Hi guys, I'm using Asterisk 1.4 and i need a way to hangup all channels.
I want to use the teleyapper system for broadcasting call for security reason
but i need that all channels are free when a security call is ready to start!
I already search in the old post without success.
Can anyone help
Hi,
why have many files on
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ the
change date 18 aug 2009? See:
asterisk-1.2.24-patch.gz07-Aug-2007 17:10 3.2K
asterisk-1.2.24-patch.gz.asc07-Aug-2007 17:10 1.1K
asterisk-1.2.24-patch.gz.sha1 07-Aug-2007
On Sat, 27 Nov 2010, Giuseppe D'alessio wrote:
Hi guys, I'm using Asterisk 1.4 and i need a way to hangup all channels.
1) sudo /etc/init.d/asterisk restart
2) Write a script to do asterisk -r -x 'core show channels', parse the
output and do asterisk -r -x 'channel request hangup
Can also do asterisk -r -x 'restart now'
asterisk*CLI help restart
restart gracefully Restart Asterisk gracefully
restart now Restart Asterisk immediately
restart when convenient Restart Asterisk at empty call volume
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
Hi,
After upgrading to DAHDI 2.4.0 from Zaptel, I noticed a lot of HDLC
errors on my console:
[Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Bad
FCS (8) on Primary D-channel of span 1
[Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Abort
(6) on Primary D-channel
On Sat, 27 Nov 2010, Ali Khalfan wrote:
I'm trying to find a way to setup a SIP server that will mainly echo
back a request from one agent only,
my question is do i need to setup any of the other conf files besides
extensions.conf and sip.conf?
0) Posting the same request an hour later
Original Message
Subject: Re: [asterisk-users] sip echo server
From: Steve Edwards asterisk@sedwards.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sat 27 Nov 2010 12:43:57 PM EST
On Sat, 27 Nov 2010, Ali Khalfan
From: Steve Edwards asterisk@sedwards.com
I'm still not sure what you are asking for. Are you wanting to pick up a
SIP [hard|soft] phone, dial an extension and hear yourself talk? If so,
the relevant configuration files will be:
On Sat, 27 Nov 2010, Ali Khalfan wrote:
yes, this is
Hi, its possible to mantain the original
CallerId when making transfers? (atx or blind)
Example: A calls to B, A transfer to C, C see the CallerID of B,
and not A...
Its possible?
Thanks1
--
Hi List,
Anybody any ideas on these?
[Nov 26 15:14:10] WARNING[3265] chan_sip.c: Remote host can't match request
NOTIFY to call '1c4890a52552c39b0b81702353087...@192.168.33.12'. Giving up.
[Nov 26 15:16:44] WARNING[3265] chan_sip.c: Remote host can't match request
NOTIFY to call
On Sat, Nov 27, 2010 at 11:40 AM, Fabiano Carlos Heringer
b...@grupoheringer.com.br wrote:
Hi, it´s possible to mantain the original CallerId when making transfers?
(atx or blind)
Example: A calls to B, A transfer to C, C see the CallerID of B, and not A...
It´s possible?
Asterisk 1.8 added
The AstLinux Team is happy to announce the release of AstLinux 0.7.4.
This is a dual release which allows you to chose between Asterisk 1.4.36
or 1.8.0.
There are several security updates and other improvements. All current
AstLinux users should upgrade as soon as feasible.
One of the more
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