From 2c4f19eb2c4fac892d3a2e912e639fcde4386724 Mon Sep 17 00:00:00 2001 From: Olle Johansson Date: Wed, 28 Jan 2009 13:11:44 +0000 Subject: Merged revisions 171837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2 lines Add a better explanation of the difference between the device namespace and the dialplan for newbies. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171838 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- configs/sip.conf.sample | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) (limited to 'configs') diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index c365b0185..d169cc22d 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -49,6 +49,25 @@ ; ; module reload chan_sip.so Reload configuration file ; +;------- Naming devices ------------------------------------------------------ +; +; When naming devices, make sure you understand how Asterisk matches calls +; that come in. +; 1. Asterisk checks the SIP From: address username and matches against +; names of devices with type=user +; The name is the text between square brackets [name] +; 2. Asterisk checks the IP address (and port number) that the INVITE +; was sent from and matches against any devices with type=peer +; +; Don't mix extensions with the names of the devices. Devices need a unique +; name. The device name is *not* used as phone numbers. Phone numbers are +; anything you declare as an extension in the dialplan (extensions.conf). +; +; Note: The parameter "username" is not the username and in most cases is +; not needed at all. Check below. In later releases, it's renamed +; to "defaultuser" which is a better name, since it is used in +; combination with the "defaultip" setting. +;----------------------------------------------------------------------------- ; ** Deprecated configuration options ** ; The "call-limit" configuation option is deprecated. It still works in -- cgit v1.2.3