...@opensourcedevel.com
To: asterisk-users@lists.digium.com
Sent: Wednesday, June 17, 2009 10:06 PM
Subject: [asterisk-users] Incoming SIP and the 's' extension
Hello, all. My apologies up front but I must be brain cramping on
something very simple. I've tried to pare down my configuration
On Thu, 2009-06-18 at 03:50 +, Joseph L. Casale wrote:
I then fire up twinkle on my desktop and dial sip:3...@pbx.mycompany.com.
The Asterisk console shows:
[Jun 17 22:58:55] NOTICE[32060]: chan_sip.c:18160 handle_request_invite:
Call from '' to extension '36' rejected because extension
Hello, all. My apologies up front but I must be brain cramping on
something very simple. I've tried to pare down my configuration to the
absolute minimum for SIP traffic just to understand how it works. My
incoming calls are not finding the s extension in my dial-plan. I am
assuming SIP calls
I then fire up twinkle on my desktop and dial sip:3...@pbx.mycompany.com.
The Asterisk console shows:
[Jun 17 22:58:55] NOTICE[32060]: chan_sip.c:18160 handle_request_invite:
Call from '' to extension '36' rejected because extension not found.
If I use the same extensions.conf but change s to 36,
