I’m using AudioCodes VoIP boxes in our system, with Asterisk as call router. I’m using G723.1, with asterisk configured in pass-through mode. I’m also using a proprietary AudioCodes extension to G723.1, which allows larger frame sizes (60msec and 90 msec). The AudioCodes boxes using the ptime parameter in their SIP INVITE messages.
Unfortunately, Asterisk is discarding that parameter when it sends INVITE messages to the destination, which forces them to default back to the 30msec frame size.
Is there some way I can resolve this omission?? If not, can we modify Asterisk so that it does propagate the “a=ptime:90” parameter??