no, it's a set of script that I'm supposed to update. However the result
will be similar.
On 02/04/2012 17:42, Israel Gottlieb wrote:
are you by chance using the a2billing script?
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No, I don't do transcoding, I've disabled all the codec except for the g729.
But in my last test I've found out what is the problem (not yet how to
solve it)
I make all my calls through a php agi, this old script works well on
asterisk 1.4 and I want to move on 1.8.
Just for test I've created
are you by chance using the a2billing script?
On Mon, Apr 2, 2012 at 5:43 PM, Syco syco...@gmail.com wrote:
No, I don't do transcoding, I've disabled all the codec except for the
g729.
But in my last test I've found out what is the problem (not yet how to
solve it)
I make all my calls
On 31/03/2012, at 3:28 AM, Syco wrote:
But if I change the dialplan, remove background and wait functions, add play
with a g729 audio file instead, I could do again just 80 concurrent call.
How many g729 licenses do you have? You sure you're not transcoding?
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Cheers,
Matt Riddell
On 30 Mar 2012, at 10:14, Syco wrote:
Finally the problem is: I cannot manage more than 80 concurrent calls.
What happens on the 81st call?..
S
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things correctly).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Howes
Sent: Friday, March 30, 2012 7:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] concurrent
-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Howes
Sent: Friday, March 30, 2012 7:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] concurrent channels limit
Ok, this was a stupid thing (my fault), with -r 1000 I get easily 1000
concurrent calls that terminate in 20 seconds.
This calls just answer, play a file the first 2 seconds and then wait.
Then sipp close because of two many errors, this is the log:
sipp: The following events occured: