Hi,
I have been investigating an issue with our Asterisk-to-Asterisk setup using IAX to solve the sound quality. The scenario can be shown as follows:
Cisco AS5350 -> Asterisk1 (us) - Asterisk2 (customer) - End-User.
The problem is that everytime there is an incoming call from the PSTN (AS5350), it is forwarded accordingly to the End-User but the voice is choppy/robotic/slow ( I hope you know what I mean). I have been playing around with many settings including disabling VAD in the Cisco, specifically set Asterisk to use internal timer (settings in the sip.conf). None of these works.