I'm trying to integrate a cisco IP-IP gateway with an asterisk. The Gateway is a Cisco 2851 running the image c2800nm-ipvoice_ivs-mz.124-24.T1.bin, and it acts as an H323 gateway for a Call Manager server 6.1.
Between the IP-IP GW and the asterisk the protocol is SIP.
The asterisk is connected to a PBX as a SIP trunk.
If a call is placed from an IP phone registred in the call manager to a phone in the PBX everything works fine. But if a call is placed from PBX to an IP phone, the IP phone rings even if somebody answers the phone, and finally the call is dropped.
I captured the messages in the router by using the command "debug ccsip messages" and the Gateway doesn't send the OK message to the Asterisk when it recevices the call.
If the gateway receives call, I understand that a SIP flow call must have an INVITE, then the the gateway must send TRYING, RINGING and then OK to the Asterisk after the RTP traffic, but the OK is never sent.
Can enyone help me with this problem. I send the relevant configuration of the Gateway, a network graphic and the debug results.
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