Can someone show me where to get started? It’s not a case or one way audio, but rather non consistent audio. Using bandwidth.com as my SIP provider, I make a call to a regular phone, say my home phone. I can hear the other end, and vice versa, but about every 10-30 seconds, audio drops out both ways. It comes back after a period of about 3 seconds or so. Using a Grandstream BT100. The correct ports are being forwarded (with bandwidth.com you forward all your incoming UDP ports 1-65535 to asterisk)
Where to start?
edit: using IAX2 and SIP with Voicepulse I get the same issue so we can rule out the VoIP provider as the culprit.
edit 2: Just turned RTP debugging on. For the portions of the conversation that were Ok, I got the typical Got, Sent, Got, Sent, etc from the client. then when the audio drops, all I get are a ton of “Sent RTP packet to” messages followed by nothing but a bunch of “Got RTP packet from” messages and then we go back to normal (got, sent, got, sent) which is when audio starts working again. Thoughts?
edit 3: Same issue with both XLite and the hardphone.