But there is a simpler way to do the dial plan first. Ask Verizon what format your numbers should be. I would assume they would accept the full "+12025551212" (all +1 dialing). If so, the guide I have for bandwidth.comhas all the rules laid out. It should work the same, only the gateway name would be different.
On Fri, Nov 20, 2009 at 6:31 PM, Tony Graziano <tgrazi...@myitdepartment.net > wrote: > 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/ >> >
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