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authoroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2007-12-19 08:57:45 +0000
committeroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2007-12-19 08:57:45 +0000
commitf93a8656aa75831a8bce3a72757914e30c5fda99 (patch)
tree56f51d46c980f75be5e3a9f311d6da01cd392ce1 /configs
parent7c309cc622bcc5f5ae6aa72ab0daf5be574476b1 (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.sample42
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