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