I have a 908e with FXS 0/1-4 connected to the analog trunks of an AVAYA IPO 500V2. When I call any of my 4 SIP DIDs I intermittently get no-ring and a busy signal (it does occasionally pick up just fine). The AVAYA system shows that the line has been answered and has been handed off to the auto attendant. Eventually the auto attendant times out and the call is handed off to a hunt group. When the user picks up the line disconnects. This whole time the call through the Adtran has a busy signal.
Any ideas as to why this is happening?
I have attached the debug output from the Adtran using the following commands:
debug sip stack messages
debug voice verbose
debug interface fxs
Your debug looks like the call is somehow looping back into the TA908e. Could there be a forwarding loop in the PBX or the call is somehow being redirected back to the box?
Forwarding looks to be user based on the IPO and no one has any forwarding set. I have attached the line settings from the IPO for one of the incoming lines that is plugged into the Adtran. The other 3 are identical. The SIP lines are on their own incoming call route right now (Group ID 1) and anything incoming gets sent to the auto attendant via a short code (same as all other lines).
If you connect a phone or but set to the FXS port of the Adtran and call does the voice work then? If it does then it would be a port configuration/incompatibility issue on the PBX side i.e. PBX ports should be Office facing ports FXO (or in some cases the are referred to as Off Network depending on PBX vendor). If it doesn't I would suspect an RTP negotiation issue with the SIP provider things like Codec Mismatch, RTP vs SRTP, you should be able to have the ITSP check and see if they are getting mismatch with the RTP stream.