Is pack2packet bridging the process that bypasses the asterisk server and direct-connects the phone to my voip provider? I’m getting intermittent one-way audio problems with my carrier, and noticed when I dont see “packet2packet bridging” is when the problem occurs.
I am using IAX to connect to my voip provider, and using a polycom sip hard-phone.
-Erich Stein
I believe Packet2Packet bridging is the mode asterisk goes into when the stars align. This means the same technology, codec, packet size, etc on both ends of the call. I believe if asterisk needs to do transcoding you won’t see that… although i could be wrong.
One way audio is generally a result of a re-invite where asterisk is attempting to step out of the audio path of the call.
You could try setting up your sip peers to have reinvite=no to see if that fixes things.
I thoguht it was “canreinvite=no” was that bypass command?
-E
canreinvite=no forces everything to go through asterisk. packet to packet bridging is not the same as external bridging.