I did enable the dial plan, but I definitely need to go through his document and see if I did something wrong.
Melting Pot Technologies GMail wrote: > I would check your dial plans here like Tony said. It looks like the > server is telling itself the call is denied. Did you enable the dial > plan? You can see the different local components communicating in the > logs. > > On 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;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>;tag=8483786813757111981\r\nTo: >> <sip:16155008...@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>;tag=8483786813757111981\r\nTo: >> <sip:16155008...@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>;tag=425578349234274908\r\nTo: >>>>>>>>> >>>>>>>>> >>>>>>>>> <sip:1...@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>;tag=425578349234274908\r\nTo: >>>>>>>>> >>>>>>>>> >>>>>>>>> <sip:1...@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/
