat = force_rport ; Pretend there was an rport parameter even if there wasn't
  873. ; nat = comedia ; Send media to the port Asterisk received it from regardless
  874. ; ; of where the SDP says to send it.
  875. ; nat = auto_force_rport ; Set the force_rport option if Asterisk detects NAT (default)
  876. ; nat = auto_comedia ; Set the comedia option if Asterisk detects NAT
  877. ;
  878. ; The nat settings can be combined. For example, to set both force_rport and comedia
  879. ; one would set nat=force_rport,comedia. If any of the comma-separated options is 'no',
  880. ; Asterisk will ignore any other settings and set nat=no. If one of the "auto" settings
  881. ; is used in conjunction with its non-auto counterpart (nat=comedia,auto_comedia), then
  882. ; the non-auto option will be ignored.
  883. ;
  884. ; The RFC 3581-defined 'rport' parameter allows a client to request that Asterisk send
  885. ; SIP responses to it via the source IP and port from which the request originated
  886. ; instead of the address/port listed in the top-most Via header. This is useful if a
  887. ; client knows that it is behind a NAT and therefore cannot guess from what address/port
  888. ; its request will be sent. Asterisk will always honor the 'rport' parameter if it is
  889. ; sent. The force_rport setting causes Asterisk to always send responses back to the
  890. ; address/port from which it received requests; even if the other side doesn't support
  891. ; adding the 'rport' parameter.
  892. ;
  893. ; 'comedia RTP handling' refers to the technique of sending RTP to the port that the
  894. ; the other endpoint's RTP arrived from, and means 'connection-oriented media'. This is
  895. ; only partially related to RFC 4145 which was referred to as COMEDIA while it was in
  896. ; draft form. This method is used to accomodate endpoints that may be located behind
  897. ; NAT devices, and as such the address/port they tell Asterisk to send RTP packets to
  898. ; for their media streams is not the actual address/port that will be used on the nearer
  899. ; side of the NAT.
  900. ;
  901. ; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
  902. ; the nat setting in a peer definition, then the peer username will be discoverable
  903. ; by outside parties as Asterisk will respond to different ports for defined and
  904. ; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
  905. ; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the
  906. ; other, then valid peers with settings differing from those in the general section will
  907. ; be discoverable.
  908. ;
  909. ; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by
  910. ; RFC 4961; Asterisk will always send RTP packets from the same port number it expects
  911. ; to receive them on.
  912. ;
  913. ; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using
  914. ; the media_address configuration option. This is only applicable to the general section and
  915. ; can not be set per-user or per-peer.
  916. ;
  917. ; media_address = 172.16.42.1
  918. ;
  919. ; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the
  920. ; perceived external network address has changed. When the stun_monitor is installed and
  921. ; configured, chan_sip will renew all outbound registrations when the monitor detects any sort
  922. ; of network change has occurred. By default this option is enabled, but only takes effect once
  923. ; res_stun_monitor is configured. If res_stun_monitor is enabled and you wish to not
  924. ; generate all outbound registrations on a network change, use the option below to disable
  925. ; this feature.
  926. ;
  927. ; subscribe_network_change_event = yes ; on by default
  928. ;
  929. ; ICE/STUN/TURN usage can be enabled globally or on a per-peer basis using the icesupport
  930. ; configuration option. When set to yes ICE support is enabled. When set to no it is disabled.
  931. ; It is disabled by default.
  932. ;
  933. ; icesupport = yes
  934.  
  935. ;----------------------------------- MEDIA HANDLING --------------------------------
  936. ; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
  937. ; no reason for Asterisk to stay in the media path, the media will be redirected.
  938. ; This does not really work well in the case where Asterisk is outside and the
  939. ; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
  940. ;
  941. ;directmedia=yes ; Asterisk by default tries to redirect the
  942. ; RTP media stream to go directly from
  943. ; the caller to the callee. Some devices do not
  944. ; support this (especially if one of them is behind a NAT).
  945. ; The default setting is YES. If you have all clients
  946. ; behind a NAT, or for some other reason want Asterisk to
  947. ; stay in the audio path, you may want to turn this off.
  948.  
  949. ; This setting also affect direct RTP
  950. ; at call setup (a new feature in 1.4 - setting up the
  951. ; call directly between the endpoints instead of sending
  952. ; a re-INVITE).
  953.  
  954. ; Additionally this option does not disable all reINVITE operations.
  955. ; It only controls Asterisk generating reINVITEs for the specific
  956. ; purpose of setting up a direct media path. If a reINVITE is
  957. ; needed to switch a media stream to inactive (when placed on
  958. ; hold) or to T.38, it will still be done, regardless of this
  959. ; setting. Note that direct T.38 is not supported.
  960.  
  961. ;directmedia=nonat ; An additional option is to allow media path redirection
  962. ; (reinvite) but only when the peer where the media is being
  963. ; sent is known to not be behind a NAT (as the RTP core can
  964. ; determine it based on the apparent IP address the media
  965. ; arrives from).
  966.  
  967. ;directmedia=update ; Yet a third option... use UPDATE for media path redirection,
  968. ; instead of INVITE. This can be combined with 'nonat', as
  969. ; 'directmedia=update,nonat'. It implies 'yes'.
  970.  
  971. ;directmedia=outgoing ; When sending directmedia reinvites, do not send an immediate
  972. ; reinvite on an incoming call leg. This option is useful when
  973. ; peered with another SIP user agent that is known to send
  974. ; immediate direct media reinvites upon call establishment. Setting
  975. ; the option in this situation helps to prevent potential glares.
  976. ; Setting this option implies 'yes'.
  977.  
  978. ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
  979. ; the call directly with media peer-2-peer without re-invites.
  980. ; Will not work for video and cases where the callee sends
  981. ; RTP payloads and fmtp headers in the 200 OK that does not match the
  982. ; callers INVITE. This will also fail if directmedia is enabled when
  983. ; the device is actually behind NAT.
  984.  
  985. ;directmediadeny=0.0.0.0/0 ; Use directmediapermit and directmediadeny to restrict
  986. ;directmediapermit=172.16.0.0/16; which peers should be able to pass directmedia to each other
  987. ; (There is no default setting, this is just an example)
  988. ; Use this if some of your phones are on IP addresses that
  989. ; can not reach each other directly. This way you can force
  990. ; RTP to always flow through asterisk in such cases.
  991. ;directmediaacl=acl_example ; Use named ACLs defined in acl.conf
  992.  
  993. ;ignoresdpversion=yes ; By default, Asterisk will honor the session version
  994. ; number in SDP packets and will only modify the SDP
  995. ; session if the version number changes. This option will
  996. ; force asterisk to ignore the SDP session version number
  997. ; and treat all SDP data as new data. This is required
  998. ; for devices that send us non standard SDP packets
  999. ; (observed with Microsoft OCS). By default this option is
  1000. ; off.
  1001.  
  1002. ;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
  1003. ; Like the useragent parameter, the default user agent string
  1004. ; also contains the Asterisk version.
  1005. ;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
  1006. ; This field MUST NOT contain spaces
  1007. ;encryption=no ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
  1008. ; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
  1009. ; the peer does not support SRTP. Defaults to no.
  1010. ;encryption_taglen=80 ; Set the auth tag length offered in the INVITE either 32/80 default 80
  1011. ;
  1012. ;avpf=yes ; Enable inter-operability with media streams using the AVPF RTP profile.
  1013. ; This will cause all offers and answers to use AVPF (or SAVPF). This
  1014. ; option may be specified at the global or peer scope.
  1015. ;----------------------------------------- REALTIME SUPPORT ------------------------
  1016. ; For additional information on ARA, the Asterisk Realtime Architecture,
  1017. ; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
  1018. ;
  1019. ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
  1020. ; just like friends added from the config file only on a
  1021. ; as-needed basis? (yes|no)
  1022.  
  1023. ;rtsavesysname=yes ; Save systemname in realtime database at registration
  1024. ; Default= no
  1025.  
  1026. ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
  1027. ; If set to yes, when a SIP UA registers successfully, the ip address,
  1028. ; the origination port, the registration period, and the username of
  1029. ; the UA will be set to database via realtime.
  1030. ; If not present, defaults to 'yes'. Note: realtime peers will
  1031. ; probably not function across reloads in the way that you expect, if
  1032. ; you turn this option off.
  1033. ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
  1034. ; as if it had just registered? (yes|no|<seconds>)
  1035. ; If set to yes, when the registration expires, the friend will
  1036. ; vanish from the configuration until requested again. If set
  1037. ; to an integer, friends expire within this number of seconds
  1038. ; instead of the registration interval.
  1039.  
  1040. ;ignoreregexpire=yes ; Enabling this setting has two functions:
  1041. ;
  1042. ; For non-realtime peers, when their registration expires, the
  1043. ; information will _not_ be removed from memory or the Asterisk database
  1044. ; if you attempt to place a call to the peer, the existing information
  1045. ; will be used in spite of it having expired
  1046. ;
  1047. ; For realtime peers, when the peer is retrieved from realtime storage,
  1048. ; the registration information will be used regardless of whether
  1049. ; it has expired or not; if it expires while the realtime peer
  1050. ; is still in memory (due to caching or other reasons), the
  1051. ; information will not be removed from realtime storage
  1052.  
  1053. ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
  1054. ; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
  1055. ; domains, each of which can direct the call to a specific context if desired.
  1056. ; By default, all domains are accepted and sent to the default context or the
  1057. ; context associated with the user/peer placing the call.
  1058. ; REGISTER to non-local domains will be automatically denied if a domain
  1059. ; list is configured.
  1060. ;
  1061. ; Domains can be specified using:
  1062. ; domain=<domain>[,<context>]
  1063. ; Examples:
  1064. ; domain=myasterisk.dom
  1065. ; domain=customer.com,customer-context
  1066. ;
  1067. ; In addition, all the 'default' domains associated with a server should be
  1068. ; added if incoming request filtering is desired.
  1069. ; autodomain=yes
  1070. ;
  1071. ; To disallow requests for domains not serviced by this server:
  1072. ; allowexternaldomains=no
  1073.  
  1074. ;domain=mydomain.tld,mydomain-incoming
  1075. ; Add domain and configure incoming context
  1076. ; for external calls to this domain
  1077. ;domain=1.2.3.4 ; Add IP address as local domain
  1078. ; You can have several "domain" settings
  1079. ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
  1080. ; Default is yes
  1081. ;autodomain=yes ; Turn this on to have Asterisk add local host
  1082. ; name and local IP to domain list.
  1083.  
  1084. ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
  1085. ; non-peers, use your primary domain "identity"
  1086. ; for From: headers instead of just your IP
  1087. ; address. This is to be polite and
  1088. ; it may be a mandatory requirement for some
  1089. ; destinations which do not have a prior
  1090. ; account relationship with your server.
  1091.  
  1092. ;------------------------------ Advice of Charge CONFIGURATION --------------------------
  1093. ; snom_aoc_enabled = yes; ; This options turns on and off support for sending AOC-D and
  1094. ; AOC-E to snom endpoints. This option can be used both in the
  1095. ; peer and global scope. The default for this option is off.
  1096.  
  1097.  
  1098. ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
  1099. ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
  1100. ; SIP channel. Defaults to "no". An enabled jitterbuffer will
  1101. ; be used only if the sending side can create and the receiving
  1102. ; side can not accept jitter. The SIP channel can accept jitter,
  1103. ; thus a jitterbuffer on the receive SIP side will be used only
  1104. ; if it is forced and enabled.
  1105.  
  1106. ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
  1107. ; channel. Defaults to "no".
  1108.  
  1109. ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
  1110.  
  1111. ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
  1112. ; resynchronized. Useful to improve the quality of the voice, with
  1113. ; big jumps in/broken timestamps, usually sent from exotic devices
  1114. ; and programs. Defaults to 1000.
  1115.  
  1116. ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
  1117. ; channel. Two implementations are currently available - "fixed"
  1118. ; (with size always equals to jbmaxsize) and "adaptive" (with
  1119. ; variable size, actually the new jb of IAX2). Defaults to fixed.
  1120.  
  1121. ; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
  1122. ; The option represents the number of milliseconds by which the new jitter buffer
  1123. ; will pad its size. the default is 40, so without modification, the new
  1124. ; jitter buffer will set its size to the jitter value plus 40 milliseconds.
  1125. ; increasing this value may help if your network normally has low jitter,
  1126. ; but occasionally has spikes.
  1127.  
  1128. ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
  1129.  
  1130. ;-----------------------------------------------------------------------------------
  1131.  
  1132. [authentication]
  1133. ; Global credentials for outbound calls, i.e. when a proxy challenges your
  1134. ; Asterisk server for authentication. These credentials override
  1135. ; any credentials in peer/register definition if realm is matched.
  1136. ;
  1137. ; This way, Asterisk can authenticate for outbound calls to other
  1138. ; realms. We match realm on the proxy challenge and pick an set of
  1139. ; credentials from this list
  1140. ; Syntax:
  1141. ; auth = <user>:<secret>@<realm>
  1142. ; auth = <user>#<md5secret>@<realm>
  1143. ; Example:
  1144. ;auth=mark:topsecret@digium.com
  1145. ;
  1146. ; You may also add auth= statements to [peer] definitions
  1147. ; Peer auth= override all other authentication settings if we match on realm
  1148.  
  1149. ;------------------------------------------------------------------------------
  1150. ; DEVICE CONFIGURATION
  1151. ;
  1152. ; SIP entities have a 'type' which determines their roles within Asterisk.
  1153. ; * For entities with 'type=peer':
  1154. ; Peers handle both inbound and outbound calls and are matched by ip/port, so for
  1155. ; The case of incoming calls from the peer, the IP address must match in order for
  1156. ; The invitation to work. This means calls made from either direction won't work if
  1157. ; The peer is unregistered while host=dynamic or if the host is otherise not set to
  1158. ; the correct IP of the sender.
  1159. ; * For entities with 'type=user':
  1160. ; Asterisk users handle inbound calls only (meaning they call Asterisk, Asterisk can't
  1161. ; call them) and are matched by their authorization information (authname and secret).
  1162. ; Asterisk doesn't rely on their IP and will accept calls regardless of the host setting
  1163. ; as long as the incoming SIP invite authorizes successfully.
  1164. ; * For entities with 'type=friend':
  1165. ; Asterisk will create the entity as both a friend and a peer. Asterisk will accept
  1166. ; calls from friends like it would for users, requiring only that the authorization
  1167. ; matches rather than the IP address. Since it is also a peer, a friend entity can
  1168. ; be called as long as its IP is known to Asterisk. In the case of host=dynamic,
  1169. ; this means it is necessary for the entity to register before Asterisk can call it.
  1170. ;
  1171. ; Use remotesecret for outbound authentication, and secret for authenticating
  1172. ; inbound requests. For historical reasons, if no remotesecret is supplied for an
  1173. ; outbound registration or call, the secret will be used.
  1174. ;
  1175. ; For device names, we recommend using only a-z, numerics (0-9) and underscore
  1176. ;
  1177. ; For local phones, type=friend works most of the time
  1178. ;
  1179. ; If you have one-way audio, you probably have NAT problems.
  1180. ; If Asterisk is on a public IP, and the phone is inside of a NAT device
  1181. ; you will need to configure nat option for those phones.
  1182. ; Also, turn on qualify=yes to keep the nat session open
  1183. ;
  1184. ; Configuration options available
  1185. ; --------------------
  1186. ; context
  1187. ; callingpres
  1188. ; permit
  1189. ; deny
  1190. ; secret
  1191. ; md5secret
  1192. ; remotesecret
  1193. ; transport
  1194. ; dtmfmode
  1195. ; directmedia
  1196. ; nat
  1197. ; callgroup
  1198. ; pickupgroup
  1199. ; language
  1200. ; allow
  1201. ; disallow
  1202. ; insecure
  1203. ; trustrpid
  1204. ; progressinband
  1205. ; promiscredir
  1206. ; useclientcode
  1207. ; accountcode
  1208. ; setvar
  1209. ; callerid
  1210. ; amaflags
  1211. ; callcounter
  1212. ; busylevel
  1213. ; allowoverlap
  1214. ; allowsubscribe
  1215. ; allowtransfer
  1216. ; ignoresdpversion
  1217. ; subscribecontext
  1218. ; template
  1219. ; videosupport
  1220. ; maxcallbitrate
  1221. ; rfc2833compensate
  1222. ; mailbox
  1223. ; session-timers
  1224. ; session-expires
  1225. ; session-minse
  1226. ; session-refresher
  1227. ; t38pt_usertpsource
  1228. ; regexten
  1229. ; fromdomain
  1230. ; fromuser
  1231. ; host
  1232. ; port
  1233. ; qualify
  1234. ; keepalive
  1235. ; defaultip
  1236. ; defaultuser
  1237. ; rtptimeout
  1238. ; rtpholdtimeout
  1239. ; sendrpid
  1240. ; outboundproxy
  1241. ; rfc2833compensate
  1242. ; callbackextension
  1243. ; registertrying
  1244. ; timert1
  1245. ; timerb
  1246. ; qualifyfreq
  1247. ; t38pt_usertpsource
  1248. ; contactpermit ; Limit what a host may register as (a neat trick
  1249. ; contactdeny ; is to register at the same IP as a SIP provider,
  1250. ; contactacl ; then call oneself, and get redirected to that
  1251. ; ; same location).
  1252. ; directmediapermit
  1253. ; directmediadeny
  1254. ; directmediaacl
  1255. ; unsolicited_mailbox
  1256. ; use_q850_reason
  1257. ; maxforwards
  1258. ; encryption
  1259. ; description ; Used to provide a description of the peer in console output
  1260. ; dtlsenable
  1261. ; dtlsverify
  1262. ; dtlsrekey
  1263. ; dtlscertfile
  1264. ; dtlsprivatekey
  1265. ; dtlscipher
  1266. ; dtlscafile
  1267. ; dtlscapath
  1268. ; dtlssetup
  1269. ;
  1270.  
  1271. ;------------------------------------------------------------------------------
  1272. ; DTLS-SRTP CONFIGURATION
  1273. ;
  1274. ; DTLS-SRTP support is available if the underlying RTP engine in use supports it.
  1275. ;
  1276. ; dtlsenable = yes ; Enable or disable DTLS-SRTP support
  1277. ; dtlsverify = yes ; Verify that the provided peer certificate is valid
  1278. ; dtlsrekey = 60 ; Interval at which to renegotiate the TLS session and rekey the SRTP session
  1279. ; ; If this is not set or the value provided is 0 rekeying will be disabled
  1280. ; dtlscertfile = file ; Path to certificate file to present
  1281. ; dtlsprivatekey = file ; Path to private key for certificate file
  1282. ; dtlscipher = <SSL cipher string> ; Cipher to use for TLS negotiation
  1283. ; ; A list of valid SSL cipher strings can be found at:
  1284. ; ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
  1285. ; dtlscafile = file ; Path to certificate authority certificate
  1286. ; dtlscapath = path ; Path to a directory containing certificate authority certificates
  1287. ; dtlssetup = actpass ; Whether we are willing to accept connections, connect to the other party, or both.
  1288. ; ; Valid options are active (we want to connect to the other party), passive (we want to
  1289. ; ; accept connections only), and actpass (we will do both). This value will be used in
  1290. ; ; the outgoing SDP when offering and for incoming SDP offers when the remote party sends
  1291. ; ; actpass
  1292.  
  1293. ;[sip_proxy]
  1294. ; For incoming calls only. Example: FWD (Free World Dialup)
  1295. ; We match on IP address of the proxy for incoming calls
  1296. ; since we can not match on username (caller id)
  1297. ;type=peer
  1298. ;context=from-fwd
  1299. ;host=fwd.pulver.com
  1300.  
  1301. ;[sip_proxy-out]
  1302. ;type=peer ; we only want to call out, not be called
  1303. ;remotesecret=guessit ; Our password to their service
  1304. ;defaultuser=yourusername ; Authentication user for outbound proxies
  1305. ;fromuser=yourusername ; Many SIP providers require this!
  1306. ;fromdomain=provider.sip.domain
  1307. ;host=box.provider.com
  1308. ;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will
  1309. ; ; accept both tcp and udp. The default transport type is only used for
  1310. ; ; outbound messages until a Registration takes place. During the
  1311. ; ; peer Registration the transport type may change to another supported
  1312. ; ; type if the peer requests so.
  1313.  
  1314. ;usereqphone=yes ; This provider requires ";user=phone" on URI
  1315. ;callcounter=yes ; Enable call counter
  1316. ;busylevel=2 ; Signal busy at 2 or more calls
  1317. ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
  1318. ;port=80 ; The port number we want to connect to on the remote side
  1319. ; Also used as "defaultport" in combination with "defaultip" settings
  1320.  
  1321. ;--- sample definition for a provider
  1322. ;[provider1]
  1323. ;type=peer
  1324. ;host=sip.provider1.com
  1325. ;fromuser=4015552299 ; how your provider knows you
  1326. ;remotesecret=youwillneverguessit ; The password we use to authenticate to them
  1327. ;secret=gissadetdu ; The password they use to contact us
  1328. ;callbackextension=123 ; Register with this server and require calls coming back to this extension
  1329. ;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
  1330. ; ; accept both tcp and udp. Default is udp. The first transport
  1331. ; ; listed will always be used for outgoing connections.
  1332. ;unsolicited_mailbox=4015552299 ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old
  1333. ; ; message count will be stored in the configured virtual mailbox. It can be used
  1334. ; ; by any device supporting MWI by specifying <configured value>@SIP_Remote as the
  1335. ; ; mailbox.
  1336.  
  1337. ;
  1338. ; Because you might have a large number of similar sections, it is generally
  1339. ; convenient to use templates for the common parameters, and add them
  1340. ; the the various sections. Examples are below, and we can even leave
  1341. ; the templates uncommented as they will not harm:
  1342.  
  1343. [basic-options](!); a template
  1344. dtmfmode = rfc2833
  1345. context = from-office
  1346. type = friend
  1347.  
  1348. [natted-phone](!,basic-options); another template inheriting basic-options
  1349. directmedia = no
  1350. host = dynamic
  1351.  
  1352. [public-phone](!,basic-options); another template inheriting basic-options
  1353. directmedia = yes
  1354.  
  1355. [my-codecs](!); a template for my preferred codecs
  1356. disallow = all
  1357. allow = ilbc
  1358. allow = g729
  1359. allow = gsm
  1360. allow = g723
  1361. allow = ulaw
  1362. ; Or, more simply:
  1363. ;allow=!all,ilbc,g729,gsm,g723,ulaw
  1364.  
  1365. [ulaw-phone](!); and another one for ulaw-only
  1366. disallow = all
  1367. allow = ulaw
  1368. ; Again, more simply:
  1369. ;allow=!all,ulaw
  1370.  
  1371. ; and finally instantiate a few phones
  1372. ;
  1373. ; [2133](natted-phone,my-codecs)
  1374. ; secret = peekaboo
  1375. ; [2134](natted-phone,ulaw-phone)
  1376. ; secret = not_very_secret
  1377. ; [2136](public-phone,ulaw-phone)
  1378. ; secret = not_very_secret_either
  1379. ; ...
  1380. ;
  1381.  
  1382. ; Standard configurations not using templates look like this:
  1383. ;
  1384. ;[grandstream1]
  1385. ;type=friend
  1386. ;context=from-sip ; Where to start in the dialplan when this phone calls
  1387. ;recordonfeature=dynamicfeature1 ; Feature to use when INFO with Record: on is received.
  1388. ;recordofffeature=dynamicfeature2 ; Feature to use when INFO with Record: off is received.
  1389. ;callerid=John Doe <1234> ; Full caller ID, to override the phones config
  1390. ; on incoming calls to Asterisk
  1391. ;description=Courtesy Phone ; Description of the peer. Shown when doing 'sip show peers'.
  1392. ;host=192.168.0.23 ; we have a static but private IP address
  1393. ; No registration allowed
  1394. ;directmedia=yes ; allow RTP voice traffic to bypass Asterisk
  1395. ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
  1396. ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
  1397. ; from the phone to asterisk (deprecated)
  1398. ; 1 for the explicit peer, 1 for the explicit user,
  1399. ; remember that a friend equals 1 peer and 1 user in
  1400. ; memory
  1401. ; There is no combined call counter for a "friend"
  1402. ; so there's currently no way in sip.conf to limit
  1403. ; to one inbound or outbound call per phone. Use
  1404. ; the group counters in the dial plan for that.
  1405. ;
  1406. ;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
  1407. ;disallow=all ; need to disallow=all before we can use allow=
  1408. ;allow=ulaw ; Note: In user sections the order of codecs
  1409. ; listed with allow= does NOT matter!
  1410. ;allow=alaw
  1411. ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
  1412. ;allow=g729 ; Pass-thru only unless g729 license obtained
  1413. ;callingpres=allowed_passed_screen ; Set caller ID presentation
  1414. ; See README.callingpres for more information
  1415.  
  1416. ;[xlite1]
  1417. ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
  1418. ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
  1419. ;type=friend
  1420. ;regexten=1234 ; When they register, create extension 1234
  1421. ;callerid="Jane Smith" <5678>
  1422. ;host=dynamic ; This device needs to register
  1423. ;directmedia=no ; Typically set to NO if behind NAT
  1424. ;disallow=all
  1425. ;allow=gsm ; GSM consumes far less bandwidth than ulaw
  1426. ;allow=ulaw
  1427. ;allow=alaw
  1428. ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
  1429. ;registertrying=yes ; Send a 100 Trying when the device registers.
  1430.  
  1431. ;[snom]
  1432. ;type=friend ; Friends place calls and receive calls
  1433. ;context=from-sip ; Context for incoming calls from this user
  1434. ;secret=blah
  1435. ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
  1436. ;language=de ; Use German prompts for this user
  1437. ;host=dynamic ; This peer register with us
  1438. ;dtmfmode=inband ; Choices are inband, rfc2833, or info
  1439. ;defaultip=192.168.0.59 ; IP used until peer registers
  1440. ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
  1441. ;subscribemwi=yes ; Only send notifications if this phone
  1442. ; subscribes for mailbox notification
  1443. ;vmexten=voicemail ; dialplan extension to reach mailbox
  1444. ; sets the Message-Account in the MWI notify message
  1445. ; defaults to global vmexten which defaults to "asterisk"
  1446. ;disallow=all
  1447. ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
  1448.  
  1449.  
  1450. ;[polycom]
  1451. ;type=friend ; Friends place calls and receive calls
  1452. ;context=from-sip ; Context for incoming calls from this user
  1453. ;secret=blahpoly
  1454. ;host=dynamic ; This peer register with us
  1455. ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
  1456. ;defaultuser=polly ; Username to use in INVITE until peer registers
  1457. ;defaultip=192.168.40.123
  1458. ; Normally you do NOT need to set this parameter
  1459. ;disallow=all
  1460. ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
  1461. ;progressinband=no ; Polycom phones don't work properly with "never"
  1462.  
  1463.  
  1464. ;[pingtel]
  1465. ;type=friend
  1466. ;secret=blah
  1467. ;host=dynamic
  1468. ;insecure=port ; Allow matching of peer by IP address without
  1469. ; matching port number
  1470. ;insecure=invite ; Do not require authentication of incoming INVITEs
  1471. ;insecure=port,invite ; (both)
  1472. ;qualify=1000 ; Consider it down if it's 1 second to reply
  1473. ; Helps with NAT session
  1474. ; qualify=yes uses default value
  1475. ;qualifyfreq=60 ; Qualification: How often to check for the
  1476. ; host to be up in seconds
  1477. ; Set to low value if you use low timeout for
  1478. ; NAT of UDP sessions
  1479. ;
  1480. ; Call group and Pickup group should be in the range from 0 to 63
  1481. ;
  1482. ;callgroup=1,3-4 ; We are in caller groups 1,3,4
  1483. ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
  1484. ;namedcallgroup=engineering,sales,netgroup,protgroup ; We are in named call groups engineering,sales,netgroup,protgroup
  1485. ;namedpickupgroup=sales ; We can do call pick-p for named call group sales
  1486. ;defaultip=192.168.0.60 ; IP address to use if peer has not registered
  1487. ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
  1488. ;permit=192.168.0.60/255.255.255.0
  1489. ;permit=192.168.0.60/24 ; we can also use CIDR notation for subnet masks
  1490. ;permit=2001:db8::/32 ; IPv6 ACLs can be specified if desired. IPv6 ACLs
  1491. ; apply only to IPv6 addresses, and IPv4 ACLs apply
  1492. ; only to IPv4 addresses.
  1493. ;acl=named_acl_example ; Use named ACLs defined in acl.conf
  1494.  
  1495. ;[cisco1]
  1496. ;type=friend
  1497. ;secret=blah
  1498. ;qualify=200 ; Qualify peer is no more than 200ms away
  1499. ;host=dynamic ; This device registers with us
  1500. ;directmedia=no ; Asterisk by default tries to redirect the
  1501. ; RTP media stream (audio) to go directly from
  1502. ; the caller to the callee. Some devices do not
  1503. ; support this (especially if one of them is
  1504. ; behind a NAT).
  1505. ;defaultip=192.168.0.4 ; IP address to use until registration
  1506. ;defaultuser=goran ; Username to use when calling this device before registration
  1507. ; Normally you do NOT need to set this parameter
  1508. ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from or to this device
  1509. ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
  1510. ; cause the given audio file to
  1511. ; be played upon completion of
  1512. ; an attended transfer.
  1513.  
  1514. ;[pre14-asterisk]
  1515. ;type=friend
  1516. ;secret=digium
  1517. ;host=dynamic
  1518. ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
  1519. ; You must have this turned on or DTMF reception will work improperly.
  1520. ;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
  1521. ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
  1522. ; external IP address of the remote device. If port forwarding is done at the client side
  1523. ; then UDPTL will flow to the remote device.
  1524.  
  1525. [200]
  1526. username=200
  1527. type=friend
  1528. secret=200
  1529. callerid="Klaus Salazar" <200>
  1530. host=dynamic
  1531. nat=no
  1532. ;disallow=all
  1533. allow=ullaw
  1534. context=internal
  1535. mailbox=200@voicemail
  1536. callgroup=1
  1537. pickupgroup=1
  1538.  
  1539. [201]
  1540. username=200
  1541. type=friend
  1542. secret=201
  1543. callerid="Brian Salazar" <201>
  1544. host=dynamic
  1545. nat=no
  1546. ;disallow=all
  1547. allow=ullaw
  1548. context=internal
  1549. mailbox=201@voicemail
  1550. callgroup=1
  1551. pickupgroup=1
  1552.  
  1553.  
  1554. [202]
  1555. username=202
  1556. type=friend
  1557. secret=202
  1558. callerid="Eduardo Salazar" <202>
  1559. host=dynamic
  1560. nat=no
  1561. ;disallow=all
  1562. allow=ullaw
  1563. context=internal
  1564. mailbox=202@voicemail
  1565. callgroup=1
  1566. pickupgroup=1
Advertisement
Add Comment
Please, Sign In to add comment
Advertisement
Public Pastes
Advertisement
We use cookies for various purposes including analytics. By continuing to use Pastebin, you agree to our use of cookies as described in the Cookies Policy.  OK, I Understand
Not a member of Pastebin yet?
Sign Up, it unlocks many cool features!