I read your guide, and I don't see anything that sticks out at me that I'm doing wrong. Verizon says they are sending 10 and expecting 10 digits. I have enabled and tweaked the default dialing plans. I'm using Cisco 7940s, because I had a few laying around. I am home now, so I just tried Xlite and get the same results. I'm getting some demo Polycom Soundpoint 450s and 550s on Monday so I will see if they have any different results. I will go through it again from top to bottom and make sure am not missing anything obvious. If I still get the same results, I will send a complete trace. Thanks again for your help.
Tony Graziano wrote: > 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.com <http://bandwidth.com> has 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 <mailto: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 > <mailto: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 > <http://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 > <http://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 > <mailto:mkitchin.pub...@gmail.com> <mkitchin.pub...@gmail.com > <mailto: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 > <mailto: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 > > <mailto:sip%3a1...@pcelbcn0001.dsi.globalipcom.com>>;tag=8483786813757111981\r\nTo: > <sip:16155008...@pcelbcn0001.dsi.globalipcom.com > > <mailto: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 > <mailto: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 > <http://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 > > <mailto:sip%3a1...@pcelbcn0001.dsi.globalipcom.com>>;tag=8483786813757111981\r\nTo: > <sip:16155008...@pcelbcn0001.dsi.globalipcom.com > > <mailto: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 > <http://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 > <mailto: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 > <mailto:mkitchin.pub...@gmail.com>" > > <mkitchin.pub...@gmail.com > <mailto: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 > <mailto:mkitchin.pub...@gmail.com>" > >>> <mkitchin.pub...@gmail.com > <mailto: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 > <mailto:mkitchin.pub...@gmail.com>" > >>>>> <mkitchin.pub...@gmail.com > <mailto: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 > <mailto: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 > <http://pcelbcn0001.munged.munged.com:5070> > >>>>>>> SIP/2.0\r\nCall-ID: > >>>>>>> 8d16619b30dd4006b74d20218ff1d...@10.87.20.5 > <mailto:8d16619b30dd4006b74d20218ff1d...@10.87.20.5>\r\nCSeq: 1 > >>>>>>> REGISTER\r\nFrom: > >>>>>>> <sip:1...@pcelbcn0001.munged.munged.com > > <mailto:sip%3a...@pcelbcn0001.munged.munged.com>>;tag=425578349234274908\r\nTo: > >>>>>>> > >>>>>>> <sip:1...@pcelbcn0001.munged.munged.com > <mailto: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 > <mailto:8d16619b30dd4006b74d20218ff1d...@10.87.20.5>\r\nCSeq: 1 > >>>>>>> REGISTER\r\nFrom: > >>>>>>> <sip:1...@pcelbcn0001.munged.munged.com > > <mailto:sip%3a...@pcelbcn0001.munged.munged.com>>;tag=425578349234274908\r\nTo: > >>>>>>> > >>>>>>> <sip:1...@pcelbcn0001.munged.munged.com > > <mailto: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 > <mailto: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 > <http://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 > <http://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 > <mailto: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 > <mailto: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/
