{\rtf1\ansi\ansicpg1252\deff0\deflang1033{\fonttbl{\f0\fswiss\fcharset0 Arial;}} {\*\generator Msftedit 5.41.15.1515;}\viewkind4\uc1\pard\f0\fs20 ;\par ; SIP Configuration example for Asterisk\par ;\par ; Syntax for specifying a SIP device in extensions.conf is\par ; SIP/devicename where devicename is defined in a section below.\par ;\par ; You may also use\par ; SIP/username@domain to call any SIP user on the Internet\par ; (Don't forget to enable DNS SRV records if you want to use this)\par ;\par ; If you define a SIP proxy as a peer below, you may call\par ; SIP/proxyhostname/user or SIP/user@proxyhostname\par ; where the proxyhostname is defined in a section below\par ;\par ; Useful CLI commands to check peers/users:\par ; sip show peers Show all SIP peers (including friends)\par ; sip show users Show all SIP users (including friends)\par ; sip show registry Show status of hosts we register with\par ;\par ; sip debug Show all SIP messages\par ;\par ; module reload chan_sip.so Reload configuration file\par ; Active SIP peers will not be reconfigured\par ;\par \par [general]\par context=default ; Default context for incoming calls\par ;allowguest=no ; Allow or reject guest calls (default is yes)\par allowoverlap=no ; Disable overlap dialing support. (Default is yes)\par ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)\par ; Default is enabled\par ;realm=ergatel.net ; Realm for digest authentication\par ; defaults to "asterisk". If you set a system name in\par ; asterisk.conf, it defaults to that system name\par ; Realms MUST be globally unique according to RFC 3261\par ; Set this to your host name or domain name\par bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)\par ; bindport is the local UDP port that Asterisk will listen on\par bindaddr=IP_ADDRESS ; IP address to bind to (0.0.0.0 binds to all)\par srvlookup=yes ; Enable DNS SRV lookups on outbound calls\par ; Note: Asterisk only uses the first host\par ; in SRV records\par ; Disabling DNS SRV lookups disables the\par ; ability to place SIP calls based on domain\par ; names to some other SIP users on the Internet\par ;pedantic=yes ; Enable checking of tags in headers,\par ; international character conversions in URIs\par ; and multiline formatted headers for strict\par ; SIP compatibility (defaults to "no")\par \par ; See doc/ip-tos.txt for a description of these parameters.\par ;tos_sip=cs3 ; Sets TOS for SIP packets.\par ;tos_audio=ef ; Sets TOS for RTP audio packets.\par ;tos_video=af41 ; Sets TOS for RTP video packets.\par \par ;maxexpiry=3600 ; Maximum allowed time of incoming registrations\par ; and subscriptions (seconds)\par ;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)\par ;defaultexpiry=120 ; Default length of incoming/outgoing registration\par ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts\par ; Defaults to 100 ms\par ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY\par ;checkmwi=10 ; Default time between mailbox checks for peers\par ;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC\par ; fully. Enable this option to not get error messages\par ; when sending MWI to phones with this bug.\par ;vmexten=voicemail ; dialplan extension to reach mailbox sets the\par ; Message-Account in the MWI notify message\par ; defaults to "asterisk"\par disallow=all ; First disallow all codecs\par allow=ulaw ; Allow codecs in order of preference\par allow=alaw\par ;allow=g729\par ;allow=g726\par ;allow=ilbc ; see doc/rtp-packetization for framing options\par ;allow=alaw\par ; This option specifies a preference for which music on hold class this channel\par ; should listen to when put on hold if the music class has not been set on the\par ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer\par ; channel putting this one on hold did not suggest a music class.\par ;\par ; This option may be specified globally, or on a per-user or per-peer basis.\par ;\par ;mohinterpret=default\par ;\par ; This option specifies which music on hold class to suggest to the peer channel\par ; when this channel places the peer on hold. It may be specified globally or on\par ; a per-user or per-peer basis.\par ;;\par ;mohsuggest=default\par ;\par ;language=en ; Default language setting for all users/peers\par ; This may also be set for individual users/peers\par ;relaxdtmf=yes ; Relax dtmf handling\par ;trustrpid = no ; If Remote-Party-ID should be trusted\par ;sendrpid = yes ; If Remote-Party-ID should be sent\par ;progressinband=never ; If we should generate in-band ringing always\par ; use 'never' to never use in-band signalling, even in cases\par ; where some buggy devices might not render it\par ; Valid values: yes, no, never Default: never\par useragent= Ergatel ; Allows you to change the user agent string\par ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address\par ; Note that promiscredir when redirects are made to the\par ; local system will cause loops since Asterisk is incapable\par ; of performing a "hairpin" call.\par ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains\par ; a valid phone number\par ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833\par ; Other options:\par ; info : SIP INFO messages\par ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)\par ; auto : Use rfc2833 if offered, inband otherwise\par \par ;compactheaders = yes ; send compact sip headers.\par ;\par ;videosupport=yes ; Turn on support for SIP video. You need to turn this on\par ; in the this section to get any video support at all.\par ; You can turn it off on a per peer basis if the general\par ; video support is enabled, but you can't enable it for\par ; one peer only without enabling in the general section.\par ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)\par ; Videosupport and maxcallbitrate is settable\par ; for peers and users as well\par ;callevents=no ; generate manager events when sip ua\par ; performs events (e.g. hold)\par ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,\par ; for any reason, always reject with '401 Unauthorized'\par ; instead of letting the requester know whether there was\par ; a matching user or peer for their request\par \par ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing\par ; order instead of RFC3551 packing order (this is required\par ; for Sipura and Grandstream ATAs, among others). This is ; order instead of RFC3551 packing order (this is required\par ; for Sipura and Grandstream ATAs, among others). This is\par ; contrary to the RFC3551 specification, the peer _should_\par ; be negotiating AAL2-G726-32 instead :-(\par \par ;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches\par ; your localnet setting. Unless you have some sort of strange network\par ; setup you will not need to enable this.\par \par ;\par ; If regcontext is specified, Asterisk will dynamically create and destroy a\par ; NoOp priority 1 extension for a given peer who registers or unregisters with\par ; us and have a "regexten=" configuration item.\par ; Multiple contexts may be specified by separating them with '&'. The\par ; actual extension is the 'regexten' parameter of the registering peer or its\par ; name if 'regexten' is not provided. If more than one context is provided,\par ; the context must be specified within regexten by appending the desired\par ; context after '@'. More than one regexten may be supplied if they are\par ; separated by '&'. Patterns may be used in regexten.\par ;\par ;regcontext=sipregistrations\par ;\par ;--------------------------- RTP timers ----------------------------------------------------\par ; These timers are currently used for both audio and video streams. The RTP timeouts\par ; are only applied to the audio channel.\par ; The settings are settable in the global section as well as per device\par ;\par ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity\par ; on the audio channel\par ; when we're not on hold. This is to be able to hangup\par ; a call in the case of a phone disappearing from the net,\par ; like a powerloss or grandma tripping over a cable.\par ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity\par ; on the audio channel\par ; when we're on hold (must be > rtptimeout)\par ;rtpkeepalive= ; Send keepalives in the RTP stream to keep NAT open\par ; (default is off - zero)\par ;--------------------------- SIP DEBUGGING ---------------------------------------------------\par ;sipdebug = yes ; Turn on SIP debugging by default, from\par ; the moment the channel loads this configuration\par ;recordhistory=yes ; Record SIP history by default\par ; (see sip history / sip no history)\par ;dumphistory=yes ; Dump SIP history at end of SIP dialogue\par ; SIP history is output to the DEBUG logging channel\par ; SIP history is output to the DEBUG logging channel\par \par \par ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------\par ; You can subscribe to the status of extensions with a "hint" priority\par ; (See extensions.conf.sample for examples)\par ; chan_sip support two major formats for notifications: dialog-info and SIMPLE\par ;\par ; You will get more detailed reports (busy etc) if you have a call limit set\par ; for a device. When the call limit is filled, we will indicate busy. Note that\par ; you need at least 2 in order to be able to do attended transfers.\par ;\par ; For queues, you will need this level of detail in status reporting, regardless\par ; if you use SIP subscriptions. Queues and manager use the same internal interface\par ; for reading status information.\par ;\par ; Note: Subscriptions does not work if you have a realtime dialplan and use the\par ; realtime switch.\par ;\par ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)\par ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests\par ; Useful to limit subscriptions to local extensions\par ; Settable per peer/user also\par ;notifyringing = yes ; Control whether subscriptions already INUSE get sent\par ; RINGING when another call is sent (default: no)\par ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)\par ; Turning on notifyringing and notifyhold will add a lot\par ; more database transactions if you are using realtime.\par ;limitonpeers = yes ; Apply call limits on peers only. This will improve\par ; status notification when you are using type=friend\par ; Inbound calls, that really apply to the user part\par ; of a friend will now be added to and compared with\par ; the peer limit instead of applying two call limits,\par ; one for the peer and one for the user.\par ; "sip show inuse" will only show active calls on\par ; the peer side of a "type=friend" object if this\par ; setting is turned on.\par \par ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------\par ;\par ; This setting is available in the [general] section as well as in device configurations.\par ; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided\par ; both parties have T38 support enabled in their Asterisk configuration\par ; This has to be enabled in the general section for all devices to work. You can then\par ; disable it on a per device basis.;\par ; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.\par ;\par t38pt_udptl = yes ; Default false\par ;\par ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------\par ; Asterisk can register as a SIP user agent to a SIP proxy (provider)\par ; Format for the register statement is:\par ; register => user[:secret[:authuser]]@host[:port][/extension]\par ;\par ; If no extension is given, the 's' extension is used. The extension needs to\par ; be defined in extensions.conf to be able to accept calls from this SIP proxy\par ; (provider).\par ;\par ; host is either a host name defined in DNS or the name of a section defined\par ; below.\par ;\par ; Examples:\par ;\par ;register => 1234:password@mysipprovider.com\par ;\par ; This will pass incoming calls to the 's' extension\par ;\par ;\par ;register => 2345:password@sip_proxy/1234\par ;\par ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider\par ; connect to local extension 1234 in extensions.conf, default context,\par ; unless you configure a [sip_proxy] section below, and configure a\par ; context.\par ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]\par ; Tip 2: Use separate type=peer and type=user sections for SIP providers\par ; (instead of type=friend) if you have calls in both directions\par \par ;registertimeout=20 ; retry registration calls every 20 seconds (default)\par ;registerattempts=10 ; Number of registration attempts before we give up\par ; 0 = continue forever, hammering the other server\par ; until it accepts the registration\par ; Default is 0 tries, continue forever\par \par ;----------------------------------------- NAT SUPPORT ------------------------\par ; The externip, externhost and localnet settings are used if you use Asterisk\par ; behind a NAT device to communicate with services on the outside.\par ;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP\par ; messages if we're behind a NAT\par \par ; The externip and localnet is used\par ; when registering and communicating with other proxies\par ; that we're registered with\par ;externhost=foo.dyndns.net ; Alternatively you can specify an\par ; external host, and Asterisk will\par ; perform DNS queries periodically. Not\par ; recommended for production\par ; environments! Use externip instead\par ;externrefresh=10 ; How often to refresh externhost if\par ; used\par ; You may add multiple local networks. A reasonable\par ; set of defaults are:\par ;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks\par ;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918\par ;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation\par ;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network\par \par ; The nat= setting is used when Asterisk is on a public IP, communicating with\par ; devices hidden behind a NAT device (broadband router). If you have one-way\par ; audio problems, you usually have problems with your NAT configuration or your\par ; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP\par ; ports for incoming audio in rtp.conf\par ;\par ;nat=no ; Global NAT settings (Affects all peers and users)\par ; yes = Always ignore info and assume NAT\par ; no = Use NAT mode only according to RFC3581 (;rport)\par ; never = Never attempt NAT mode or RFC3581 support\par ; route = Assume NAT, don't send rport\par ; (work around more UNIDEN bugs)\par \par ;----------------------------------- MEDIA HANDLING --------------------------------\par ; By default, Asterisk tries to re-invite the audio to an optimal path. If there's\par ; no reason for Asterisk to stay in the media path, the media will be redirected.\par ; This does not really work with in the case where Asterisk is outside and have\par ; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat\par ;\par ;canreinvite=yes ; Asterisk by default tries to redirect the\par ; RTP media stream (audio) to go directly from\par ; the caller to the callee. Some devices do not\par ; support this (especially if one of them is behind a NAT).\par ; The default setting is YES. If you have all clients\par \par ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up\par ; the call directly with media peer-2-peer without re-invites.\par ; Will not work for video and cases where the callee sends\par ; RTP payloads and fmtp headers in the 200 OK that does not match the\par ; callers INVITE. This will also fail if canreinvite is enabled when\par ; the device is actually behind NAT.\par \par ;canreinvite=nonat ; An additional option is to allow media path redirection\par ; (reinvite) but only when the peer where the media is being\par ; sent is known to not be behind a NAT (as the RTP core can\par ; determine it based on the apparent IP address the media\par ; arrives from).\par \par ;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,\par ; instead of INVITE. This can be combined with 'nonat', as\par ; 'canreinvite=update,nonat'. It implies 'yes'.\par \par ;----------------------------------------- REALTIME SUPPORT ------------------------\par ; For additional information on ARA, the Asterisk Realtime Architecture,\par ; please read realtime.txt and extconfig.txt in the /doc directory of the\par ; source code.\par ;\par rtcachefriends=yes ; Cache realtime friends by adding them to the internal list\par ; just like friends added from the config file only on a\par ; as-needed basis? (yes|no)\par \par ;rtsavesysname=yes ; Save systemname in realtime database at registration\par ; Default= no\par \par ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)\par ; If set to yes, when a SIP UA registers successfully, the ip address,\par ; the origination port, the registration period, and the username of\par ; the UA will be set to database via realtime.\par ; If not present, defaults to 'yes'. Note: realtime peers will\par ; probably not function across reloads in the way that you expect, if\par ; you turn this option off.\par ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule\par ; as if it had just registered? (yes|no|)\par ; If set to yes, when the registration expires, the friend will\par ; vanish from the configuration until requested again. If set\par ; to an integer, friends expire within this number of seconds\par ; instead of the registration interval.\par \par ;ignoreregexpire=yes ; Enabling this setting has two functions:\par \par ;domain=mydomain.tld,mydomain-incoming\par ; Add domain and configure incoming context\par ; for external calls to this domain\par ;domain=1.2.3.4 ; Add IP address as local domain\par ; You can have several "domain" settings\par ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains\par ; Default is yes\par ;autodomain=yes ; Turn this on to have Asterisk add local host\par ; name and local IP to domain list.\par \par ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to\par ; non-peers, use your primary domain "identity"\par ; for From: headers instead of just your IP\par ; address. This is to be polite and\par ; it may be a mandatory requirement for some\par ; destinations which do not have a prior\par ; account relationship with your server.\par \par ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------\par ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a\par ; SIP channel. Defaults to "no". An enabled jitterbuffer will\par ; be used only if the sending side can create and the receiving\par ; side can not accept jitter. The SIP channel can accept jitter,\par ; thus a jitterbuffer on the receive SIP side will be used only\par ; if it is forced and enabled.\par \par ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP\par ; channel. Defaults to "no".\par \par ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.\par \par ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is\par ; resynchronized. Useful to improve the quality of the voice, with\par ; big jumps in/broken timestamps, usually sent from exotic devices\par ; and programs. Defaults to 1000.\par \par ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP\par ; channel. Two implementations are currently available - "fixed"\par ; (with size always equals to jbmaxsize) and "adaptive" (with\par ; variable size, actually the new jb of IAX2). Defaults to fixed.\par \par ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".\par ;-----------------------------------------------------------------------------------\par }