[Freeswitch-users] No audio on either of an established call

[Michael Collins]

I'd start by getting a debug log of the call and posting it on
pastebin.freeswitch.org. Hopefully that will yield some clues.
-MC


On Wed, Jul 31, 2013 at 7:33 PM, Peter <eidevm5 at gmail.com> wrote:

> I currently have 2 SIP clients (Linphone) successfully calling each other,
> but there is no audio on either end.
>
> The set up is as follows:
>
>
>
> Linphone1 (1000)  --> Kamailio 1   <-------> Freeswitch   <------>
> Kamalio 2  <---- Linphone2 (2000)
>
>
> Using Freeswitch 1.2.12 on CentOS (installed via RPM)
>
> Freeswitch has two interfaces:
>
> external - 10.1.1.206
> internal -  10.10.10.206
>
> Each of the Linphone clients are registered their respective Kamailio
> instance and Kamailio is configured to route via the appropriate interface
> on Freeswitch.
>
> The SIP negotiation is working as I can call either Linphone client.
>
> I've done a tcpdump on each side of  Freeswitch and can see the RTP
> traffic between the Linphone and the appropriate interface on Freeswitch.
>
> I've tried different codec combinations (mostly G711 and iLBC), and
> different SIP clients but still get no audio.
>
> Any pointers on how to track down the issue?
>
> Thanks
>
> Peter
>
>
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> 
> 
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-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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