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Updating docs
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
@@ -43,13 +43,11 @@
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; -------------------------------------------------------------
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; Useful CLI commands to check peers/users:
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; sip show peers Show all SIP peers (including friends)
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; sip show users Show all SIP users (including friends)
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; sip show registry Show status of hosts we register with
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;
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; sip set debug Show all SIP messages
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;
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; module reload chan_sip.so Reload configuration file
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; Active SIP peers will not be reconfigured
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;
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; ** Deprecated configuration options **
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@@ -380,15 +378,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; more database transactions if you are using realtime.
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;callcounter = yes ; Enable call counters on devices. This can be set per
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; device too.
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;counteronpeer = yes ; Apply call counting on peers only. This will improve
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; status notification when you are using type=friend
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; Inbound calls, that really apply to the user part
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; of a friend will now be added to and compared with
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; the peer counter instead of applying two call counters,
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; one for the peer and one for the user.
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; "sip show inuse" will only show active calls on
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; the peer side of a "type=friend" object if this
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; setting is turned on.
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;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
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;
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@@ -438,7 +427,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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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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; Tip 2: Use separate inbound and outbound 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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@@ -703,75 +692,92 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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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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; DEVICE CONFIGURATION
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;
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; The SIP channel has two types of devices, the friend and the peer.
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; * The type=friend is a device type that accepts both incoming and outbound calls,
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; where Asterisk match on the From: username on incoming calls.
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; (A synonym for friend is "user"). This is a type you use for your local
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; SIP phones.
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; * The type=peer also handles both incoming and outbound calls. On inbound calls,
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; Asterisk only matches on IP/port, not on names. This is mostly used for SIP
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; trunks.
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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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; remotesecret
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; secret secret
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; md5secret md5secret
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; transport transport
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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 (deprecated)
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; callcounter callcounter
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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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; session-timers busylevel
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; session-expires
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; session-minse template
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; session-refresher fromdomain
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; t38pt_usertpsource 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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; defaultuser
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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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; registertrying
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; session-timers
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; session-expires
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; session-minse
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; session-refresher
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; timert1
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; timerb
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; qualifyfreq
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; t38pt_usertpsource
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; contactpermit ; Limit what a host may register as (a neat trick
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; contactdeny ; is to register at the same IP as a SIP provider,
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; ; then call oneself, and get redirected to that
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; ; same location).
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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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; Configuration options available
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; --------------------
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; context
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; callingpres
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; permit
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; deny
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; secret
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; md5secret
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; remotesecret
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; transport
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; dtmfmode
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; canreinvite
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; nat
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; callgroup
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; pickupgroup
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; language
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; allow
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; disallow
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; insecure
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; trustrpid
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; progressinband
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; promiscredir
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; useclientcode
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; accountcode
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; setvar
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; callerid
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; amaflags
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; callcounter
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; busylevel
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; allowoverlap
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; allowsubscribe
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; allowtransfer
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; subscribecontext
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; template
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; videosupport
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; maxcallbitrate
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; rfc2833compensate
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; mailbox
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; session-timers
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; session-expires
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; session-minse
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; session-refresher
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; t38pt_usertpsource
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; regexten
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; fromdomain
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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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; defaultuser
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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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; registertrying
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; timert1
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; timerb
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; qualifyfreq
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; t38pt_usertpsource
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; contactpermit ; Limit what a host may register as (a neat trick
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; contactdeny ; is to register at the same IP as a SIP provider,
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; ; then call oneself, and get redirected to that
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; ; same location).
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;[sip_proxy]
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; For incoming calls only. Example: FWD (Free World Dialup)
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@@ -810,21 +816,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; ; accept both tcp and udp. Default is udp. The first transport
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; ; listed will always be used for outgoing connections.
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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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