[asterisk-commits] oej: branch 1.6.1 r172218 - in /branches/1.6.1: ./ configs/sip.conf.sample

[SVN commits to the Asterisk project]

Author: oej
Date: Thu Jan 29 04:14:16 2009
New Revision: 172218

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=172218
Log:
Merged revisions 171880 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

........
r171880 | oej | 2009-01-28 14:26:31 +0100 (Ons, 28 Jan 2009) | 2 lines

Add some more notes about device matching.

........

Modified:
    branches/1.6.1/   (props changed)
    branches/1.6.1/configs/sip.conf.sample

Propchange: branches/1.6.1/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.

Modified: branches/1.6.1/configs/sip.conf.sample
URL: http://svn.digium.com/svn-view/asterisk/branches/1.6.1/configs/sip.conf.sample?view=diff&rev=172218&r1=172217&r2=172218
==============================================================================
--- branches/1.6.1/configs/sip.conf.sample (original)
+++ branches/1.6.1/configs/sip.conf.sample Thu Jan 29 04:14:16 2009
@@ -58,12 +58,18 @@
 ;	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
+;	2. Asterisk checks the From: addres and matches the list of devices
+;	   with a type=peer
+;	3. 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).
+; 
+; When setting up trunks, make sure there's no risk that any From: username
+; (caller ID) will match any of your device names, because then Asterisk 
+; might match the wrong device.
 ;
 ; 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
@@ -445,6 +451,20 @@
 ;
 ; and more readable because you don't have to write the parameters in two places
 ; (note that the "port" is ignored - this is a bug that should be fixed).
+;
+; Note that a register= line doesn't mean that we will match the incoming call in any
+; other way than described above. If you want to control where the call enters your
+; dialplan, which context, you want to define a peer with the hostname of the provider's
+; server. If the provider has multiple servers to place calls to your system, you need
+; a peer for each server.
+;
+; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may
+; contain a port number. Since the logical separator between a host and port number is a
+; ':' character, and this character is already used to separate between the optional "secret"
+; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish
+; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
+; they are blank. See the third example below for an illustration.
+;
 ;
 ; Examples:
 ;