Hi.
I've updated sound files in /var/lib/asterisk/sounds and restarted Asterisk, but it still plays old ones. I've removed old files completely then, but
still no effect. I've searched for some cahce files, but didn't find any of them.
So, could you tell me please, what is an appropriate way
language)
Well, I'm sure the path is correct.
Il 28/01/2011 11:27, Сикорский Сергей ha scritto:
Hi.
I've updated sound files in /var/lib/asterisk/sounds and restarted Asterisk,
but it still plays old ones. I've removed old files completely then,
but still no effect. I've searched for some
I found the problem. It seemed to be something stupid and it really was.
The name of a new file had missing character.
Sorry for this thread :)
28.01.2011 15:15, Сикорский Сергей пишет:
what kind of update have you done? which is your goal?
DTMF commands of our IVR have been modified, so
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ?
??
Sent: Wednesday, December 22, 2010 4:22 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Wise selecting of outgoing channel
Hi.
Hi.
There is /doc/core-en_US.xml in asterisk 1.8 source tree. Is this file
generated from documentation comments of apps/app_*.c files?
And how this file can be used? How can I convert it to pdf/html in order to use
it as applications documentation?
--
15.10.2010 9:40, Warren Selby пишет:
I think this means you need to set a call-limit for each sip peer
Is there any alternative for obsolete call-limit option in 1.6/1.8?
Thanks,
--Warren Selby
On Oct 14, 2010, at 11:36 PM, Matt Darnellmattdarn...@gmail.com wrote:
Warren,
I tried using
