Hi,
Please excuse me if my post is not for this list, kidly direct me to the correct list.
I install flexisip (Centos 7) to work with me freeswitch (Ubuntu 12.xx LTS) server.
Diagram (A):-
==========
Linphone Softphone --> internet --> Flexisip Proxy --> internet --> Freeswitch --> PSTN
===========
I install Flexisip without error and is up and running. However caller and callee get rings but upon answered cannot hear each other voices in both direction.
Codec used is G.711u/a
If i connect Linphone to Freeswitch directly (Diagram B), media is pass correctly - caller and callee can hear each other voice.
I can see in Freeswitch, Flexisip is registered to it when the Linphone software is register request to Flexisip proxy server
I had enable Flexisip to route media.
Diagram (B):-
===========
Linphone Softphone --> internet --> Freeswitch --> PSTN
===========
Do i need to install G.711u/a codec for Flexisip ? If so please provide me guide to get it work or direct me to where i can have more information about it.
Thanks and look forward to hearing from all.
Best Regards,
Jason
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