. Johansson o...@edvina.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, November 9, 2010 11:30:27 PM GMT -08:00 US/Canada Pacific
Subject: Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten
CallerID(num) Problem
10 nov 2010 kl
On Tue, 2010-11-09 at 17:38 -0800, Brett Woollum wrote:
Good idea Paul.
My debug output:
[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark
5
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing
[...@sipphones:1] Set(SIP/413-0005, CALLERID(num)=2) in
new
-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 10, 2010 8:47:38 AM GMT -08:00 US/Canada Pacific
Subject: Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten
CallerID(num) Problem
On Tue, 2010-11-09 at 17:38 -0800, Brett Woollum wrote: Good idea Paul.
My debug
-08:00 US/Canada Pacific
Subject: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num)
Problem
Hello,
I am running Asterisk 1.6 with Realtime enabled for my SIP users and peers. The
backend is a MySQL database running through the ODBC backend in Asterisk. At
this point
On Tue, Nov 9, 2010 at 6:35 PM, Brett Woollum br...@woollum.com wrote:
Nobody has any idea why the Caller ID is being overwritten when using
Asterisk Realtime for the SIP users?
No, perhaps you can _show_ us the problem.
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
- Original Message -
From: Paul Belanger paul.belan...@polybeacon.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, November 9, 2010 5:18:36 PM GMT -08:00 US/Canada Pacific
Subject: Re: [asterisk-users] Asterisk 1.6 (realtime
10 nov 2010 kl. 02.38 skrev Brett Woollum:
Good idea Paul.
My debug output:
[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Set(SIP/413-0005, CALLERID(num)=2) in new stack
[Nov 9
Hello,
I am running Asterisk 1.6 with Realtime enabled for my SIP users and peers. The
backend is a MySQL database running through the ODBC backend in Asterisk. At
this point everything works in terms of phones registering, placing calls
between them, etc. However, I am having a problem
