Currently i am deploying ip telephony system. In site a we have sccp and sip phone, in site b there is only sccp phone.
the topology is:
ip phone (sccp and sip) -- CME -- WAN -- CME -- ip phone (sccp)
when i call from sccp ip phone in site a to sccp ip phone in site b, the call works well. but when i try to call from sip ip phone in site a to sccp ip phone in site b, ip phone in site b is ringing, but when picked up , the call is disconnected. i try to run "debug ccsip message" i found disconnect cause 47.
after that, i try to simulate the call. i install new router (as cme) in site a (same site) but in different network segment. i register 1 sccp ip phone to the new cme. then i call from sip ip phone to sccp phone. the call works well.
i ask them to check if there is blocking port for sip. but they said they won't open it (it's related to their policy). they insist us to create transcoding or mtp to resolve this case.
if i configure mtp, is that going to solve the problem?