On Tue, Jul 05, 2011 at 08:30:52PM -0400, A E [Gmail] wrote:
hello people,
I am running v1.8.4.2 on debian squeeze on a sparc platform...and for some
reason I have noticed that only after a few test calls, the asterisk process
is running between 95% - 99.9% CPU when there's absolutely
On Tue, Jul 05, 2011 at 04:07:28PM +0200, Tobias Steen wrote:
It seems that the full distro package from FreePBX with Asterisk 1.8.1.4
someway hides (deletes?) the source directory for asterisk after
installation.
Is it installed from rpm packages? If so: are those available on-line?
If so:
Hello all
I'm running Asterisk 1.8.4.4 in a new installation. I'm seeing peculiar
behaviour in a context where I dispatch to different MeetMe conference
rooms. It seems the first digit is being given to Asterisk and it ALWAYS
jumps to the i extension. I originally had single digits for the MeetMe
You can't use WaitExten to receive two digits. Use Read() command.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith
Sent: Wednesday, July 06, 2011 1:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi All;
I know that incoming calls for the agent can be recorded, but how I can let the
outbound calls for the agents to be recorded? I can determine the directory to
store the outbound calls of the agents to be other than the directory to store
the incoming calls of the agents?
Regards
Bilal
On Wed, Jul 6, 2011 at 3:21 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Tue, Jul 05, 2011 at 08:30:52PM -0400, A E [Gmail] wrote:
hello people,
I am running v1.8.4.2 on debian squeeze on a sparc platform...and for
some
reason I have noticed that only after a few test calls, the
Hi, can anyone help with this?
Thanks
Lee
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: 05 July 2011 16:27
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Recording SIP history
Hi all, can
Hi All;
I know that incoming calls for the agent can be recorded, but how I can let
the outbound calls for the agents to be recorded? I can determine the
directory to store the outbound calls of the agents to be other than the
directory to store the incoming calls of the agents?
On Wed, Jul 06, 2011 at 06:15:26AM -0400, A E [Gmail] wrote:
On Wed, Jul 6, 2011 at 3:21 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Tue, Jul 05, 2011 at 08:30:52PM -0400, A E [Gmail] wrote:
hello people,
I am running v1.8.4.2 on debian squeeze on a sparc platform...and for
On Wed, Jul 6, 2011 at 7:02 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Wed, Jul 06, 2011 at 06:15:26AM -0400, A E [Gmail] wrote:
On Wed, Jul 6, 2011 at 3:21 AM, Tzafrir Cohen tzafrir.co...@xorcom.com
wrote:
On Tue, Jul 05, 2011 at 08:30:52PM -0400, A E [Gmail] wrote:
hello
On Wed, Jul 06, 2011 at 07:11:26AM -0400, A E [Gmail] wrote:
On Wed, Jul 6, 2011 at 7:02 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Wed, Jul 06, 2011 at 06:15:26AM -0400, A E [Gmail] wrote:
- when I'd run an strace on the PID of the offending thread it just rolled
some message
Hello
I was wondering if Asterisk can be configured to monitor a
connection to a VoIP provider, whether someone is currently using it
for a call or the connection is idle?
FWIW, my VoIP provider doesn't run an iperf server on their side. I
don't know if ping/traceroute is a good enough
On Wed, Jul 6, 2011 at 7:50 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Wed, Jul 06, 2011 at 07:11:26AM -0400, A E [Gmail] wrote:
On Wed, Jul 6, 2011 at 7:02 AM, Tzafrir Cohen tzafrir.co...@xorcom.com
wrote:
On Wed, Jul 06, 2011 at 06:15:26AM -0400, A E [Gmail] wrote:
- when
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Cassius Smith
Sent: Wednesday, July 06, 2011 4:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] single keypress
On 7/6/11 3:20 PM, Eric Wieling ewiel...@nyigc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Cassius Smith
Sent: Wednesday, July 06, 2011 4:37 AM
To: Asterisk Users Mailing List -
Using a Polycom 650 with 3.3.1, I could not have Directed Pickup working.
More precisely, I configured the phone using call and attendant entries
as described in this thread.
Whenever a call comes in, BLF is blinking green.
Pressing the associated key generate generates a general Call Pickup
The Asterisk Development Team announces the release of libpri version
1.4.12. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/libpri/
The following are some of the issues resolved in this release:
* Add call transfer exchange of subaddresses
So I made the change you suggested. That still hasn't worked, but I did manage
to grab some logging from a dropped call.
[Jul 6 16:19:37] DEBUG[25950] channel.c: Got a FRAME_CONTROL (8) frame on
channel DAHDI/i1/18883203585-7e
[Jul 6 16:19:37] DEBUG[25950] res_rtp_asterisk.c: Setting the
IMHO, blind tranfer definition is to NOT connect A and B back
That is correct, and is why it's called a 'blind' transfer;
the transferring party does not know or care what happens to
the call after effecting the transfer.
That's not what users migrating from some legacy PBXs expect,
Hi All;
The asterisk version I am using is 1.8.4.2 and I compiled ooh323 channel (by
selecting the add-on). But really does not work in good performance, for
example: if a call came from gnugk to asterisk and the ooh323 handled it, the
performance is bad .. some calls are drop and if it is
On 07/06/2011 04:44 PM, Alec Davis wrote:
IMHO, blind tranfer definition is to NOT connect A and B back
That is correct, and is why it's called a 'blind' transfer;
the transferring party does not know or care what happens to
the call after effecting the transfer.
That's not what users
On 7/6/2011 4:36 AM, bilal ghayyad wrote:
Hi All;
I know that incoming calls for the agent can be recorded, but how I can let the
outbound calls for the agents to be recorded? I can determine the directory to
store the outbound calls of the agents to be other than the directory to store
the
On 07/06/2011 05:52 PM, Kevin P. Fleming wrote:
On 07/06/2011 04:44 PM, Alec Davis wrote:
IMHO, blind tranfer definition is to NOT connect A and B back
That is correct, and is why it's called a 'blind' transfer;
the transferring party does not know or care what happens to
the call after
Community can help you better if you provide some details about you scenario
and requirement.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Wednesday, July 06, 2011 5:03 PM
To:
Hi,
As per my experience YATE is the best option for H323=SIP Proxy.
Regards,
Faisal
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Thursday, July 07, 2011 2:48 AM
To:
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