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Updating docs

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.6.2
Olle Johansson 17 years ago
parent 3873227e9d
commit 007807bf41
1 changed files with 87 additions and 96 deletions
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181
configs/sip.conf.sample
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@ -43,13 +43,11 @@
; ------------------------------------------------------------- ; -------------------------------------------------------------
; Useful CLI commands to check peers/users: ; Useful CLI commands to check peers/users:
; sip show peers Show all SIP peers (including friends) ; 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 show registry Show status of hosts we register with
; ;
; sip set debug Show all SIP messages ; sip set debug Show all SIP messages
; ;
; module reload chan_sip.so Reload configuration file ; module reload chan_sip.so Reload configuration file
; Active SIP peers will not be reconfigured
; ;
; ** Deprecated configuration options ** ; ** Deprecated configuration options **
@ -380,15 +378,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; more database transactions if you are using realtime. ; more database transactions if you are using realtime.
;callcounter = yes ; Enable call counters on devices. This can be set per ;callcounter = yes ; Enable call counters on devices. This can be set per
; device too. ; 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 ----------------------- ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
; ;
@ -438,7 +427,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; unless you configure a [sip_proxy] section below, and configure a ; unless you configure a [sip_proxy] section below, and configure a
; context. ; context.
; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com] ; 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 ; (instead of type=friend) if you have calls in both directions
;registertimeout=20 ; retry registration calls every 20 seconds (default) ;registertimeout=20 ; retry registration calls every 20 seconds (default)
@ -703,75 +692,92 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Peer auth= override all other authentication settings if we match on realm ; Peer auth= override all other authentication settings if we match on realm
;------------------------------------------------------------------------------ ;------------------------------------------------------------------------------
; Users and peers have different settings available. Friends have all settings, ; DEVICE CONFIGURATION
; since a friend is both a peer and a user ;
; ; The SIP channel has two types of devices, the friend and the peer.
; User config options: Peer configuration: ; * The type=friend is a device type that accepts both incoming and outbound calls,
; -------------------- ------------------- ; where Asterisk match on the From: username on incoming calls.
; context context ; (A synonym for friend is "user"). This is a type you use for your local
; callingpres callingpres ; SIP phones.
; permit permit ; * The type=peer also handles both incoming and outbound calls. On inbound calls,
; deny deny ; Asterisk only matches on IP/port, not on names. This is mostly used for SIP
; remotesecret ; trunks.
; secret secret ;
; md5secret md5secret ; For device names, we recommend using only a-z, numerics (0-9) and underscore
; transport transport ;
; dtmfmode dtmfmode ; For local phones, type=friend works most of the time
; canreinvite canreinvite ;
; nat nat ; If you have one-way audio, you probably have NAT problems.
; callgroup callgroup ; If Asterisk is on a public IP, and the phone is inside of a NAT device
; pickupgroup pickupgroup ; you will need to configure nat option for those phones.
; language language ; Also, turn on qualify=yes to keep the nat session open
; allow allow ;
; disallow disallow ; Configuration options available
; insecure insecure ; --------------------
; trustrpid trustrpid ; context
; progressinband progressinband ; callingpres
; promiscredir promiscredir ; permit
; useclientcode useclientcode ; deny
; accountcode accountcode ; secret
; setvar setvar ; md5secret
; callerid callerid ; remotesecret
; amaflags amaflags ; transport
; call-limit call-limit (deprecated) ; dtmfmode
; callcounter callcounter ; canreinvite
; allowoverlap allowoverlap ; nat
; allowsubscribe allowsubscribe ; callgroup
; allowtransfer allowtransfer ; pickupgroup
; subscribecontext subscribecontext ; language
; videosupport videosupport ; allow
; maxcallbitrate maxcallbitrate ; disallow
; rfc2833compensate mailbox ; insecure
; session-timers busylevel ; trustrpid
; progressinband
; promiscredir
; useclientcode
; accountcode
; setvar
; callerid
; amaflags
; callcounter
; busylevel
; allowoverlap
; allowsubscribe
; allowtransfer
; subscribecontext
; template
; videosupport
; maxcallbitrate
; rfc2833compensate
; mailbox
; session-timers
; session-expires ; session-expires
; session-minse template ; session-minse
; session-refresher fromdomain ; session-refresher
; t38pt_usertpsource regexten ; t38pt_usertpsource
; fromuser ; regexten
; host ; fromdomain
; port ; fromuser
; qualify ; host
; defaultip ; port
; defaultuser ; qualify
; rtptimeout ; defaultip
; rtpholdtimeout ; defaultuser
; sendrpid ; rtptimeout
; outboundproxy ; rtpholdtimeout
; rfc2833compensate ; sendrpid
; callbackextension ; outboundproxy
; registertrying ; rfc2833compensate
; session-timers ; callbackextension
; session-expires ; registertrying
; session-minse ; timert1
; session-refresher ; timerb
; timert1 ; qualifyfreq
; timerb ; t38pt_usertpsource
; qualifyfreq ; contactpermit ; Limit what a host may register as (a neat trick
; t38pt_usertpsource ; contactdeny ; is to register at the same IP as a SIP provider,
; contactpermit ; Limit what a host may register as (a neat trick ; ; then call oneself, and get redirected to that
; contactdeny ; is to register at the same IP as a SIP provider, ; ; same location).
; ; then call oneself, and get redirected to that
; ; same location).
;[sip_proxy] ;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup) ; For incoming calls only. Example: FWD (Free World Dialup)
@ -810,21 +816,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; ; accept both tcp and udp. Default is udp. The first transport ; ; accept both tcp and udp. Default is udp. The first transport
; ; listed will always be used for outgoing connections. ; ; 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 ; Because you might have a large number of similar sections, it is generally
; convenient to use templates for the common parameters, and add them ; convenient to use templates for the common parameters, and add them

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