On Saturday 10 October 2009 21:45:37 RSCL Mumbai wrote:
Can I convert my .WAV IVR greetings, MOH and other recordings into G729
format to prevent transcoding and hence CPU usage ?
You can do it with a one-time conversion. This will require a single g729
license:
*CLI file convert foo.wav
On 3 Oct 2009, at 20:38, James Stocks wrote:
On 3 Oct 2009, at 16:37, Jonathan Thurman wrote:
On Sat, Oct 3, 2009 at 6:17 AM, James Stocks stoc...@stocksy.co.uk
wrote:
Hi everyone,
I hope someone can help me with a problem I'm having with Cisco 7940
phones on the SIP 8.12 image. When I
On Sun, 11 Oct 2009 09:10:27 -0500, Tilghman Lesher wrote:
On Saturday 10 October 2009 21:45:37 RSCL Mumbai wrote:
Can I convert my .WAV IVR greetings, MOH and other recordings into G729
format to prevent transcoding and hence CPU usage ?
You can do it with a one-time conversion. This will
On Sun, Oct 11, 2009 at 8:03 AM, James Stocks stoc...@stocksy.co.uk wrote:
OK. For anyone finding this thread, the problem exists in Asterisk
1.4, but upgrading to Asterisk 1.6.1.6 appears to eliminate the problem.
Sorry, I lost your last response in my inbox... Your phone configs
look fine.
- Original Message -
From: Philipp Kempgen philipp.kemp...@amooma.de
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, October 05, 2009 10:39 PM
Subject: Re: [asterisk-users] Zaptel problems on SuSE 9.3
Angus Asterisk schrieb:
Hello,
I'm working on a call recording solution. I would like recordings to
either be automatically uploaded via FTP, or posted to a URL for
processing by our main server.
Is Asterisk capable of doing this or will I have to create a separate
application that monitors a temp directory for
I just double checked the setting of the remote asterisk and it has the same
setting as mine.
Sip.conf has in Global:
dtmfmode = rfc2833
individual extension has no dtmf setting at all, so global setting take
precedence.
All units Linksys, Sipura have
DTMF Tx Method: Auto
Linksys has an
The Grandstream 286s automatically re register when a connection is
restored.
Our Grandstream 2010s don't. Does anyone know of a setting that makes them
reregister? I has tweaked Watchdog timer and anything that looked
promising.
Cary Fitch
Affordable Telecom
Dan,
You can do this directly in the dialplan. See the System command. It allows
you to call any program on the system (ftp, scp, mv, etc). Keep in mind that
depending on the volume of calls you're handling, you might run into I/O issues
on the disk side. If you're talking about a machine
On Mon, 12 Oct 2009, Dan Journo wrote:
I'm working on a call recording solution. I would like recordings to
either be automatically uploaded via FTP, or posted to a URL for
processing by our main server.
Is Asterisk capable of doing this or will I have to create a separate
application that
We are running Asterisk 1.4 and need some help to determine how (if) *
supports 3 party warm transfers. I've searched quite a bit and all I
can find is information on attended transfers. What we are looking
for is: (1) external inbound call A comes to * extension B, caller A is
placed on hold
I don't think this can be done.
In your scenario, B is effectively the host and if B drops the line, both A and
C will be dropped off as well.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
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