soft hangup channelname
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Casey
Sent: Wednesday, July 18, 2007 3:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Force SIP hang up.
Is there a way to hang up on
Can't help you with the cause but I can tell you that you can use the
soft hangup command to kill those channels instead of restarting.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael J.
Liberatore
Sent: Thursday, July 12, 2007 3:56 AM
To: asterisk-users@lists.digium.com
use the safe_asterisk script
it will restart asterisk if it crashes and it enables core dumps (your core
size limit is probably set to 0 when you start asterisk).
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Tuesday, June 26, 2007 2:22 PM
To: Asterisk
Turn off debug
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Friday, June 22, 2007 3:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: [asterisk-users] 1.4.5
I am seeing a peculiar message on my console screen
Enable verbose logging for the asterisk log
Set verbose level to 4
Review the log file for anything that looks like a phantom call.
There should be enough information to get some idea of why this is
happening.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
try enabling rtcachefriends
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Forrest Beck
Sent: Friday, May 04, 2007 9:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP RealTime Friends
Let me check my
Perhaps the context in sip.conf doesn't match the context in the dial plan.
From: [EMAIL PROTECTED] on behalf of Mr. Jones
Sent: Fri 8/11/2006 2:34 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Inbound Calls SIP/2.0 404 Not Found
I'm trying
DIALEDPEERNUMBER contains the exact peer spec for the peer that picked up.
You can use that.
From: [EMAIL PROTECTED] on behalf of Koopmann, Jan-Peter
Sent: Wed 8/2/2006 1:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] cmd DIAL - Who picked up the call?
On Wednesday, August 02, 2006 7:39 PM Vadim Berezniker wrote:
DIALEDPEERNUMBER contains the exact peer spec for the peer that
picked up. You can use that.
Consider yourself
[EMAIL PROTECTED] wrote:
Vadim Berezniker wrote:
That's not a solution, but just a workaround.
1.2.1 has a bug where it always uses an empty context when searching
for a mailbox when using realtime config.
At around line 546 of apps/app_voicemail.c there is a line that says
var
I wrote a connection pooling patch because asterisk is not usable with
MSSQL without it.
If you're using, or would like to use, MSSQL I recommend you to check it
out.
http://bugs.digium.com/file_download.php?file_id=8809type=bug
Just so you know, this is a diff against 1.2.1 and it's been only
[EMAIL PROTECTED] wrote:
On Monday 16 January 2006 09:02, [EMAIL PROTECTED] wrote:
Put in voicemail.conf searchcontexts=yes
and do not forget to stop and start *.
Reload may not do.
benchev
That's not a solution, but just a workaround.
1.2.1 has a bug where it always uses an empty context
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