diff options
author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-12-19 08:57:45 +0000 |
---|---|---|
committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-12-19 08:57:45 +0000 |
commit | f93a8656aa75831a8bce3a72757914e30c5fda99 (patch) | |
tree | 56f51d46c980f75be5e3a9f311d6da01cd392ce1 /configs | |
parent | 7c309cc622bcc5f5ae6aa72ab0daf5be574476b1 (diff) |
Adding the ability to specify the To: header in an outbound INVITE
by adding an exclamation mark to the dial string.
This patch also exists for 1.4 in the fixtoheader-1.4 branch
and has been in production for quite some time.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93897 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs')
-rw-r--r-- | configs/sip.conf.sample | 42 |
1 files changed, 31 insertions, 11 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index d8e25e642..78ed4806f 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -1,17 +1,37 @@ ; ; SIP Configuration example for Asterisk ; -; Syntax for specifying a SIP device in extensions.conf is -; SIP/devicename where devicename is defined in a section below. -; -; You may also use -; SIP/username@domain to call any SIP user on the Internet -; (Don't forget to enable DNS SRV records if you want to use this) -; -; If you define a SIP proxy as a peer below, you may call -; SIP/proxyhostname/user or SIP/user@proxyhostname -; where the proxyhostname is defined in a section below +; SIP dial strings +;----------------------------------------------------------- +; In the dialplan (extensions.conf) you can use several +; syntaxes for dialing SIP devices. +; SIP/devicename +; SIP/username@domain (SIP uri) +; SIP/username@host:port +; SIP/devicename/extension +; +; +; Devicename +; devicename is defined as a peer in a section below. +; +; username@domain +; Call any SIP user on the Internet +; (Don't forget to enable DNS SRV records if you want to use this) ; +; devicename/extension +; If you define a SIP proxy as a peer below, you may call +; SIP/proxyhostname/user or SIP/user@proxyhostname +; where the proxyhostname is defined in a section below +; This syntax also works with ATA's with FXO ports +; +; All of these dial strings specify the SIP request URI. +; In addition, you can specify a specific To: header by adding an +; exclamation mark after the dial string, like +; +; SIP/sales@mysipproxy!sales@edvina.net +; +; CLI Commands +; ------------------------------------------------------------- ; 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) @@ -23,7 +43,7 @@ ; Active SIP peers will not be reconfigured ; -; ** Deprecated options ** +; ** Deprecated configuration options ** ; The "call-limit" configuation option is deprecated. It still works in ; this version of Asterisk, but will disappear in the next version. ; You are encouraged to use the dialplan groupcount functionality |