I am wondering if anyone knows a solution to this issue?
My Asterisk server is registered with a SIP proxy. On incoming calls, I always get the following symptoms: the first few seconds when the welcome wav file is played, the sound seems to be chopped. In the ‘full’ logfile I can see the following:
It seems to oscillate between the codecs 2 (gsm) and 4 (ulaw).
Often, the first incoming call after Asterisk starts up, the oscillating keeps happening for the entire duration of the call (this call is useless). On all next calls, it only happens in the first few seconds, but it happens every time and it garbles my welcome message.
It seems to start at the exact moment where I background a wav-file, I’ve tried adding a wait period before it, so it could settle down before using the call, but that didn’t help.
I would rather not disable these codecs; does anyone know this problem?