Hi,
I've recently encountered some problem with my SIP service whereby i
call out to a specific number and i encounter a one way voice. If i'm
the initiator, i cannot hear the other party but he can hear me. At
first i thought it was a return route issue (as i'm going thru NAT) , so
i switch my SIP to a public IP but i still face the same problem. Its
really only that specific PSTN number that i have dialed facing this
problem. The only difference that i can think of is that PSTN number is
on a different route. I did a NGREP from my SIP server for the PSTN
number that works (2-way voice) and the Number that doesn't work (1-way
voice) . The only difference is there is an extra :
NGW --> Proxy
SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP
Proxy --> SIP Device
SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP
for the PSTN number that works (the one with 2-way voice).
Anyone has idea what does the Session Progress is for ? Or what problem
am i facing ?
Thanks a mILLION !
Regards,
Sam