broken audio/one way audio

[giulia...@gmail.com]

Hi everyone,

I have configured a 20 dongle freebpx (Ast 13, Freepbx 13.0.195, calls go thru NAT) that we use between our offices but i'm running into some problems.

Some calls have one way audio or broken audio. I have checked and it doesn't seem to be an issue with the internet connection. I have configured adaptive jitterbuffer on chan_SIP and in some situations the calls were cut at 1750 seconds and the other issue still persists.

I'm not that experienced with freepbx/asterisk.

Can someone help me configure jitterbuffer and AGC on chan_pjSIP or AGC on chan_SIP? I have searched for a solution but i don't understand how to apply the dial plan configuration for those calls on pjsip.

Has anyone experienced this issues? if so please let me know what solution you have found.

Thanks in advance

G