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:
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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