[Freeswitch-users] choppy sound on voice recording andconference
Anestis Mavro
devel at omninet.eu
Thu Jan 19 00:02:52 MSK 2012
How can I help on this? I could offer access to the second node, which can
receive calls from the same provider. I did not check its behavior, since it
has just been installed; and this is a Debian installation.
It is not in production.
Should we continue talking "off list" ?
_____
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, January 18, 2012 10:41 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] choppy sound on voice recording
andconference
We can lab it up and see if we can replicate it.
On Wed, Jan 18, 2012 at 1:58 PM, Anestis Mavrofillidis <anestis at omninet.gr>
wrote:
Yes, it is the commercial module. I have bought a few licenses for this
server and I am ready to buy a few for a second node, but this problem is
now critical. even voicemail is not usable! You hear the announcement, but
the recorded message is very bad. (just to remind you: only the incoming
calls on this trunk are affected)
Thanks
Anestis
_____
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, January 18, 2012 8:33 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] choppy sound on voice recording
andconference
Is it our commercial G.729 module or something else? We only have control
over our own code.
On Wed, Jan 18, 2012 at 6:47 AM, Anestis Mavro <devel at omninet.eu> wrote:
Hello,
I have a problem with one of my sip trunks. The provider uses g729 with a
ptime of 40ms and all incoming calls that are being recorded or go to the
conference have a big delay and choppy sound, like a mismatch of ptime.
I have already tried to put rtp-autofix-timing=false into the profiles and
even changed the inbound-codec.. to g729 at 40i, but nothing helps.
We have done a test with the provider switching for a test call to 20ms and
the sound was perfect.
The issue now is that the provider can't keep this setting only for us, he
wants to send with 40ms.
Is there a way to configure it on Freeswitch?
Thank you
Anestis
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http://www.eset.com
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Twitter: http://twitter.com/FreeSWITCH_wire
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