Asterisk dropping audio every 10 seconds or so

Can someone show me where to get started? It’s not a case or one way audio, but rather non consistent audio. Using 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 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.

What codec are you using?

What are your server specs? Can you verify server resources aren’t being maxed out for some odd reason?

ulaw and athlon XP 1700+ with 1GB of RAM. Top shows both RAM and CPU are doing OK.


OK. Even more weird. I can log into my voicemail on the server and listen all day to the menus and never lose audio once. WTFM8? It must be something either with my internet connection or my router itself. How can I check those?

bump again. I plan on taking my server home this weekend. I use the same internet carrier at home as work so maybe that will act like some sort of control variable. We shall see.