Hi
I am having an issue with my SIP connection whereby when a call goes in I can see everything is working correctly but when the call gets transferred to a call center agent the RTP stream is gone.
When the call goes to the IVR it is working correctly but when the call goes to the agent the SIP server isn’t displaying nether is the port it is using. On the wire shark trace calls are flowing between port 50046 and 5246. From the gateway there is voice from the PBX there is nothing.
The persistent NAT are disabled:
no ip nat service sip udp port 5060
no ip nat service sip tcp port 5060
I have attached a screenshot from a one call that is captured on the PBX:
What could be the problem and how can i tell if there is a route that is blocking it or if there is a way to see if the RTP stream is correct.
