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@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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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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Official FreeSWITCH Sites
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http://wiki.freeswitch.org
http://www.cluecon.com

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