When I make a follow-me or call forward call the rtp stream is not generated. The signaling is perfect but no rtp packets. I can place a call inbound from my cell phone to SIP Cisco 7940 and the rtp packets are sent in both directions. I can also send a call outbound from the sip phone to my cell phone and i always have audio end to end.
I am running Asterisk 1.4.21.2-2. My system is behind a router. I have captured packets on both sides of my internal and external network. No rtp stream is flowing at all. I have exaimed the sdb in the outgoing INVITE and the 200 OK on the return messages. Both sides are requesting and agreeing on G.711.
Can anybody give me some advice? I am will to share additional information about my configuration.
Thank you,
John_2008
Member Since:
2008-12-22