Yes, the INVITE is coming in to port 5080. Here is a fragment of the TCP dump...
20:35:20.680594 IP bm-srv-01.voicenetwork.ca.23960 > 123.123.123.123.30500: UDP, length 172 20:35:20.700720 IP 74-51-40-188.voicenetwork.ca.5060 > 123.123.123.123.5080: SIP, length: 974 20:35:20.701954 IP bm-srv-01.voicenetwork.ca.23960 > 123.123.123.123.30500: UDP, length 172 20:35:20.716718 IP 123.123.123.123.5080 > 74-51-40-188.voicenetwork.ca.5060: SIP, length: 743 20:35:20.721350 IP bm-srv-01.voicenetwork.ca.23960 > 123.123.123.123.30500: UDP, length 172 20:35:20.740643 IP bm-srv-01.voicenetwork.ca.23960 > 123.123.123.123.30500: UDP, length 172 20:35:20.760635 IP bm-srv-01.voicenetwork.ca.23960 > 123.123.123.123.30500: UDP, length 172 20:35:20.792415 IP 74-51-40-188.voicenetwork.ca.5060 > 123.123.123.123.5080: SIP, length: 805 20:35:21.399765 IP 123.123.123.123.30500 > bm-srv-01.voicenetwork.ca.23960: UDP, length 172 20:35:21.420746 IP 123.123.123.123.30500 > bm-srv-01.voicenetwork.ca.23960: UDP, length 172 20:35:21.423869 IP bm-srv-01.voicenetwork.ca > 123.123.123.123: ICMP bm-srv-01.voicenetwork.ca udp port 23960 unreachable, length 208 20:35:21.441759 IP 123.123.123.123.30500 > bm-srv-01.voicenetwork.ca.23960: UDP, length 172 20:35:21.444943 IP bm-srv-01.voicenetwork.ca > 123.123.123.123: ICMP bm-srv-01.voicenetwork.ca udp port 23960 unreachable, length 208 20:35:21.459734 IP 123.123.123.123.30500 > bm-srv-01.voicenetwork.ca.23960: UDP, length 172 20:35:21.466007 IP bm-srv-01.voicenetwork.ca > 123.123.123.123: ICMP bm-srv-01.voicenetwork.ca udp port 23960 unreachable, length 208 20:35:21.480694 IP 123.123.123.123.30500 > bm-srv-01.voicenetwork.ca.23960: UDP, length 172 20:35:21.483860 IP bm-srv-01.voicenetwork.ca > 123.123.123.123: ICMP bm-srv-01.voicenetwork.ca udp port 23960 unreachable, length 208 20:35:21.501487 IP 123.123.123.123.30500 > bm-srv-01.voicenetwork.ca.23960: UDP, length 172 20:35:21.504884 IP bm-srv-01.voicenetwork.ca > 123.123.123.123: ICMP bm-srv-01.voicenetwork.ca udp port 23960 unreachable, length 208 20:35:21.519723 IP 123.123.123.123.30500 > bm-srv-01.voicenetwork.ca.23960: UDP, length 172 20:35:21.525664 IP bm-srv-01.voicenetwork.ca > 123.123.123.123: ICMP bm-srv-01.voicenetwork.ca udp port 23960 unreachable, length 208 What is strange, is that you see that at some point my own host (123.123.123.123) is being contacted on port 23960 which was formerly the originating port of the provider (although maybe this is expected, I don't know). I am not sure if this port is actually opened or not on the sipX side. Here is a fragment of the INVITE: Time: 2011-10-25T20:35:09.610000Z Frame: 9 sipxbridge.xml:2663 Source: 74.51.40.188:5060 Dest: voip.sipxecs.tld-sipXbridge ... v=0 o=FreeSWITCH 2592907442 2592907443 IN IP4 68.68.29.232 s=FreeSWITCH c=IN IP4 68.68.29.232 t=0 0 m=audio 23960 RTP/AVP 18 0 101 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=silenceSupp:off - - - - --- AJ On Tue, Oct 25, 2011 at 4:30 PM, Tony Graziano <tgrazi...@myitdepartment.net> wrote: > No. If the phone is connect to sipx, and sipx is doing the trunking, sipx is > anchoring the media. > Ensure the ITSP is ending the INVITE on the incoming call to port 5080. > > On Tue, Oct 25, 2011 at 4:07 PM, Adrien Guillon <aj.guil...@gmail.com> > wrote: >> >> 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/ > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: tgrazi...@voice.myitdepartment.net > Fax: 434.465.6833 > > Email: tgrazi...@myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: helpd...@voice.myitdepartment.net > > Helpdesk Contract Customers: > http://support.myitdepartment.net > Blog: > http://blog.myitdepartment.net > > Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > Ask about our Internet Fax services! > > _______________________________________________ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
