I see two things that make me go hmmm.... "2009-11-20T23:11:36.845000Z": 19:OUTGOING:INFO:nshpbx1.sipx.voip:Thread-17:00000000:sipXbridge:"Sent SIP Message :\n----Remote Host:10.87.20.5---- Port: 38526----\nSIP/2.0 403 Forbidden\r\nVia: SIP/2.0/TCP 10.87.20.5;branch=z9hG4bK-sipXecs-0016a7a00a17b94001516f24875b44f09653;rport=38526\r\nVia: SIP/2.0/TCP 10.87.20.5;branch=z9hG4bK-sipXecs-0013a7669e28a72d44481cd5a375af980b06~f493f0f98bcc796b31453652fad2d124\r\nVia: SIP/2.0/UDP 10.87.20.254:5060;branch=z9hG4bK0ee159ab\r\nFrom: \"Kitchin Matthew\" <sip:1...@sipx.voip>;tag=00127f98eaa9004405071fb9-0d560bee\r\nTo: <sip:916155008...@sipx.voip>\r\nCall-ID: 00127f98-eaa9001d-4a738572-3b6e1...@10.87.20.254\r\ncseq: 102 INVITE\r\nServer: sipXecs/4.0.4 sipXecs/sipxbridge (Linux)\r\nContact: <sip:~~id~bri...@10.87.20.5:5090>\r\nSupported: replaces,100rel\r\nReason: ~~id~bridge;cause=213;text=\"Relayed Error Response\"\r\nContent-Length: 0\r\n\r\n--------------------END--------------------\n"
403 Forbidden - I'm assuming 10.87.20.5 is your sipx system? If so, are you *sure* Verizon will accept calls from this IP? <sip:~~id~bri...@10.87.20.5:5090>\r\nSupported: replaces,100rel\r\nReason: ~~id~bridge;cause=213;text=\" Relayed Error Response\"\r\nContent-Length: - This might indicate an issue with your UA. What is your phone you are sending this call from? The log is too short to be helpful because it's not providing this information. I see an open issue (XX-5823 <http://track.sipfoundry.org/browse/XX-5823>) related to handling the response, no action was taken because the phone used was not very firendly anyway. A complete trace of a failed call would be helpful. Directions are here: http://sipx-wiki.calivia.com/index.php/Display_SIP_message_flow_using_Sipviewer Since this is not in production, I would remove all the logs after putting the log modes in the correct level, place a test call and do the merge. Attached the merged.xml file to an email and send it in a reply to the list. You can also do the sipviewer and view it on a winpc at your end with no issue. On Fri, Nov 20, 2009 at 6:17 PM, mkitchin.pub...@gmail.com < mkitchin.pub...@gmail.com> wrote: > Not sure if there was a delay of some sort, but I got lots of activity > now. It is below. I will dig through them and see what I can find. They > announced the routing update (I'm not the network guy) and the equipment > did learn it. I can definitely ping their .1 address. > > > > "2009-11-20T23:11:36.831000Z":18:OUTGOING:INFO:nshpbx1.sipx.voip:Thread-17:00000000:sipXbridge:"Sent > SIP Message :\n----Remote Host:172.30.209.62---- Port: 5070----\nACK > sip:16155008...@pcelbcn0001.dsi.globalipcom.com<sip%3a16155008...@pcelbcn0001.dsi.globalipcom.com> > ;user=phone > SIP/2.0\r\nCall-ID: > 00127f98-eaa9001d-4a738572-3b6e1...@10.87.20.254.0\r\nmax-forwards: > 70\r\nFrom: \"Kitchin Matthew\" > <sip:1...@pcelbcn0001.dsi.globalipcom.com<sip%3a1...@pcelbcn0001.dsi.globalipcom.com> > >;tag=8483786813757111981\r\nTo: > <sip:16155008...@pcelbcn0001.dsi.globalipcom.com<sip%3a16155008...@pcelbcn0001.dsi.globalipcom.com> > ;user=phone>;tag=aprqngfrt-od3bjv2000020\r\nVia: > SIP/2.0/UDP > pcelbcn0001.dsi.globalipcom.com:5080 > ;branch=z9hG4bKf79a1b7109513b78e1683a1505f003513933\r\nCSeq: > 1 ACK\r\nRoute: <sip:172.30.209.62:5070;transport=udp;lr>\r\nUser-Agent: > sipXecs/4.0.4 sipXecs/sipxbridge (Linux)\r\nContent-Length: > 0\r\n\r\n--------------------END--------------------\n" > > "2009-11-20T23:11:36.845000Z":19:OUTGOING:INFO:nshpbx1.sipx.voip:Thread-17:00000000:sipXbridge:"Sent > SIP Message :\n----Remote Host:10.87.20.5---- Port: 38526----\nSIP/2.0 > 403 Forbidden\r\nVia: SIP/2.0/TCP > > 10.87.20.5;branch=z9hG4bK-sipXecs-0016a7a00a17b94001516f24875b44f09653;rport=38526\r\nVia: > SIP/2.0/TCP > > 10.87.20.5;branch=z9hG4bK-sipXecs-0013a7669e28a72d44481cd5a375af980b06~f493f0f98bcc796b31453652fad2d124\r\nVia: > SIP/2.0/UDP 10.87.20.254:5060;branch=z9hG4bK0ee159ab\r\nFrom: \"Kitchin > Matthew\" > <sip:1...@sipx.voip>;tag=00127f98eaa9004405071fb9-0d560bee\r\nTo: > <sip:916155008...@sipx.voip>\r\nCall-ID: > 00127f98-eaa9001d-4a738572-3b6e1...@10.87.20.254\r\ncseq: 102 > INVITE\r\nServer: sipXecs/4.0.4 sipXecs/sipxbridge (Linux)\r\nContact: > <sip:~~id~bri...@10.87.20.5:5090>\r\nSupported: > replaces,100rel\r\nReason: ~~id~bridge;cause=213;text=\"Relayed Error > Response\"\r\nContent-Length: > 0\r\n\r\n--------------------END--------------------\n" > > "2009-11-20T23:11:36.850000Z":20:INCOMING:INFO:nshpbx1.sipx.voip:Thread-16:00000000:sipXbridge:"Read > SIP Message :\n----Remote Host:172.30.209.62---- Port: 5070----\nSIP/2.0 > 100 Trying\r\nCall-ID: > 00127f98-eaa9001d-4a738572-3b6e1...@10.87.20.254.0\r\ncseq: 1 > INVITE\r\nFrom: \"Kitchin Matthew\" > <sip:1...@pcelbcn0001.dsi.globalipcom.com<sip%3a1...@pcelbcn0001.dsi.globalipcom.com> > >;tag=8483786813757111981\r\nTo: > <sip:16155008...@pcelbcn0001.dsi.globalipcom.com<sip%3a16155008...@pcelbcn0001.dsi.globalipcom.com> > ;user=phone>\r\nVia: > SIP/2.0/UDP > pcelbcn0001.dsi.globalipcom.com:5080 > ;received=10.87.20.5;branch=z9hG4bKf79a1b7109513b78e1683a1505f003513933\r\nContent-Length: > 0\r\n\r\n====================END====================\n" > > "2009-11-20T23:11:36.853000Z":21:INCOMING:INFO:nshpbx1.sipx.voip:PipelineThread-0:00000000:sipXbridge:"Read > SIP Message :\n----Remote Host:10.87.20.5---- Port: 38526----\nACK > sip:16155008...@pcelbcn0001.dsi.globalipcom.com:5070 SIP/2.0\r\nRoute: > <sip:10.87.20.5:5090;lr>\r\nContact: > <sip:1...@10.87.20.254:5060;transport=udp;x-sipX-nonat>\r\nFrom: > \"Kitchin Matthew\" > <sip:1...@sipx.voip>;tag=00127f98eaa9004405071fb9-0d560bee\r\nTo: > <sip:916155008...@sipx.voip>\r\nCall-ID: > 00127f98-eaa9001d-4a738572-3b6e1...@10.87.20.254\r\ncseq: 102 > ACK\r\nMax-Forwards: 20\r\nVia: SIP/2.0/TCP > > 10.87.20.5;branch=z9hG4bK-sipXecs-0016a7a00a17b94001516f24875b44f09653;rport=38526\r\nContent-Length: > 0\r\n\r\n====================END====================\n" > > > Melting Pot Technologies GMail wrote: > > How are you/they learning the route to your/their network? Can you > > ping their .1 address they gave you from the IPPBX? > > > > On Nov 20, 2009, at 6:03 PM, "mkitchin.pub...@gmail.com" > > <mkitchin.pub...@gmail.com> wrote: > > > >> It didn't put anything in a new log file. I've obviously got some > >> work to do on my end. > >> In a document I gave them several weeks ago, I did provide them, the > >> IP of my server. > >> > >> Melting Pot Technologies GMail wrote: > >>> Can you run: > >>> > >>> cd /var/log/sipxpbx > >>> > >>> rm -f ./sipxbridge.log > >>> > >>> Make a test call, and post the results from sipxbridge.log > >>> > >>> If your a static configuration they are more than likely pointing to > >>> a specific address on your end. Did they say anything about that? > >>> > >>> On Nov 20, 2009, at 5:25 PM, "mkitchin.pub...@gmail.com" > >>> <mkitchin.pub...@gmail.com> wrote: > >>> > >>>> Yes. I just found that under advanced settings. that seems to have > >>>> gotten rid of that error. Thank you! I still can't make any > >>>> outbound calls, but hopefully I will be able to find some more logs > >>>> showing why. The last entries in my sipxbridge log are below. My > >>>> inbound calls from Verizon are still set from them to come in on > >>>> 5060. I have it set to 5080 at the moment so it won;t conflict with > >>>> my phones attempting to talk to the server. I assume that should > >>>> only affect inbound calls, but assuming can make an ass out of me. > >>>> > >>>> > "2009-11-20T21:57:48.801000Z":1:JAVA:INFO:nshpbx1.sipx.voip:main:00000000:SipXbridgeXmlRpcServerImpl:"Supported > >>>> protocol = SSLv2Hello" > >>>> > "2009-11-20T21:57:48.814000Z":2:JAVA:INFO:nshpbx1.sipx.voip:main:00000000:SipXbridgeXmlRpcServerImpl:"Supported > >>>> protocol = SSLv3" > >>>> > "2009-11-20T21:57:48.814000Z":3:JAVA:INFO:nshpbx1.sipx.voip:main:00000000:SipXbridgeXmlRpcServerImpl:"Supported > >>>> protocol = TLSv1" > >>>> > "2009-11-20T21:57:49.840000Z":4:JAVA:INFO:nshpbx1.sipx.voip:main:00000000:Gateway:"------- > >>>> REGISTERING--------" > >>>> > >>>> > >>>> Melting Pot Technologies GMail wrote: > >>>>> Did you uncheck register on initialization? > >>>>> > >>>>> On Nov 20, 2009, at 4:24 PM, "mkitchin.pub...@gmail.com" > >>>>> <mkitchin.pub...@gmail.com> wrote: > >>>>> > >>>>>> In case you didn't have enough emails from me, here is a little more > >>>>>> info. I put in 123 for the username, so that is obviously where the > >>>>>> 'sip:1...@pcelbcn0001' entry is coming from. The service errors > >>>>>> out if I > >>>>>> don't put in a username and password, but Verizon isn't > >>>>>> requesting we > >>>>>> use one. > >>>>>> > >>>>>> mkitchin.pub...@gmail.com wrote: > >>>>>>> Here are some log file entries that appear relevant to me: > >>>>>>> > >>>>>>> > "2009-11-20T20:08:02.577000Z":5:OUTGOING:INFO:nshpbx1.sipx.voip:main:00000000:sipXbridge:"Sent > >>>>>>> > >>>>>>> SIP Message :\n----Remote Host:172.30.209.62---- Port: > >>>>>>> 5070----\nREGISTER sip:pcelbcn0001.munged.munged.com:5070 > >>>>>>> SIP/2.0\r\nCall-ID: > >>>>>>> 8d16619b30dd4006b74d20218ff1d...@10.87.20.5\r\ncseq: 1 > >>>>>>> REGISTER\r\nFrom: > >>>>>>> <sip:1...@pcelbcn0001.munged.munged.com<sip%3a...@pcelbcn0001.munged.munged.com> > >;tag=425578349234274908\r\nTo: > >>>>>>> > >>>>>>> <sip:1...@pcelbcn0001.munged.munged.com<sip%3a...@pcelbcn0001.munged.munged.com>>\r\nVia: > SIP/2.0/UDP > >>>>>>> pcelbcn0001.munged.munged.com:5060 > ;branch=z9hG4bK4bc3200f1f2720efc3aa25861a8eaa62333134\r\nMax-Forwards: > >>>>>>> > >>>>>>> 70\r\nUser-Agent: sipXecs/4.0.4 sipXecs/sipxbridge > >>>>>>> (Linux)\r\nAllow: > >>>>>>> INVITE,BYE,ACK,CANCEL,OPTIONS\r\nRoute: > >>>>>>> <sip:172.30.209.62:5070;transport=udp;lr>\r\nContact: > >>>>>>> <sip:1...@pcelbcn0001.munged.munged.com:5060 > ;transport=udp>\r\nExpires: > >>>>>>> > >>>>>>> 600\r\nContent-Length: > >>>>>>> 0\r\n\r\n--------------------END--------------------\n" > >>>>>>> > "2009-11-20T20:08:02.613000Z":6:INCOMING:INFO:nshpbx1.sipx.voip:Thread-13:00000000:sipXbridge:"Read > >>>>>>> > >>>>>>> SIP Message :\n----Remote Host:172.30.209.62---- Port: > >>>>>>> 5070----\nSIP/2.0 403 Forbidden\r\nVia: SIP/2.0/UDP > >>>>>>> pcelbcn0001.munged.munged.com:5060 > ;received=10.87.20.5;branch=z9hG4bK4bc3200f1f2720efc3aa25861a8eaa62333134\r\nCall-ID: > >>>>>>> > >>>>>>> 8d16619b30dd4006b74d20218ff1d...@10.87.20.5\r\ncseq: 1 > >>>>>>> REGISTER\r\nFrom: > >>>>>>> <sip:1...@pcelbcn0001.munged.munged.com<sip%3a...@pcelbcn0001.munged.munged.com> > >;tag=425578349234274908\r\nTo: > >>>>>>> > >>>>>>> <sip:1...@pcelbcn0001.munged.munged.com<sip%3a...@pcelbcn0001.munged.munged.com> > >;tag=aprqngfrt-gjiai91000020\r\nContent-Length: > >>>>>>> > >>>>>>> 0\r\n\r\n====================END====================\n" > >>>>>>> <?xml version="1.0" encoding="UTF-8" standalone="no"?> > >>>>>>> <!DOCTYPE log SYSTEM "logger.dtd"> > >>>>>>> <log> > >>>>>>> </log> > >>>>>>> > "2009-11-20T20:08:12.601000Z":1:JAVA:INFO:nshpbx1.sipx.voip:main:00000000:SipXbridgeXmlRpcServerImpl:"Supported > >>>>>>> > >>>>>>> protocol = SSLv2Hello" > >>>>>>> > "2009-11-20T20:08:12.602000Z":2:JAVA:INFO:nshpbx1.sipx.voip:main:00000000:SipXbridgeXmlRpcServerImpl:"Supported > >>>>>>> > >>>>>>> protocol = SSLv3" > >>>>>>> > "2009-11-20T20:08:12.602000Z":3:JAVA:INFO:nshpbx1.sipx.voip:main:00000000:SipXbridgeXmlRpcServerImpl:"Supported > >>>>>>> > >>>>>>> protocol = TLSv1" > >>>>>>> > "2009-11-20T20:08:12.683000Z":4:JAVA:ERR:nshpbx1.sipx.voip:main:00000000:sipXbridge:"gov.nist.javax.sip.SipStackImpl.createListeningPoint(SipStackImpl.java:1080) > >>>>>>> > >>>>>>> [Invalid argument address = 10.87.20.5 port = 5060 transport = > >>>>>>> udp]" > >>>>>>> > "2009-11-20T20:08:12.686000Z":5:JAVA:ERR:nshpbx1.sipx.voip:main:00000000:Gateway:"Cannot > >>>>>>> > >>>>>>> initialize gateway" > >>>>>>> javax.sip.InvalidArgumentException: Address already in use > >>>>>>> > >>>>>>> > >>>>>>> mkitchin.pub...@gmail.com wrote: > >>>>>>>> This was an inevitable question from me. I need some help > >>>>>>>> connecting > >>>>>>>> to Verizon SIP over a private DS3. There is no firewall or NAT > >>>>>>>> involved. The information they gave me is below. > >>>>>>>> > >>>>>>>> From Verizon: > >>>>>>>> Inbound calls will route from the 172.30.9.0/24 port 5060 > >>>>>>>> network and > >>>>>>>> you should be able to ping 172.30.9.1. This is the only > >>>>>>>> address you > >>>>>>>> will be able to ping for security reasons. > >>>>>>>> For outbound calls please configure the SIP target (to the VzB > >>>>>>>> network) to one of the settings below. > >>>>>>>> IP: 172.30.209.62 port: 5070 > >>>>>>>> OR > >>>>>>>> FQDN: pcelbcn0001.munged.munged.com > >>>>>>>> > >>>>>>>> I'm using the sipexec server as the SBC. It is at 10.87.20.5. I > >>>>>>>> have > >>>>>>>> tried to translate this into all the correct fields on the > >>>>>>>> configuration guide here: > >>>>>>>> > http://sipx-wiki.calivia.com/index.php/SIP_Trunking_with_sipXecs:_Overview_and_Configuration > >>>>>>>> > >>>>>>>> > >>>>>>>> No matter what I try, The Sip Trunking service fails to start with > >>>>>>>> the 'Address already in use' error below. I googled several of the > >>>>>>>> lines, and I found some bug reports and other writeups that didn't > >>>>>>>> appear to relate to my problem. I cleared one other error by > >>>>>>>> putting > >>>>>>>> in a fake username and password under ITSP account. I don't > >>>>>>>> have an > >>>>>>>> username and password. I would assume that is because this is a > >>>>>>>> private connection. As you can see, I have received minimal > >>>>>>>> information from Verizon. I also have no NAT or firewall > >>>>>>>> involved, so > >>>>>>>> several of the configuration screens regarding NAT don't really > >>>>>>>> pertain to me, but I had to put in a value of some sort. On > >>>>>>>> System, > >>>>>>>> Servers, NAT, Public IP address for example, I had to put > >>>>>>>> something, > >>>>>>>> so I put 10.87.20.5. Verizon has performed a miracle in their > >>>>>>>> minds > >>>>>>>> by simply agreeing to work with a 'vendorless open source PBX', > >>>>>>>> and > >>>>>>>> we are supposed to have their Interop test with wireshark > >>>>>>>> captures on > >>>>>>>> Monday. I need to do anything possible to get this working by > >>>>>>>> then. > >>>>>>>> With the information I have, can someone help me figure out > >>>>>>>> exactly > >>>>>>>> what values should be put where in the various config screens? > >>>>>>>> A few > >>>>>>>> are obvious, but a few aren't for me at least given give that > >>>>>>>> there > >>>>>>>> is no firewall, NAT or ITSP account. > >>>>>>>> > >>>>>>>> Thanks a ton, > >>>>>>>> Matthew > >>>>>>>> > >>>>>>>> javax.sip.InvalidArgumentException: Address already in use > >>>>>>>> at > >>>>>>>> > gov.nist.javax.sip.SipStackImpl.createListeningPoint(SipStackImpl.java:1083) > >>>>>>>> > >>>>>>>> > >>>>>>>> at > >>>>>>>> > org.sipfoundry.sipxbridge.Gateway.initializeSipListeningPoints(Gateway.java:540) > >>>>>>>> > >>>>>>>> > >>>>>>>> at org.sipfoundry.sipxbridge.Gateway.start(Gateway.java:1000) > >>>>>>>> at org.sipfoundry.sipxbridge.Gateway.main(Gateway.java:1353) > >>>>>>>> Caused by: java.io.IOException: Address already in use > >>>>>>>> at > >>>>>>>> > gov.nist.javax.sip.stack.UDPMessageProcessor.<init>(UDPMessageProcessor.java:130) > >>>>>>>> > >>>>>>>> > >>>>>>>> at > >>>>>>>> > gov.nist.javax.sip.stack.SIPTransactionStack.createMessageProcessor(SIPTransactionStack.java:1890) > >>>>>>>> > >>>>>>>> > >>>>>>>> at > >>>>>>>> > gov.nist.javax.sip.SipStackImpl.createListeningPoint(SipStackImpl.java:1064) > >>>>>>>> > >>>>>>>> > >>>>>>>> ... 3 more > >>>>>>>> SipXbridge : Exception caught while running > >>>>>>>> org.sipfoundry.sipxbridge.SipXbridgeException: Cannot > >>>>>>>> initialize gateway > >>>>>>>> at > >>>>>>>> > org.sipfoundry.sipxbridge.Gateway.initializeSipListeningPoints(Gateway.java:598) > >>>>>>>> > >>>>>>>> > >>>>>>>> at org.sipfoundry.sipxbridge.Gateway.start(Gateway.java:1000) > >>>>>>>> at org.sipfoundry.sipxbridge.Gateway.main(Gateway.java:1353) > >>>>>>>> Caused by: javax.sip.InvalidArgumentException: Address already > >>>>>>>> in use > >>>>>>>> at > >>>>>>>> > gov.nist.javax.sip.SipStackImpl.createListeningPoint(SipStackImpl.java:1083) > >>>>>>>> > >>>>>>>> > >>>>>>>> at > >>>>>>>> > org.sipfoundry.sipxbridge.Gateway.initializeSipListeningPoints(Gateway.java:540) > >>>>>>>> > >>>>>>>> > >>>>>>>> ... 2 more > >>>>>>>> Caused by: java.io.IOException: Address already in use > >>>>>>>> at > >>>>>>>> > gov.nist.javax.sip.stack.UDPMessageProcessor.<init>(UDPMessageProcessor.java:130) > >>>>>>>> > >>>>>>>> > >>>>>>>> at > >>>>>>>> > gov.nist.javax.sip.stack.SIPTransactionStack.createMessageProcessor(SIPTransactionStack.java:1890) > >>>>>>>> > >>>>>>>> > >>>>>>>> at > >>>>>>>> > gov.nist.javax.sip.SipStackImpl.createListeningPoint(SipStackImpl.java:1064) > >>>>>>>> > >>>>>>>> > >>>>>>>> ... 3 more > >>>>>>> > >>>>>> > >>>>>> _______________________________________________ > >>>>>> sipx-users mailing list sipx-users@list.sipfoundry.org > >>>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users > >>>>>> Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > >>>>>> sipXecs IP PBX -- http://www.sipfoundry.org/ > >>>> > >> > > _______________________________________________ > sipx-users mailing list sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > sipXecs IP PBX -- http://www.sipfoundry.org/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/
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