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https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48143 | file | 2006-11-30 12:57:35 -0500 (Thu, 30 Nov 2006) | 10 lines Merged revisions 48142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov 2006) | 2 lines Document 'port' for SIP peers, came up because of the current mailing list thread. (issue #8450 reported by blitzrage) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
700 lines
31 KiB
Plaintext
700 lines
31 KiB
Plaintext
;
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; SIP Configuration example for Asterisk
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;
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; Syntax for specifying a SIP device in extensions.conf is
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; SIP/devicename where devicename is defined in a section below.
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;
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; You may also use
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; SIP/username@domain to call any SIP user on the Internet
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; (Don't forget to enable DNS SRV records if you want to use this)
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;
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; If you define a SIP proxy as a peer below, you may call
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; SIP/proxyhostname/user or SIP/user@proxyhostname
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; where the proxyhostname is defined in a section below
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;
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; Useful CLI commands to check peers/users:
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; sip list peers Show all SIP peers (including friends)
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; sip list users Show all SIP users (including friends)
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; sip list registry Show status of hosts we register with
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;
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; sip debug Show all SIP messages
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;
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; sip reload Reload configuration file
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; Active SIP peers will not be reconfigured
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;
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[general]
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context=default ; Default context for incoming calls
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;allowguest=no ; Allow or reject guest calls (default is yes)
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;match_auth_username=yes ; if available, match user entry using the
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; 'username' field from the authentication line
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; instead of the From: field.
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allowoverlap=no ; Disable overlap dialing support. (Default is yes)
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;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
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; Default is enabled
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;realm=mydomain.tld ; Realm for digest authentication
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; defaults to "asterisk". If you set a system name in
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; asterisk.conf, it defaults to that system name
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; Realms MUST be globally unique according to RFC 3261
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; Set this to your host name or domain name
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bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
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; bindport is the local UDP port that Asterisk will listen on
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bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
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srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; Note: Asterisk only uses the first host
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; in SRV records
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; Disabling DNS SRV lookups disables the
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; ability to place SIP calls based on domain
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; names to some other SIP users on the Internet
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;domain=mydomain.tld ; Set default domain for this host
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; If configured, Asterisk will only allow
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; INVITE and REFER to non-local domains
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; Use "sip show domains" to list local domains
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;pedantic=yes ; Enable checking of tags in headers,
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; international character conversions in URIs
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; and multiline formatted headers for strict
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; SIP compatibility (defaults to "no")
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; See doc/README.tos for a description of these parameters.
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;tos_sip=cs3 ; Sets TOS for SIP packets.
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;tos_audio=ef ; Sets TOS for RTP audio packets.
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;tos_video=af41 ; Sets TOS for RTP video packets.
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;maxexpiry=3600 ; Maximum allowed time of incoming registrations
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; and subscriptions (seconds)
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;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
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;defaultexpiry=120 ; Default length of incoming/outgoing registration
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;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
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; Defaults to 100 ms
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;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
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;checkmwi=10 ; Default time between mailbox checks for peers
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;vmexten=voicemail ; dialplan extension to reach mailbox sets the
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; Message-Account in the MWI notify message
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; defaults to "asterisk"
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;disallow=all ; First disallow all codecs
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;allow=ulaw ; Allow codecs in order of preference
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;allow=ilbc ; see doc/rtp-packetization for framing options
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;
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; This option specifies a preference for which music on hold class this channel
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; should listen to when put on hold if the music class has not been set on the
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; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
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; channel putting this one on hold did not suggest a music class.
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;
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; This option may be specified globally, or on a per-user or per-peer basis.
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;
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;mohinterpret=default
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;
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; This option specifies which music on hold class to suggest to the peer channel
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; when this channel places the peer on hold. It may be specified globally or on
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; a per-user or per-peer basis.
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;
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;mohsuggest=default
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;
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;language=en ; Default language setting for all users/peers
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; This may also be set for individual users/peers
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;relaxdtmf=yes ; Relax dtmf handling
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;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
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; when we're not on hold. This is to be able to hangup
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; a call in the case of a phone disappearing from the net,
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; like a powerloss or grandma tripping over a cable.
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;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
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; when we're on hold (must be > rtptimeout)
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;trustrpid = no ; If Remote-Party-ID should be trusted
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;sendrpid = yes ; If Remote-Party-ID should be sent
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;progressinband=never ; If we should generate in-band ringing always
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; use 'never' to never use in-band signalling, even in cases
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; where some buggy devices might not render it
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; Valid values: yes, no, never Default: never
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;useragent=Asterisk PBX ; Allows you to change the user agent string
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;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
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; Note that promiscredir when redirects are made to the
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; local system will cause loops since Asterisk is incapable
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; of performing a "hairpin" call.
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;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
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; a valid phone number
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;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
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; Other options:
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; info : SIP INFO messages
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; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
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; auto : Use rfc2833 if offered, inband otherwise
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;compactheaders = yes ; send compact sip headers.
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;
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;videosupport=yes ; Turn on support for SIP video. You need to turn this on
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; in the this section to get any video support at all.
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; You can turn it off on a per peer basis if the general
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; video support is enabled, but you can't enable it for
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; one peer only without enabling in the general section.
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;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
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; Videosupport and maxcallbitrate is settable
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; for peers and users as well
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;callevents=no ; generate manager events when sip ua
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; performs events (e.g. hold)
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;limitpeersonly=no ; Apply all call limits ("limit=") only to peers, never
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; to users. This improves handling of call limits
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; and device states in certain situations. The user part
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; of a type=friend will still be affected by the call
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; limit, but Asterisk will only use one object for
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; counting the simultaneous calls.
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;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
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; for any reason, always reject with '401 Unauthorized'
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; instead of letting the requester know whether there was
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; a matching user or peer for their request
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;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
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; order instead of RFC3551 packing order (this is required
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; for Sipura and Grandstream ATAs, among others). This is
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; contrary to the RFC3551 specification, the peer _should_
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; be negotiating AAL2-G726-32 instead :-(
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;
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; If regcontext is specified, Asterisk will dynamically create and destroy a
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; NoOp priority 1 extension for a given peer who registers or unregisters with
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; us and have a "regexten=" configuration item.
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; Multiple contexts may be specified by separating them with '&'. The
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; actual extension is the 'regexten' parameter of the registering peer or its
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; name if 'regexten' is not provided. If more than one context is provided,
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; the context must be specified within regexten by appending the desired
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; context after '@'. More than one regexten may be supplied if they are
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; separated by '&'. Patterns may be used in regexten.
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;
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;regcontext=sipregistrations
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;
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;--------------------------- SIP DEBUGGING ---------------------------------------------------
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;sipdebug = yes ; Turn on SIP debugging by default, from
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; the moment the channel loads this configuration
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;recordhistory=yes ; Record SIP history by default
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; (see sip history / sip no history)
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;dumphistory=yes ; Dump SIP history at end of SIP dialogue
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; SIP history is output to the DEBUG logging channel
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;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
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; You can subscribe to the status of extensions with a "hint" priority
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; (See extensions.conf.sample for examples)
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; chan_sip support two major formats for notifications: dialog-info and SIMPLE
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;
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; You will get more detailed reports (busy etc) if you have a call limit set
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; for a device. When the call limit is filled, we will indicate busy. Note that
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; you need at least 2 in order to be able to do attended transfers.
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;
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; For queues, you will need this level of detail in status reporting, regardless
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; if you use SIP subscriptions. Queues and manager use the same internal interface
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; for reading status information.
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;
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; Note: Subscriptions does not work if you have a realtime dialplan and use the
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; realtime switch.
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;
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;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
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;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
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; Useful to limit subscriptions to local extensions
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; Settable per peer/user also
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;notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
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;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
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; Turning on notifyringing and notifyhold will add a lot
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; more database transactions if you are using realtime.
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;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
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;
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; This setting is available in the [general] section as well as in device configurations.
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; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
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; both parties have T38 support enabled in their Asterisk configuration (either general or
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; peer/user/friend sections)
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;
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; t38pt_udptl = yes ; Default false
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;
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;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
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; Asterisk can register as a SIP user agent to a SIP proxy (provider)
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; Format for the register statement is:
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; register => user[:secret[:authuser]]@host[:port][/extension]
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;
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; If no extension is given, the 's' extension is used. The extension needs to
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; be defined in extensions.conf to be able to accept calls from this SIP proxy
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; (provider).
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;
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; host is either a host name defined in DNS or the name of a section defined
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; below.
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;
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; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
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; this is equivalent to having the following line in the general section:
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;
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; register => username:secret@host/callbackextension
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;
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; and more readable because you don't have to write the parameters in two places
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; (note that the "port" is ignored - this is a bug that should be fixed).
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;
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; Examples:
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;
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;register => 1234:password@mysipprovider.com
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;
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; This will pass incoming calls to the 's' extension
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;
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;
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;register => 2345:password@sip_proxy/1234
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;
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; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
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; connect to local extension 1234 in extensions.conf, default context,
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; unless you configure a [sip_proxy] section below, and configure a
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; context.
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; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
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; Tip 2: Use separate type=peer and type=user sections for SIP providers
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; (instead of type=friend) if you have calls in both directions
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;registertimeout=20 ; retry registration calls every 20 seconds (default)
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;registerattempts=10 ; Number of registration attempts before we give up
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; 0 = continue forever, hammering the other server
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; until it accepts the registration
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; Default is 0 tries, continue forever
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;----------------------------------------- NAT SUPPORT ------------------------
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; The externip, externhost and localnet settings are used if you use Asterisk
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; behind a NAT device to communicate with services on the outside.
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;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP
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; messages if we're behind a NAT
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; The externip and localnet is used
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; when registering and communicating with other proxies
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; that we're registered with
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;externhost=foo.dyndns.net ; Alternatively you can specify an
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; external host, and Asterisk will
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; perform DNS queries periodically. Not
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; recommended for production
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; environments! Use externip instead
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;externrefresh=10 ; How often to refresh externhost if
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; used
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; You may add multiple local networks. A reasonable
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; set of defaults are:
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;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
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;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
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;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
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;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
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; The nat= setting is used when Asterisk is on a public IP, communicating with
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; devices hidden behind a NAT device (broadband router). If you have one-way
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; audio problems, you usually have problems with your NAT configuration or your
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; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
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; ports for incoming audio in rtp.conf
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;
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;nat=no ; Global NAT settings (Affects all peers and users)
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; yes = Always ignore info and assume NAT
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; no = Use NAT mode only according to RFC3581 (;rport)
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; never = Never attempt NAT mode or RFC3581 support
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; route = Assume NAT, don't send rport
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; (work around more UNIDEN bugs)
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;----------------------------------- MEDIA HANDLING --------------------------------
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; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
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; no reason for Asterisk to stay in the media path, the media will be redirected.
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; This does not really work with in the case where Asterisk is outside and have
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; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
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;
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;canreinvite=yes ; Asterisk by default tries to redirect the
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; RTP media stream (audio) to go directly from
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; the caller to the callee. Some devices do not
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; support this (especially if one of them is behind a NAT).
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; The default setting is YES. If you have all clients
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; behind a NAT, or for some other reason wants Asterisk to
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; stay in the audio path, you may want to turn this off.
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; This setting also affect direct RTP
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; at call setup (a new feature in 1.4 - setting up the
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; call directly between the endpoints instead of sending
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; a re-INVITE).
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;canreinvite=nonat ; An additional option is to allow media path redirection
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; (reinvite) but only when the peer where the media is being
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; sent is known to not be behind a NAT (as the RTP core can
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; determine it based on the apparent IP address the media
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; arrives from).
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;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
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; instead of INVITE. This can be combined with 'nonat', as
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; 'canreinvite=update,nonat'. It implies 'yes'.
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;----------------------------------------- REALTIME SUPPORT ------------------------
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; For additional information on ARA, the Asterisk Realtime Architecture,
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; please read realtime.txt and extconfig.txt in the /doc directory of the
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; source code.
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;
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;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
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; just like friends added from the config file only on a
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; as-needed basis? (yes|no)
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;rtsavesysname=yes ; Save systemname in realtime database at registration
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; Default= no
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;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
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; If set to yes, when a SIP UA registers successfully, the ip address,
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; the origination port, the registration period, and the username of
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; the UA will be set to database via realtime.
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; If not present, defaults to 'yes'.
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;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
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; as if it had just registered? (yes|no|<seconds>)
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; If set to yes, when the registration expires, the friend will
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; vanish from the configuration until requested again. If set
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; to an integer, friends expire within this number of seconds
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; instead of the registration interval.
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;ignoreregexpire=yes ; Enabling this setting has two functions:
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;
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; For non-realtime peers, when their registration expires, the
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; information will _not_ be removed from memory or the Asterisk database
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; if you attempt to place a call to the peer, the existing information
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; will be used in spite of it having expired
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;
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; For realtime peers, when the peer is retrieved from realtime storage,
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; the registration information will be used regardless of whether
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; it has expired or not; if it expires while the realtime peer
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; is still in memory (due to caching or other reasons), the
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; information will not be removed from realtime storage
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;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
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; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
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; domains, each of which can direct the call to a specific context if desired.
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; By default, all domains are accepted and sent to the default context or the
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; context associated with the user/peer placing the call.
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; Domains can be specified using:
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; domain=<domain>[,<context>]
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; Examples:
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; domain=myasterisk.dom
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; domain=customer.com,customer-context
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;
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; In addition, all the 'default' domains associated with a server should be
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; added if incoming request filtering is desired.
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; autodomain=yes
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;
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; To disallow requests for domains not serviced by this server:
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; allowexternaldomains=no
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;domain=mydomain.tld,mydomain-incoming
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; Add domain and configure incoming context
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; for external calls to this domain
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;domain=1.2.3.4 ; Add IP address as local domain
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; You can have several "domain" settings
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;allowexternalinvites=no ; Disable INVITE and REFER to non-local domains
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; Default is yes
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;autodomain=yes ; Turn this on to have Asterisk add local host
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; name and local IP to domain list.
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; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
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; non-peers, use your primary domain "identity"
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; for From: headers instead of just your IP
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; address. This is to be polite and
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; it may be a mandatory requirement for some
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; destinations which do not have a prior
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; account relationship with your server.
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;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
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; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
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; SIP channel. Defaults to "no". An enabled jitterbuffer will
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; be used only if the sending side can create and the receiving
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; side can not accept jitter. The SIP channel can accept jitter,
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; thus a jitterbuffer on the receive SIP side will be used only
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; if it is forced and enabled.
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; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
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; channel. Defaults to "no".
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; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
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; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
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; resynchronized. Useful to improve the quality of the voice, with
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; big jumps in/broken timestamps, usually sent from exotic devices
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; and programs. Defaults to 1000.
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; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
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; channel. Two implementations are currently available - "fixed"
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; (with size always equals to jbmaxsize) and "adaptive" (with
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; variable size, actually the new jb of IAX2). Defaults to fixed.
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; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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;-----------------------------------------------------------------------------------
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[authentication]
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; Global credentials for outbound calls, i.e. when a proxy challenges your
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; Asterisk server for authentication. These credentials override
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; any credentials in peer/register definition if realm is matched.
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;
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; This way, Asterisk can authenticate for outbound calls to other
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; realms. We match realm on the proxy challenge and pick an set of
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; credentials from this list
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; Syntax:
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; auth = <user>:<secret>@<realm>
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; auth = <user>#<md5secret>@<realm>
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; Example:
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;auth=mark:topsecret@digium.com
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;
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; You may also add auth= statements to [peer] definitions
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; Peer auth= override all other authentication settings if we match on realm
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;------------------------------------------------------------------------------
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; Users and peers have different settings available. Friends have all settings,
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; since a friend is both a peer and a user
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;
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; User config options: Peer configuration:
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; -------------------- -------------------
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; context context
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; callingpres callingpres
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; permit permit
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; deny deny
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; secret secret
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; md5secret md5secret
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; dtmfmode dtmfmode
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; canreinvite canreinvite
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; nat nat
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; callgroup callgroup
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; pickupgroup pickupgroup
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; language language
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; allow allow
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; disallow disallow
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; insecure insecure
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; trustrpid trustrpid
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; progressinband progressinband
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; promiscredir promiscredir
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; useclientcode useclientcode
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; accountcode accountcode
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; setvar setvar
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; callerid callerid
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; amaflags amaflags
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; call-limit call-limit
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; allowoverlap allowoverlap
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; allowsubscribe allowsubscribe
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; allowtransfer allowtransfer
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; subscribecontext subscribecontext
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; videosupport videosupport
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; maxcallbitrate maxcallbitrate
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; rfc2833compensate mailbox
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; username
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; template
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; fromdomain
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; regexten
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; fromuser
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; host
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; port
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; qualify
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; defaultip
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; rtptimeout
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; rtpholdtimeout
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; sendrpid
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; outboundproxy
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; rfc2833compensate
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; callbackextension
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;[sip_proxy]
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; For incoming calls only. Example: FWD (Free World Dialup)
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; We match on IP address of the proxy for incoming calls
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; since we can not match on username (caller id)
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;type=peer
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;context=from-fwd
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;host=fwd.pulver.com
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;[sip_proxy-out]
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;type=peer ; we only want to call out, not be called
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;secret=guessit
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;username=yourusername ; Authentication user for outbound proxies
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;fromuser=yourusername ; Many SIP providers require this!
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;fromdomain=provider.sip.domain
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;host=box.provider.com
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;usereqphone=yes ; This provider requires ";user=phone" on URI
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;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer
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;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
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; Call-limits will not be enforced on real-time peers,
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; since they are not stored in-memory
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;port=80 ; The port number we want to connect to on the remote side
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;--- sample definition for a provider
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;[provider1]
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;type=peer
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;host=sip.provider1.com
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;username=4015552299 ; how your provider knows you
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;secret=youwillneverguessit
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;callbackextension=123 ; Register with this server and require calls coming back to this extension
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;------------------------------------------------------------------------------
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; Definitions of locally connected SIP devices
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;
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; type = user a device that authenticates to us by "from" field to place calls
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; type = peer a device we place calls to or that calls us and we match by host
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; type = friend two configurations (peer+user) in one
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;
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; For device names, we recommend using only a-z, numerics (0-9) and underscore
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;
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; For local phones, type=friend works most of the time
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;
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; If you have one-way audio, you probably have NAT problems.
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; If Asterisk is on a public IP, and the phone is inside of a NAT device
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; you will need to configure nat option for those phones.
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; Also, turn on qualify=yes to keep the nat session open
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;
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; Because you might have a large number of similar sections, it is generally
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; convenient to use templates for the common parameters, and add them
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; the the various sections. Examples are below, and we can even leave
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; the templates uncommented as they will not harm:
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[basic-options](!) ; a template
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dtmfmode=rfc2833
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context=from-office
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type=friend
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[natted-phone](!,basic-options) ; another template inheriting basic-options
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nat=yes
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canreinvite=no
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host=dynamic
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[public-phone](!,basic-options) ; another template inheriting basic-options
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nat=no
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canreinvite=yes
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[my-codecs](!) ; a template for my preferred codecs
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disallow=all
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allow=ilbc
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allow=g729
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allow=gsm
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allow=g723
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allow=ulaw
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[ulaw-phone](!) ; and another one for ulaw-only
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disallow=all
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allow=ulaw
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; and finally instantiate a few phones
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;
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; [2133](natted-phone,my-codecs)
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; secret = peekaboo
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; [2134](natted-phone,ulaw-hone)
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; secret = not_very_secret
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; [2136](public-phone,ulaw-hone)
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; secret = not_very_secret_either
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; ...
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;
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; Standard configurations not using templates look like this:
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;
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;[grandstream1]
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;type=friend
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;context=from-sip ; Where to start in the dialplan when this phone calls
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;callerid=John Doe <1234> ; Full caller ID, to override the phones config
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; on incoming calls to Asterisk
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;host=192.168.0.23 ; we have a static but private IP address
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; No registration allowed
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;nat=no ; there is not NAT between phone and Asterisk
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;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
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;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
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;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
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; from the phone to asterisk
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; 1 for the explicit peer, 1 for the explicit user,
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; remember that a friend equals 1 peer and 1 user in
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; memory
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; This will affect your subscriptions as well.
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; There is no combined call counter for a "friend"
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; so there's currently no way in sip.conf to limit
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; to one inbound or outbound call per phone. Use
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; the group counters in the dial plan for that.
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;
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;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
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;disallow=all ; need to disallow=all before we can use allow=
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;allow=ulaw ; Note: In user sections the order of codecs
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; listed with allow= does NOT matter!
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;allow=alaw
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;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
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;allow=g729 ; Pass-thru only unless g729 license obtained
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;callingpres=allowed_passed_screen ; Set caller ID presentation
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; See README.callingpres for more information
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;[xlite1]
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; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
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; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
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;type=friend
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;regexten=1234 ; When they register, create extension 1234
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;callerid="Jane Smith" <5678>
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;host=dynamic ; This device needs to register
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;nat=yes ; X-Lite is behind a NAT router
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;canreinvite=no ; Typically set to NO if behind NAT
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;disallow=all
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;allow=gsm ; GSM consumes far less bandwidth than ulaw
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;allow=ulaw
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;allow=alaw
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;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
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;[snom]
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;type=friend ; Friends place calls and receive calls
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;context=from-sip ; Context for incoming calls from this user
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;secret=blah
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;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
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;language=de ; Use German prompts for this user
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;host=dynamic ; This peer register with us
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;dtmfmode=inband ; Choices are inband, rfc2833, or info
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;defaultip=192.168.0.59 ; IP used until peer registers
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;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
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;subscribemwi=yes ; Only send notifications if this phone
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; subscribes for mailbox notification
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;vmexten=voicemail ; dialplan extension to reach mailbox
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; sets the Message-Account in the MWI notify message
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; defaults to global vmexten which defaults to "asterisk"
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;disallow=all
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;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
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;[polycom]
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;type=friend ; Friends place calls and receive calls
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;context=from-sip ; Context for incoming calls from this user
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;secret=blahpoly
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;host=dynamic ; This peer register with us
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;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
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;username=polly ; Username to use in INVITE until peer registers
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; Normally you do NOT need to set this parameter
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;disallow=all
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;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
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;progressinband=no ; Polycom phones don't work properly with "never"
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;[pingtel]
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;type=friend
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;secret=blah
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;host=dynamic
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;insecure=port ; Allow matching of peer by IP address without
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; matching port number
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;insecure=invite ; Do not require authentication of incoming INVITEs
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;insecure=port,invite ; (both)
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;qualify=1000 ; Consider it down if it's 1 second to reply
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; Helps with NAT session
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; qualify=yes uses default value
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;
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; Call group and Pickup group should be in the range from 0 to 63
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;
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;callgroup=1,3-4 ; We are in caller groups 1,3,4
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;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
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;defaultip=192.168.0.60 ; IP address to use if peer has not registered
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;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
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;permit=192.168.0.60/255.255.255.0
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;[cisco1]
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;type=friend
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;secret=blah
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;qualify=200 ; Qualify peer is no more than 200ms away
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;nat=yes ; This phone may be natted
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; Send SIP and RTP to the IP address that packet is
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; received from instead of trusting SIP headers
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;host=dynamic ; This device registers with us
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;canreinvite=no ; Asterisk by default tries to redirect the
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; RTP media stream (audio) to go directly from
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; the caller to the callee. Some devices do not
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; support this (especially if one of them is
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; behind a NAT).
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;defaultip=192.168.0.4 ; IP address to use until registration
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;username=goran ; Username to use when calling this device before registration
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; Normally you do NOT need to set this parameter
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;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
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;[pre14-asterisk]
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;type=friend
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;secret=digium
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;host=dynamic
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;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
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; You must have this turned on or DTMF reception will work improperly.
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