Hi everyone, I have been working on incoming calls from a sip trunk, and debugging potential issues. Right now, calls are disconnected immediately after I dial an extension from the AA (when I call externally). I'm pretty sure the NAT is configured properly, and I'm starting to narrow down the problem. The NAT uses nf_conntrack_sip rather than explicitly opening RTP ports. I used tcpdump to monitor incoming calls, and I find events such as (right before disconnection):
19:40:25.689135 IP bm-srv-01.voicenetwork.ca > 123.456.1.12: ICMP bm-srv-01.voicenetwork.ca udp port 19222 unreachable, length 208 I have discussed this with a friend, and one potential issue could be how the phone network is configured. My phones are firewalled so that they can only communicate with the SipX server. I am not sure if the transfer negotiation is attempting to pass the connection directly to the phone, which then has no path back (and is not really reachable from the NAT system). Any suggestions? AJ _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
