Hi All,
I'm trying to upgrade asterisk server to 1.8.x from my asterisk 1.6,
But when making A Call from SIP Client, I got cli Warning ... and no call
has been made.
My Sip Client is using lib java peers client http://peers.sourceforge.net/
with standard codec PCMU/PCMA
[Jan 28 23:03:32]
You can also Try:
ulimit -c unlimited , then restart asterisk
Juan.
Linux User #441131
On Thu, Jan 26, 2012 at 5:53 PM, Chad Wallace
cwall...@lodgingcompany.comwrote:
On Thu, 26 Jan 2012 10:35:14 -0700
Mike Diehl mdi...@diehlnet.com wrote:
While trying to track down a T.38 issue, I came
Hi all,
I'm working with the Digium fax for Asterisk product, which is working pretty
reliably for me.
However, the sendfax application isn't sending status events to AMI. The
receivefax application does.
For example, with this call file:
Channel:
I run asterisk from inittab. So, I'd have to create a shell script to do the
ulimit, and then start asterisk. Is there any reason NOT to launch a shell
script from inittab?
On Thursday 26 January 2012 3:53:42 pm Chad Wallace wrote:
On Thu, 26 Jan 2012 10:35:14 -0700
Mike Diehl
Thanks
Any more ideas?
אייל מהלל
מתכנת IVR
משרד: 03-6034293 שלוחה 220
נייד: 054-4793007
פקס: 03-6006081
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Johan Wilfer
Sent: Thursday, January 26, 2012 11:24 AM
To:
You can use controlplayback
On Jan 25, 2012 9:00 PM, Eyal e...@mcr-m.com wrote:
Hi,
How can I play a sound file from the middle and end it after a certain
number of seconds?
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