This is driving me crazy, when making an outgoing call the first 30 seconds is always perfect, then the party on the receiving end can always hear me perfectly but after 30-60 seconds the audio coming back to me from them starts to get choppy and drops out.
I've tried this with multiple devices, from multiple locations some behind NAT, others not. This is using the ulaw codec, although i've tried it with alaw as well. Problem happens via IAX as well as a SIP channel, both calling PSTN numbers. Network performance to the Asterisk server is good, 15-20ms, performed ping tests for 1-2 hours with almost no packet loss. I'm willing to check or post anything needed here, but I need some fresh ideas since i've checked everything I can. Regards, -J _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
