Phones Polycom IP650 and Spectralink 8002
I’ve seen this posted in numerous other areas but no fix shown.
A good example of what we’re experiencing is described here:
mail-archive.com/asterisk@uc … 05938.html
However we’re doing SIP<->SIP not Zap<->SIP
When a caller calls in over a SIP trunk to our ITSP (Who’s using Metaswitch) the calls hit the IVR just fine, the caller can hear the IVR and can select options. However if they select an option to dial a ring group and we pickup there’s a 2-3 second delay in audio, the call shows connected, I can even see RTP flowing between the phone and the asterisk server but there’s not actual audible audio.
I’m at a complete loss on what’s happening here, it doesn’t make sense but it’s almost like asterisk is taking a second or two to sync up RTP between the ITSP and the phone (Because the IVR works fine).
If anyone has some insight on this please advise, I can share logs etc as needed but there’s nothing really to see because form a networking / packet flow standpoint it all looks right…