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[Linphone-developers] Flexisip

From: Jason Benedict Low
Subject: [Linphone-developers] Flexisip
Date: Wed, 19 Apr 2017 13:49:39 +0800

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,

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