Merged revisions 168711 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

........
r168711 | oej | 2009-01-15 18:55:53 +0100 (Tor, 15 Jan 2009) | 4 lines

Clarify some misunderstandings and make it even more clear that you can refer to a peer
in the register= line.


........


git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@168715 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 8 insertions, 4 deletions

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index a7293b239..513d45418 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -404,17 +404,21 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;----------------------------------------- OUTBOUND SIP REGISTRATIONS  ------------------------
 ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
 ; Format for the register statement is:
-;       register => [transport://]user[:secret[:authuser]]@host[:port][/extension][~expiry]
+;       register => [transport://]user[:secret[:authuser]]@domain[:port][/extension][~expiry]
 ;
 ; 
 ;
+; domain is either 
+;	- domain in DNS
+; 	- host name in DNS
+;	- the name of a peer defined below or in realtime
+; The domain is where you register your username, so your SIP uri you are registering to 
+; is username@domain
+;
 ; If no extension is given, the 's' extension is used. The extension needs to
 ; be defined in extensions.conf to be able to accept calls from this SIP proxy
 ; (provider).
 ;
-; host is either a host name defined in DNS or the name of a section defined
-; below.
-;
 ; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
 ; this is equivalent to having the following line in the general section:
 ;