diff options
Diffstat (limited to 'configs/sip.conf.sample')
-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 |