Hi, My incomming channels using SIP/g711 and outgoing channels using SIP/g729.
Asterisk Box doing the Trans coding. when the call start to ringing ( outgoing channels SIP/g729). Lots of noises / static. The incoming channels provider mention to me the incoming channels SIP/g711 frame size 20 msec. But may be the destination using the 30 msec frame. This is happen randomly? is it a 20/30 ms frame size mismatch ? Can some one help me to solve this issue. so every 60ms ... 10ms of audio gets dropped. Is there any way to setup the RTP payload in asterisk for g729 codec? Using the asterisk 1.2.16 Thanks Lloyd
