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  1. ;
  2. ; SIP Configuration example for Asterisk
  3. ;
  4. ; Syntax for specifying a SIP device in extensions.conf is
  5. ; SIP/devicename where devicename is defined in a section below.
  6. ;
  7. ; You may also use
  8. ; SIP/username@domain to call any SIP user on the Internet
  9. ; (Don't forget to enable DNS SRV records if you want to use this)
  10. ;
  11. ; If you define a SIP proxy as a peer below, you may call
  12. ; SIP/proxyhostname/user or SIP/user@proxyhostname
  13. ; where the proxyhostname is defined in a section below
  14. ;
  15. ; Useful CLI commands to check peers/users:
  16. ;   sip show peers      Show all SIP peers (including friends)
  17. ;   sip show users      Show all SIP users (including friends)
  18. ;   sip show registry       Show status of hosts we register with
  19. ;
  20. ;   sip debug           Show all SIP messages
  21. ;
  22. ;   reload chan_sip.so      Reload configuration file
  23. ;               Active SIP peers will not be reconfigured
  24. ;
  25.  
  26. [general]
  27. context=default         ; Default context for incoming calls
  28. allowguest=yes          ; Allow or reject guest calls (default is yes)
  29. allowoverlap=no         ; Disable overlap dialing support. (Default is yes)
  30. ;allowtransfer=no       ; Disable all transfers (unless enabled in peers or users)
  31.                 ; Default is enabled
  32. ;realm=mydomain.tld     ; Realm for digest authentication
  33.                 ; defaults to "asterisk". If you set a system name in
  34.                 ; asterisk.conf, it defaults to that system name
  35.                 ; Realms MUST be globally unique according to RFC 3261
  36.                 ; Set this to your host name or domain name
  37. bindport=5060           ; UDP Port to bind to (SIP standard port is 5060)
  38.                 ; bindport is the local UDP port that Asterisk will listen on
  39. bindaddr=0.0.0.0        ; IP address to bind to (0.0.0.0 binds to all)
  40. srvlookup=yes           ; Enable DNS SRV lookups on outbound calls
  41.                 ; Note: Asterisk only uses the first host
  42.                 ; in SRV records
  43.                 ; Disabling DNS SRV lookups disables the
  44.                 ; ability to place SIP calls based on domain
  45.                 ; names to some other SIP users on the Internet
  46.                
  47. ;domain=mydomain.tld        ; Set default domain for this host
  48.                 ; If configured, Asterisk will only allow
  49.                 ; INVITE and REFER to non-local domains
  50.                 ; Use "sip show domains" to list local domains
  51. ;pedantic=yes           ; Enable checking of tags in headers,
  52.                 ; international character conversions in URIs
  53.                 ; and multiline formatted headers for strict
  54.                 ; SIP compatibility (defaults to "no")
  55.  
  56. ; See doc/ip-tos.txt for a description of these parameters.
  57. ;tos_sip=cs3                    ; Sets TOS for SIP packets.
  58. ;tos_audio=ef                   ; Sets TOS for RTP audio packets.
  59. ;tos_video=af41                 ; Sets TOS for RTP video packets.
  60.  
  61. ;maxexpiry=3600         ; Maximum allowed time of incoming registrations
  62.                 ; and subscriptions (seconds)
  63. ;minexpiry=60           ; Minimum length of registrations/subscriptions (default 60)
  64. ;defaultexpiry=120      ; Default length of incoming/outgoing registration
  65. ;t1min=100          ; Minimum roundtrip time for messages to monitored hosts
  66.                 ; Defaults to 100 ms
  67. ;notifymimetype=text/plain  ; Allow overriding of mime type in MWI NOTIFY
  68. ;checkmwi=10            ; Default time between mailbox checks for peers
  69. ;buggymwi=no            ; Cisco SIP firmware doesn't support the MWI RFC
  70.                 ; fully. Enable this option to not get error messages
  71.                 ; when sending MWI to phones with this bug.
  72. ;vmexten=voicemail      ; dialplan extension to reach mailbox sets the
  73.                 ; Message-Account in the MWI notify message
  74.                 ; defaults to "asterisk"
  75. ;disallow=all           ; First disallow all codecs
  76. ;allow=ulaw         ; Allow codecs in order of preference
  77. ;allow=ilbc         ; see doc/rtp-packetization for framing options
  78. ;
  79. ; This option specifies a preference for which music on hold class this channel
  80. ; should listen to when put on hold if the music class has not been set on the
  81. ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
  82. ; channel putting this one on hold did not suggest a music class.
  83. ;
  84. ; This option may be specified globally, or on a per-user or per-peer basis.
  85. ;
  86. ;mohinterpret=default
  87. ;
  88. ; This option specifies which music on hold class to suggest to the peer channel
  89. ; when this channel places the peer on hold. It may be specified globally or on
  90. ; a per-user or per-peer basis.
  91. ;
  92. ;mohsuggest=default
  93. ;
  94. ;language=en            ; Default language setting for all users/peers
  95.                 ; This may also be set for individual users/peers
  96. ;relaxdtmf=yes          ; Relax dtmf handling
  97. ;trustrpid = no         ; If Remote-Party-ID should be trusted
  98. ;sendrpid = yes         ; If Remote-Party-ID should be sent
  99. ;progressinband=never       ; If we should generate in-band ringing always
  100.                 ; use 'never' to never use in-band signalling, even in cases
  101.                 ; where some buggy devices might not render it
  102.                 ; Valid values: yes, no, never Default: never
  103. ;useragent=Asterisk PBX     ; Allows you to change the user agent string
  104. ;promiscredir = no          ; If yes, allows 302 or REDIR to non-local SIP address
  105.                             ; Note that promiscredir when redirects are made to the
  106.                             ; local system will cause loops since Asterisk is incapable
  107.                             ; of performing a "hairpin" call.
  108. ;usereqphone = no       ; If yes, ";user=phone" is added to uri that contains
  109.                 ; a valid phone number
  110. ;dtmfmode = rfc2833     ; Set default dtmfmode for sending DTMF. Default: rfc2833
  111.                 ; Other options:
  112.                 ; info : SIP INFO messages
  113.                 ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
  114.                 ; auto : Use rfc2833 if offered, inband otherwise
  115.  
  116. ;compactheaders = yes       ; send compact sip headers.
  117. ;
  118. ;videosupport=yes       ; Turn on support for SIP video. You need to turn this on
  119.                 ; in the this section to get any video support at all.
  120.                 ; You can turn it off on a per peer basis if the general
  121.                 ; video support is enabled, but you can't enable it for
  122.                 ; one peer only without enabling in the general section.
  123. ;maxcallbitrate=384     ; Maximum bitrate for video calls (default 384 kb/s)
  124.                 ; Videosupport and maxcallbitrate is settable
  125.                 ; for peers and users as well
  126. ;callevents=no          ; generate manager events when sip ua
  127.                 ; performs events (e.g. hold)
  128. ;alwaysauthreject = yes     ; When an incoming INVITE or REGISTER is to be rejected,
  129.                     ; for any reason, always reject with '401 Unauthorized'
  130.                 ; instead of letting the requester know whether there was
  131.                 ; a matching user or peer for their request
  132.  
  133. ;g726nonstandard = yes      ; If the peer negotiates G726-32 audio, use AAL2 packing
  134.                 ; order instead of RFC3551 packing order (this is required
  135.                 ; for Sipura and Grandstream ATAs, among others). This is
  136.                 ; contrary to the RFC3551 specification, the peer _should_
  137.                 ; be negotiating AAL2-G726-32 instead :-(
  138.  
  139. ;matchexterniplocally = yes     ; Only substitute the externip or externhost setting if it matches
  140.                                 ; your localnet setting. Unless you have some sort of strange network
  141.                                 ; setup you will not need to enable this.
  142.  
  143. ;
  144. ; If regcontext is specified, Asterisk will dynamically create and destroy a
  145. ; NoOp priority 1 extension for a given peer who registers or unregisters with
  146. ; us and have a "regexten=" configuration item.  
  147. ; Multiple contexts may be specified by separating them with '&'. The
  148. ; actual extension is the 'regexten' parameter of the registering peer or its
  149. ; name if 'regexten' is not provided.  If more than one context is provided,
  150. ; the context must be specified within regexten by appending the desired
  151. ; context after '@'.  More than one regexten may be supplied if they are
  152. ; separated by '&'.  Patterns may be used in regexten.
  153. ;
  154. ;regcontext=sipregistrations
  155. ;
  156. ;--------------------------- RTP timers ----------------------------------------------------
  157. ; These timers are currently used for both audio and video streams. The RTP timeouts
  158. ; are only applied to the audio channel.
  159. ; The settings are settable in the global section as well as per device
  160. ;
  161. ;rtptimeout=60          ; Terminate call if 60 seconds of no RTP or RTCP activity
  162.                 ; on the audio channel
  163.                 ; when we're not on hold. This is to be able to hangup
  164.                 ; a call in the case of a phone disappearing from the net,
  165.                 ; like a powerloss or grandma tripping over a cable.
  166. ;rtpholdtimeout=300     ; Terminate call if 300 seconds of no RTP or RTCP activity
  167.                 ; on the audio channel
  168.                 ; when we're on hold (must be > rtptimeout)
  169. ;rtpkeepalive=<secs>        ; Send keepalives in the RTP stream to keep NAT open
  170.                 ; (default is off - zero)
  171. ;--------------------------- SIP DEBUGGING ---------------------------------------------------
  172. ;sipdebug = yes         ; Turn on SIP debugging by default, from
  173.                 ; the moment the channel loads this configuration
  174. ;recordhistory=yes      ; Record SIP history by default
  175.                 ; (see sip history / sip no history)
  176. ;dumphistory=yes        ; Dump SIP history at end of SIP dialogue
  177.                 ; SIP history is output to the DEBUG logging channel
  178.  
  179.  
  180. ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
  181. ; You can subscribe to the status of extensions with a "hint" priority
  182. ; (See extensions.conf.sample for examples)
  183. ; chan_sip support two major formats for notifications: dialog-info and SIMPLE
  184. ;
  185. ; You will get more detailed reports (busy etc) if you have a call limit set
  186. ; for a device. When the call limit is filled, we will indicate busy. Note that
  187. ; you need at least 2 in order to be able to do attended transfers.
  188. ;
  189. ; For queues, you will need this level of detail in status reporting, regardless
  190. ; if you use SIP subscriptions. Queues and manager use the same internal interface
  191. ; for reading status information.
  192. ;
  193. ; Note: Subscriptions does not work if you have a realtime dialplan and use the
  194. ; realtime switch.
  195. ;
  196. ;allowsubscribe=no      ; Disable support for subscriptions. (Default is yes)
  197. ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
  198.                 ; Useful to limit subscriptions to local extensions
  199.                 ; Settable per peer/user also
  200. ;notifyringing = yes        ; Notify subscriptions on RINGING state (default: no)
  201. ;notifyhold = yes       ; Notify subscriptions on HOLD state (default: no)
  202.                 ; Turning on notifyringing and notifyhold will add a lot
  203.                 ; more database transactions if you are using realtime.
  204. ;limitonpeers = yes     ; Apply call limits on peers only. This will improve
  205.                 ; status notification when you are using type=friend
  206.                 ; Inbound calls, that really apply to the user part
  207.                 ; of a friend will now be added to and compared with
  208.                 ; the peer limit instead of applying two call limits,
  209.                 ; one for the peer and one for the user.
  210.                 ; "sip show inuse" will only show active calls on
  211.                 ; the peer side of a "type=friend" object if this
  212.                 ; setting is turned on.
  213.  
  214. ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
  215. ;
  216. ; This setting is available in the [general] section as well as in device configurations.
  217. ; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
  218. ; both parties have T38 support enabled in their Asterisk configuration
  219. ; This has to be enabled in the general section for all devices to work. You can then
  220. ; disable it on a per device basis.
  221. ;
  222. ; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.
  223. ;
  224. ; t38pt_udptl = yes            ; Default false
  225. ;
  226. ;----------------------------------------- OUTBOUND SIP REGISTRATIONS  ------------------------
  227. ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
  228. ; Format for the register statement is:
  229. ;       register => user[:secret[:authuser]]@host[:port][/extension]
  230. ;
  231. ; If no extension is given, the 's' extension is used. The extension needs to
  232. ; be defined in extensions.conf to be able to accept calls from this SIP proxy
  233. ; (provider).
  234. ;
  235. ; host is either a host name defined in DNS or the name of a section defined
  236. ; below.
  237. ;
  238. ; Examples:
  239. ;
  240. register => 463910:456039619300:463910@eugw.ast.voovox.com/222
  241. ;
  242. ;     This will pass incoming calls to the '222' extension
  243. ;
  244. ;
  245. ;register => 2345:password@sip_proxy/1234
  246. ;
  247. ;    Register 2345 at sip provider 'sip_proxy'.  Calls from this provider
  248. ;    connect to local extension 1234 in extensions.conf, default context,
  249. ;    unless you configure a [sip_proxy] section below, and configure a
  250. ;    context.
  251. ;    Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
  252. ;    Tip 2: Use separate type=peer and type=user sections for SIP providers
  253. ;           (instead of type=friend) if you have calls in both directions
  254.  
  255. registertimeout=0       ; retry registration calls every 20 seconds (default)
  256. ;registerattempts=10        ; Number of registration attempts before we give up
  257.                 ; 0 = continue forever, hammering the other server
  258.                 ; until it accepts the registration
  259.                 ; Default is 0 tries, continue forever
  260.  
  261. ;----------------------------------------- NAT SUPPORT ------------------------
  262. ; The externip, externhost and localnet settings are used if you use Asterisk
  263. ; behind a NAT device to communicate with services on the outside.
  264.  
  265. ;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP
  266.                 ; messages if we're behind a NAT
  267.  
  268.                 ; The externip and localnet is used
  269.                 ; when registering and communicating with other proxies
  270.                 ; that we're registered with
  271. externhost=sikanrong.dynalias.net   ; Alternatively you can specify an
  272.                 ; external host, and Asterisk will
  273.                 ; perform DNS queries periodically.  Not
  274.                 ; recommended for production
  275.                 ; environments!  Use externip instead
  276. externrefresh=10        ; How often to refresh externhost if
  277.                 ; used
  278.                 ; You may add multiple local networks.  A reasonable
  279.                 ; set of defaults are:
  280. ;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
  281. ;localnet=10.0.0.0/255.0.0.0    ; Also RFC1918
  282. ;localnet=172.16.0.0/12     ; Another RFC1918 with CIDR notation
  283. ;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
  284.  
  285. ; The nat= setting is used when Asterisk is on a public IP, communicating with
  286. ; devices hidden behind a NAT device (broadband router).  If you have one-way
  287. ; audio problems, you usually have problems with your NAT configuration or your
  288. ; firewall's support of SIP+RTP ports.  You configure Asterisk choice of RTP
  289. ; ports for incoming audio in rtp.conf
  290. ;
  291. ;nat=no             ; Global NAT settings  (Affects all peers and users)
  292.                                 ; yes = Always ignore info and assume NAT
  293.                                 ; no = Use NAT mode only according to RFC3581 (;rport)
  294.                                 ; never = Never attempt NAT mode or RFC3581 support
  295.                 ; route = Assume NAT, don't send rport
  296.                 ; (work around more UNIDEN bugs)
  297.  
  298. ;----------------------------------- MEDIA HANDLING --------------------------------
  299. ; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
  300. ; no reason for Asterisk to stay in the media path, the media will be redirected.
  301. ; This does not really work with in the case where Asterisk is outside and have
  302. ; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
  303. ;
  304. ;canreinvite=yes        ; Asterisk by default tries to redirect the
  305.                 ; RTP media stream (audio) to go directly from
  306.                 ; the caller to the callee.  Some devices do not
  307.                 ; support this (especially if one of them is behind a NAT).
  308.                 ; The default setting is YES. If you have all clients
  309.                 ; behind a NAT, or for some other reason wants Asterisk to
  310.                 ; stay in the audio path, you may want to turn this off.
  311.  
  312.                 ; In Asterisk 1.4 this setting also affect direct RTP
  313.                 ; at call setup (a new feature in 1.4 - setting up the
  314.                 ; call directly between the endpoints instead of sending
  315.                 ; a re-INVITE).
  316.  
  317. ;directrtpsetup=yes     ; Enable the new experimental direct RTP setup. This sets up
  318.                 ; the call directly with media peer-2-peer without re-invites.
  319.                 ; Will not work for video and cases where the callee sends
  320.                 ; RTP payloads and fmtp headers in the 200 OK that does not match the
  321.                 ; callers INVITE. This will also fail if canreinvite is enabled when
  322.                 ; the device is actually behind NAT.
  323.  
  324. ;canreinvite=nonat      ; An additional option is to allow media path redirection
  325.                 ; (reinvite) but only when the peer where the media is being
  326.                 ; sent is known to not be behind a NAT (as the RTP core can
  327.                 ; determine it based on the apparent IP address the media
  328.                 ; arrives from).
  329.  
  330. ;canreinvite=update     ; Yet a third option... use UPDATE for media path redirection,
  331.                 ; instead of INVITE. This can be combined with 'nonat', as
  332.                 ; 'canreinvite=update,nonat'. It implies 'yes'.
  333.  
  334. ;----------------------------------------- REALTIME SUPPORT ------------------------
  335. ; For additional information on ARA, the Asterisk Realtime Architecture,
  336. ; please read realtime.txt and extconfig.txt in the /doc directory of the
  337. ; source code.
  338. ;
  339. ;rtcachefriends=yes     ; Cache realtime friends by adding them to the internal list
  340.                 ; just like friends added from the config file only on a
  341.                 ; as-needed basis? (yes|no)
  342.  
  343. ;rtsavesysname=yes      ; Save systemname in realtime database at registration
  344.                 ; Default= no
  345.  
  346. ;rtupdate=yes           ; Send registry updates to database using realtime? (yes|no)
  347.                 ; If set to yes, when a SIP UA registers successfully, the ip address,
  348.                 ; the origination port, the registration period, and the username of
  349.                 ; the UA will be set to database via realtime.
  350.                 ; If not present, defaults to 'yes'.
  351. ;rtautoclear=yes        ; Auto-Expire friends created on the fly on the same schedule
  352.                 ; as if it had just registered? (yes|no|<seconds>)
  353.                 ; If set to yes, when the registration expires, the friend will
  354.                 ; vanish from the configuration until requested again. If set
  355.                 ; to an integer, friends expire within this number of seconds
  356.                 ; instead of the registration interval.
  357.  
  358. ;ignoreregexpire=yes        ; Enabling this setting has two functions:
  359.                 ;
  360.                 ; For non-realtime peers, when their registration expires, the
  361.                 ; information will _not_ be removed from memory or the Asterisk database
  362.                 ; if you attempt to place a call to the peer, the existing information
  363.                 ; will be used in spite of it having expired
  364.                 ;
  365.                 ; For realtime peers, when the peer is retrieved from realtime storage,
  366.                 ; the registration information will be used regardless of whether
  367.                 ; it has expired or not; if it expires while the realtime peer
  368.                 ; is still in memory (due to caching or other reasons), the
  369.                 ; information will not be removed from realtime storage
  370.  
  371. ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
  372. ; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
  373. ; domains, each of which can direct the call to a specific context if desired.
  374. ; By default, all domains are accepted and sent to the default context or the
  375. ; context associated with the user/peer placing the call.
  376. ; Domains can be specified using:
  377. ; domain=<domain>[,<context>]
  378. ; Examples:
  379. ; domain=myasterisk.dom
  380. ; domain=customer.com,customer-context
  381. ;
  382. ; In addition, all the 'default' domains associated with a server should be
  383. ; added if incoming request filtering is desired.
  384. ; autodomain=yes
  385. ;
  386. ; To disallow requests for domains not serviced by this server:
  387. ; allowexternaldomains=no
  388.  
  389. ;domain=mydomain.tld,mydomain-incoming
  390.                 ; Add domain and configure incoming context
  391.                 ; for external calls to this domain
  392. ;domain=1.2.3.4         ; Add IP address as local domain
  393.                 ; You can have several "domain" settings
  394. ;allowexternaldomains=no    ; Disable INVITE and REFER to non-local domains
  395.                 ; Default is yes
  396. ;autodomain=yes         ; Turn this on to have Asterisk add local host
  397.                 ; name and local IP to domain list.
  398.  
  399. ; fromdomain=mydomain.tld   ; When making outbound SIP INVITEs to
  400.                             ; non-peers, use your primary domain "identity"
  401.                             ; for From: headers instead of just your IP
  402.                             ; address. This is to be polite and
  403.                             ; it may be a mandatory requirement for some
  404.                             ; destinations which do not have a prior
  405.                             ; account relationship with your server.
  406.  
  407. ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
  408. ; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
  409.                               ; SIP channel. Defaults to "no". An enabled jitterbuffer will
  410.                               ; be used only if the sending side can create and the receiving
  411.                               ; side can not accept jitter. The SIP channel can accept jitter,
  412.                               ; thus a jitterbuffer on the receive SIP side will be used only
  413.                               ; if it is forced and enabled.
  414.  
  415. ; jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a SIP
  416.                               ; channel. Defaults to "no".
  417.  
  418. ; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.
  419.  
  420. ; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
  421.                               ; resynchronized. Useful to improve the quality of the voice, with
  422.                               ; big jumps in/broken timestamps, usually sent from exotic devices
  423.                               ; and programs. Defaults to 1000.
  424.  
  425. ; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a SIP
  426.                               ; channel. Two implementations are currently available - "fixed"
  427.                               ; (with size always equals to jbmaxsize) and "adaptive" (with
  428.                               ; variable size, actually the new jb of IAX2). Defaults to fixed.
  429.  
  430. ; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
  431. ;-----------------------------------------------------------------------------------
  432.  
  433. [authentication]
  434. ; Global credentials for outbound calls, i.e. when a proxy challenges your
  435. ; Asterisk server for authentication. These credentials override
  436. ; any credentials in peer/register definition if realm is matched.
  437. ;
  438. ; This way, Asterisk can authenticate for outbound calls to other
  439. ; realms. We match realm on the proxy challenge and pick an set of
  440. ; credentials from this list
  441. ; Syntax:
  442. ;       auth = 463910:456039619300@eugw.ast.voovox.com
  443. ;   auth = <user>#<md5secret>@<realm>
  444. ; Example:
  445. ;auth=mark:topsecret@digium.com
  446. ;
  447. ; You may also add auth= statements to [peer] definitions
  448. ; Peer auth= override all other authentication settings if we match on realm
  449.  
  450. ;------------------------------------------------------------------------------
  451. ; Users and peers have different settings available. Friends have all settings,
  452. ; since a friend is both a peer and a user
  453. ;
  454. ; User config options:        Peer configuration:
  455. ; --------------------        -------------------
  456. ; context                     context
  457. ; callingpres             callingpres
  458. ; permit                      permit
  459. ; deny                        deny
  460. ; secret                      secret
  461. ; md5secret                   md5secret
  462. ; dtmfmode                    dtmfmode
  463. ; canreinvite                 canreinvite
  464. ; nat                         nat
  465. ; callgroup                   callgroup
  466. ; pickupgroup                 pickupgroup
  467. ; language                    language
  468. ; allow                       allow
  469. ; disallow                    disallow
  470. ; insecure                    insecure
  471. ; trustrpid                   trustrpid
  472. ; progressinband              progressinband
  473. ; promiscredir                promiscredir
  474. ; useclientcode               useclientcode
  475. ; accountcode                 accountcode
  476. ; setvar                      setvar
  477. ; callerid            callerid
  478. ; amaflags            amaflags
  479. ; call-limit              call-limit
  480. ; allowoverlap            allowoverlap
  481. ; allowsubscribe          allowsubscribe
  482. ; allowtransfer               allowtransfer
  483. ; subscribecontext        subscribecontext
  484. ; videosupport            videosupport
  485. ; maxcallbitrate          maxcallbitrate
  486. ; rfc2833compensate           mailbox
  487. ; t38pt_usertpsource          username
  488. ;                             template
  489. ;                             fromdomain
  490. ;                             regexten
  491. ;                             fromuser
  492. ;                             host
  493. ;                             port
  494. ;                             qualify
  495. ;                             defaultip
  496. ;                             rtptimeout
  497. ;                             rtpholdtimeout
  498. ;                             sendrpid
  499. ;                             outboundproxy
  500. ;                             rfc2833compensate
  501. ;                             t38pt_usertpsource
  502.  
  503. ;[sip_proxy]
  504. ; For incoming calls only. Example: FWD (Free World Dialup)
  505. ; We match on IP address of the proxy for incoming calls
  506. ; since we can not match on username (caller id)
  507. ;type=peer
  508. ;context=from-fwd
  509. ;host=fwd.pulver.com
  510.  
  511. ;[sip_proxy-out]
  512. ;type=peer                      ; we only want to call out, not be called
  513. ;secret=guessit
  514. ;username=yourusername          ; Authentication user for outbound proxies
  515. ;fromuser=yourusername          ; Many SIP providers require this!
  516. ;fromdomain=provider.sip.domain
  517. ;host=box.provider.com
  518. ;usereqphone=yes            ; This provider requires ";user=phone" on URI
  519. ;call-limit=5               ; permit only 5 simultaneous outgoing calls to this peer
  520. ;outboundproxy=proxy.provider.domain    ; send outbound signaling to this proxy, not directly to the peer
  521.                     ; Call-limits will not be enforced on real-time peers,
  522.                     ; since they are not stored in-memory
  523. ;port=80                ; The port number we want to connect to on the remote side
  524.                     ; Also used as "defaultport" in combination with "defaultip" settings
  525.  
  526. ;------------------------------------------------------------------------------
  527. ; Definitions of locally connected SIP devices
  528. ;
  529. ; type = user   a device that authenticates to us by "from" field to place calls
  530. ; type = peer   a device we place calls to or that calls us and we match by host
  531. ; type = friend two configurations (peer+user) in one
  532. ;
  533. ; For device names, we recommend using only a-z, numerics (0-9) and underscore
  534. ;
  535. ; For local phones, type=friend works most of the time
  536. ;
  537. ; If you have one-way audio, you probably have NAT problems.
  538. ; If Asterisk is on a public IP, and the phone is inside of a NAT device
  539. ; you will need to configure nat option for those phones.
  540. ; Also, turn on qualify=yes to keep the nat session open
  541.  
  542. ;[grandstream1]
  543. ;type=friend           
  544. ;context=from-sip       ; Where to start in the dialplan when this phone calls
  545. ;callerid=John Doe <1234>   ; Full caller ID, to override the phones config
  546.                 ; on incoming calls to Asterisk
  547. ;host=192.168.0.23      ; we have a static but private IP address
  548.                 ; No registration allowed
  549. ;nat=no             ; there is not NAT between phone and Asterisk
  550. ;canreinvite=yes        ; allow RTP voice traffic to bypass Asterisk
  551. ;dtmfmode=info          ; either RFC2833 or INFO for the BudgeTone
  552. ;call-limit=1           ; permit only 1 outgoing call and 1 incoming call at a time
  553.                 ; from the phone to asterisk
  554.                 ; 1 for the explicit peer, 1 for the explicit user,
  555.                 ; remember that a friend equals 1 peer and 1 user in
  556.                 ; memory
  557.                 ; This will affect your subscriptions as well.
  558.                 ; There is no combined call counter for a "friend"
  559.                 ; so there's currently no way in sip.conf to limit
  560.                 ; to one inbound or outbound call per phone. Use
  561.                 ; the group counters in the dial plan for that.
  562.                 ;
  563. ;mailbox=1234@default       ; mailbox 1234 in voicemail context "default"
  564. ;disallow=all           ; need to disallow=all before we can use allow=
  565. ;allow=ulaw         ; Note: In user sections the order of codecs
  566.                 ; listed with allow= does NOT matter!
  567. ;allow=alaw
  568. ;allow=g723.1           ; Asterisk only supports g723.1 pass-thru!
  569. ;allow=g729         ; Pass-thru only unless g729 license obtained
  570. ;callingpres=allowed_passed_screen  ; Set caller ID presentation
  571.                 ; See doc/callingpres.txt for more information
  572.  
  573.  
  574. ;[xlite1]
  575. ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
  576. ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
  577. ;type=friend
  578. ;regexten=1234          ; When they register, create extension 1234
  579. ;callerid="Jane Smith" <5678>
  580. ;host=dynamic           ; This device needs to register
  581. ;nat=yes            ; X-Lite is behind a NAT router
  582. ;canreinvite=no         ; Typically set to NO if behind NAT
  583. ;disallow=all
  584. ;allow=gsm          ; GSM consumes far less bandwidth than ulaw
  585. ;allow=ulaw
  586. ;allow=alaw
  587. ;mailbox=1234@default,1233@default  ; Subscribe to status of multiple mailboxes
  588.  
  589.  
  590. ;[snom]
  591. ;type=friend            ; Friends place calls and receive calls
  592. ;context=from-sip       ; Context for incoming calls from this user
  593. ;secret=blah
  594. ;subscribecontext=localextensions   ; Only allow SUBSCRIBE for local extensions
  595. ;language=de            ; Use German prompts for this user
  596. ;host=dynamic           ; This peer register with us
  597. ;dtmfmode=inband        ; Choices are inband, rfc2833, or info
  598. ;defaultip=192.168.0.59     ; IP used until peer registers
  599. ;mailbox=1234@context,2345      ; Mailbox(-es) for message waiting indicator
  600. ;subscribemwi=yes       ; Only send notifications if this phone
  601.                 ; subscribes for mailbox notification
  602. ;vmexten=voicemail      ; dialplan extension to reach mailbox
  603.                 ; sets the Message-Account in the MWI notify message
  604.                 ; defaults to global vmexten which defaults to "asterisk"
  605. ;disallow=all
  606. ;allow=ulaw         ; dtmfmode=inband only works with ulaw or alaw!
  607.  
  608.  
  609. ;[polycom]
  610. ;type=friend            ; Friends place calls and receive calls
  611. ;context=from-sip       ; Context for incoming calls from this user
  612. ;secret=blahpoly
  613. ;host=dynamic           ; This peer register with us
  614. ;dtmfmode=rfc2833       ; Choices are inband, rfc2833, or info
  615. ;username=polly         ; Username to use in INVITE until peer registers
  616.                 ; Normally you do NOT need to set this parameter
  617. ;disallow=all
  618. ;allow=ulaw                     ; dtmfmode=inband only works with ulaw or alaw!
  619. ;progressinband=no      ; Polycom phones don't work properly with "never"
  620.  
  621.  
  622. ;[pingtel]
  623. ;type=friend
  624. ;secret=blah
  625. ;host=dynamic
  626. ;insecure=port          ; Allow matching of peer by IP address without
  627.                 ; matching port number
  628. ;insecure=invite        ; Do not require authentication of incoming INVITEs
  629. ;insecure=port,invite       ; (both)
  630. ;qualify=1000           ; Consider it down if it's 1 second to reply
  631.                 ; Helps with NAT session
  632.                 ; qualify=yes uses default value
  633. ;
  634. ; Call group and Pickup group should be in the range from 0 to 63
  635. ;
  636. ;callgroup=1,3-4        ; We are in caller groups 1,3,4
  637. ;pickupgroup=1,3-5      ; We can do call pick-p for call group 1,3,4,5
  638. ;defaultip=192.168.0.60     ; IP address to use if peer has not registered
  639. ;deny=0.0.0.0/0.0.0.0       ; ACL: Control access to this account based on IP address
  640. ;permit=192.168.0.60/255.255.255.0
  641.  
  642. ;[cisco1]
  643. ;type=friend
  644. ;secret=blah
  645. ;qualify=200            ; Qualify peer is no more than 200ms away
  646. ;nat=yes            ; This phone may be natted
  647.                 ; Send SIP and RTP to the IP address that packet is
  648.                 ; received from instead of trusting SIP headers
  649. ;host=dynamic           ; This device registers with us
  650. ;canreinvite=no         ; Asterisk by default tries to redirect the
  651.                 ; RTP media stream (audio) to go directly from
  652.                 ; the caller to the callee.  Some devices do not
  653.                 ; support this (especially if one of them is
  654.                 ; behind a NAT).
  655. ;defaultip=192.168.0.4      ; IP address to use until registration
  656. ;username=goran         ; Username to use when calling this device before registration
  657.                 ; Normally you do NOT need to set this parameter
  658. ;setvar=CUSTID=5678     ; Channel variable to be set for all calls from this device
  659.  
  660. ;[pre14-asterisk]
  661. ;type=friend
  662. ;secret=digium
  663. ;host=dynamic
  664. ;rfc2833compensate=yes      ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
  665.                 ; You must have this turned on or DTMF reception will work improperly.
  666. ;t38pt_usertpsource=yes         ; Use the source IP address of RTP as the destination IP address for UDPTL packets
  667.                                 ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
  668.                                 ; external IP address of the remote device. If port forwarding is done at the client side
  669.                                 ; then UDPTL will flow to the remote device.
  670.  
  671.  
  672. [1000]
  673. type=friend
  674. context=adhearsion
  675. host=dynamic
  676.  
  677. [2000]
  678. type=friend
  679. context=adhearsion
  680. host=dynamic
  681.  
  682. [voovox-outbound]
  683. type=friend
  684. username=463910
  685. fromuser=463910
  686. secret=xxxxxx
  687. context=adhearsion
  688. disallow=all
  689. allow=gsm
  690. allow=ilbc
  691. host=eugw.ast.voovox.com
  692. insecure=invite
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