Also, FS should be showing the call as anchored and with the IP of the sipx server (the internal ip address).
So either you dont have sipx or the trunk configured properly, or you have a strange firewall issue too. there are many things probably wrong here, so here's a short list. 1. sipx server should be behind NAT. It's IP address should be using stun or have the public address manually input. 2. the itsp should NOT be doing nat traversal for you. 3. stop using the iptables sip conntrack modules, they will not be of any help. just setup iptables to do symmetric nat. 4. make sure your trunk say to use the public address for call setup. On Tue, Oct 25, 2011 at 6:00 PM, Tony Graziano <tgrazi...@myitdepartment.net > wrote: > Who is the provider? Are theya commercial or consumer provider? It sounds > like they do not support re-invite with sdp... > > Supply a proper sip trace. See the wiki: > > > http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer > > For a proper trace, you should have the logging level for the proxy, > voicemail and auto attendant and trunking to debug prior. rotate your logs, > and make a test call. don't forget to change the logging levels back. feel > free to post the trace to the list. tcpdump doesnt really cut it because it > does not show signalling between the various components, so you won't get a > lot of input on that. > > > On Tue, Oct 25, 2011 at 4:51 PM, Adrien Guillon <aj.guil...@gmail.com>wrote: > >> A bit more information... these bad ports seem to only happen after >> the call has been picked up.... so the AA will transfer, phone will >> ring (the phone that placed the call is disconnected), and when I pick >> up the dead call I get the unreachable packets... >> >> On Tue, Oct 25, 2011 at 4:47 PM, Adrien Guillon <aj.guil...@gmail.com> >> wrote: >> > 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/ >> > > > > -- > ====================== > 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 > > <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! > > -- ====================== 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 <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!
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