[asterisk-commits] oej: trunk r153983 - /trunk/configs/sip.conf.sample
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Nov 3 12:02:16 CST 2008
Author: oej
Date: Mon Nov 3 12:02:14 2008
New Revision: 153983
URL: http://svn.digium.com/view/asterisk?view=rev&rev=153983
Log:
Updating docs
Modified:
trunk/configs/sip.conf.sample
Modified: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=153983&r1=153982&r2=153983
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Mon Nov 3 12:02:14 2008
@@ -43,13 +43,11 @@
; -------------------------------------------------------------
; Useful CLI commands to check peers/users:
; sip show peers Show all SIP peers (including friends)
-; sip show users Show all SIP users (including friends)
; sip show registry Show status of hosts we register with
;
; sip set debug Show all SIP messages
;
; module reload chan_sip.so Reload configuration file
-; Active SIP peers will not be reconfigured
;
; ** Deprecated configuration options **
@@ -380,15 +378,6 @@
; more database transactions if you are using realtime.
;callcounter = yes ; Enable call counters on devices. This can be set per
; device too.
-;counteronpeer = yes ; Apply call counting on peers only. This will improve
- ; status notification when you are using type=friend
- ; Inbound calls, that really apply to the user part
- ; of a friend will now be added to and compared with
- ; the peer counter instead of applying two call counters,
- ; one for the peer and one for the user.
- ; "sip show inuse" will only show active calls on
- ; the peer side of a "type=friend" object if this
- ; setting is turned on.
;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
;
@@ -438,7 +427,7 @@
; unless you configure a [sip_proxy] section below, and configure a
; context.
; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
-; Tip 2: Use separate type=peer and type=user sections for SIP providers
+; Tip 2: Use separate inbound and outbound sections for SIP providers
; (instead of type=friend) if you have calls in both directions
;registertimeout=20 ; retry registration calls every 20 seconds (default)
@@ -703,75 +692,92 @@
; Peer auth= override all other authentication settings if we match on realm
;------------------------------------------------------------------------------
-; Users and peers have different settings available. Friends have all settings,
-; since a friend is both a peer and a user
-;
-; User config options: Peer configuration:
-; -------------------- -------------------
-; context context
-; callingpres callingpres
-; permit permit
-; deny deny
-; remotesecret
-; secret secret
-; md5secret md5secret
-; transport transport
-; dtmfmode dtmfmode
-; canreinvite canreinvite
-; nat nat
-; callgroup callgroup
-; pickupgroup pickupgroup
-; language language
-; allow allow
-; disallow disallow
-; insecure insecure
-; trustrpid trustrpid
-; progressinband progressinband
-; promiscredir promiscredir
-; useclientcode useclientcode
-; accountcode accountcode
-; setvar setvar
-; callerid callerid
-; amaflags amaflags
-; call-limit call-limit (deprecated)
-; callcounter callcounter
-; allowoverlap allowoverlap
-; allowsubscribe allowsubscribe
-; allowtransfer allowtransfer
-; subscribecontext subscribecontext
-; videosupport videosupport
-; maxcallbitrate maxcallbitrate
-; rfc2833compensate mailbox
-; session-timers busylevel
-; session-expires
-; session-minse template
-; session-refresher fromdomain
-; t38pt_usertpsource regexten
-; fromuser
-; host
-; port
-; qualify
-; defaultip
-; defaultuser
-; rtptimeout
-; rtpholdtimeout
-; sendrpid
-; outboundproxy
-; rfc2833compensate
-; callbackextension
-; registertrying
-; session-timers
-; session-expires
-; session-minse
-; session-refresher
-; timert1
-; timerb
-; qualifyfreq
-; t38pt_usertpsource
-; contactpermit ; Limit what a host may register as (a neat trick
-; contactdeny ; is to register at the same IP as a SIP provider,
-; ; then call oneself, and get redirected to that
-; ; same location).
+; DEVICE CONFIGURATION
+;
+; The SIP channel has two types of devices, the friend and the peer.
+; * The type=friend is a device type that accepts both incoming and outbound calls,
+; where Asterisk match on the From: username on incoming calls.
+; (A synonym for friend is "user"). This is a type you use for your local
+; SIP phones.
+; * The type=peer also handles both incoming and outbound calls. On inbound calls,
+; Asterisk only matches on IP/port, not on names. This is mostly used for SIP
+; trunks.
+;
+; For device names, we recommend using only a-z, numerics (0-9) and underscore
+;
+; For local phones, type=friend works most of the time
+;
+; If you have one-way audio, you probably have NAT problems.
+; If Asterisk is on a public IP, and the phone is inside of a NAT device
+; you will need to configure nat option for those phones.
+; Also, turn on qualify=yes to keep the nat session open
+;
+; Configuration options available
+; --------------------
+; context
+; callingpres
+; permit
+; deny
+; secret
+; md5secret
+; remotesecret
+; transport
+; dtmfmode
+; canreinvite
+; nat
+; callgroup
+; pickupgroup
+; language
+; allow
+; disallow
+; insecure
+; trustrpid
+; progressinband
+; promiscredir
+; useclientcode
+; accountcode
+; setvar
+; callerid
+; amaflags
+; callcounter
+; busylevel
+; allowoverlap
+; allowsubscribe
+; allowtransfer
+; subscribecontext
+; template
+; videosupport
+; maxcallbitrate
+; rfc2833compensate
+; mailbox
+; session-timers
+; session-expires
+; session-minse
+; session-refresher
+; t38pt_usertpsource
+; regexten
+; fromdomain
+; fromuser
+; host
+; port
+; qualify
+; defaultip
+; defaultuser
+; rtptimeout
+; rtpholdtimeout
+; sendrpid
+; outboundproxy
+; rfc2833compensate
+; callbackextension
+; registertrying
+; timert1
+; timerb
+; qualifyfreq
+; t38pt_usertpsource
+; contactpermit ; Limit what a host may register as (a neat trick
+; contactdeny ; is to register at the same IP as a SIP provider,
+; ; then call oneself, and get redirected to that
+; ; same location).
;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
@@ -810,21 +816,6 @@
; ; accept both tcp and udp. Default is udp. The first transport
; ; listed will always be used for outgoing connections.
-;------------------------------------------------------------------------------
-; Definitions of locally connected SIP devices
-;
-; type = user a device that authenticates to us by "from" field to place calls
-; type = peer a device we place calls to or that calls us and we match by host
-; type = friend two configurations (peer+user) in one
-;
-; For device names, we recommend using only a-z, numerics (0-9) and underscore
-;
-; For local phones, type=friend works most of the time
-;
-; If you have one-way audio, you probably have NAT problems.
-; If Asterisk is on a public IP, and the phone is inside of a NAT device
-; you will need to configure nat option for those phones.
-; Also, turn on qualify=yes to keep the nat session open
;
; Because you might have a large number of similar sections, it is generally
; convenient to use templates for the common parameters, and add them
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