installed flexisip (Centos 7) to work with me freeswitch (Ubuntu 12.xx LTS) server.
Linphone Softphone --> internet --> Flexisip Proxy --> internet --> Freeswitch --> PSTN
install Flexisip without error and is up and running. However caller
and callee get rings but upon answered cannot hear each other voices in
Codec used is G.711u/a
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.
Also i had enable firewalls UDP port 5060, 10000-20000 plus had ensure that Freeswitch and Flexisip both are trusted with each other.
If i connect Linphone to Freeswitch directly (Diagram B), media is pass correctly - caller and callee can hear each other voice.
Linphone Softphone --> internet --> Freeswitch --> PSTN
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
Thanks and look forward to hearing from all.
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