We can lab it up and see if we can replicate it.
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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@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@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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_________________________________________________________________________
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