Hmmm. I tested this and it did not work, it still registers on port 5080. I then changed the sip trunking service and changed the public port from 5080 to port 5060 and I am not registering.
- javax.sip.InvalidArgumentException: Address already in use - at gov.nist.javax.sip.SipStackImpl.createListeningPoint(SipStackImpl.java:1193) - at org.sipfoundry.sipxbridge.Gateway.initializeSipListeningPoints(Gateway.java:545) - at org.sipfoundry.sipxbridge.Gateway.start(Gateway.java:1045) - at org.sipfoundry.sipxbridge.Gateway.main(Gateway.java:1412) - 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.OIOMessageProcessorFactory.createMessageProcessor(OIOMessageProcessorFactory.java:46) - at gov.nist.javax.sip.stack.SIPTransactionStack.createMessageProcessor(SIPTransactionStack.java:2155) - at gov.nist.javax.sip.SipStackImpl.createListeningPoint(SipStackImpl.java:1173) - ... 3 more - SipXbridge : Exception caught while running - org.sipfoundry.sipxbridge.SipXbridgeException: Cannot initialize gateway - at org.sipfoundry.sipxbridge.Gateway.initializeSipListeningPoints(Gateway.java:614) - at org.sipfoundry.sipxbridge.Gateway.start(Gateway.java:1045) - at org.sipfoundry.sipxbridge.Gateway.main(Gateway.java:1412) - Caused by: javax.sip.InvalidArgumentException: Address already in use - at gov.nist.javax.sip.SipStackImpl.createListeningPoint(SipStackImpl.java:1193) - at org.sipfoundry.sipxbridge.Gateway.initializeSipListeningPoints(Gateway.java:545) - ... 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.OIOMessageProcessorFactory.createMessageProcessor(OIOMessageProcessorFactory.java:46) - at gov.nist.javax.sip.stack.SIPTransactionStack.createMessageProcessor(SIPTransactionStack.java:2155) - at gov.nist.javax.sip.SipStackImpl.createListeningPoint(SipStackImpl.java:1173) - ... 3 more So this means there is more work to do to get these both to work or maybe you can get farther than I can... On Tue, Sep 21, 2010 at 9:25 PM, Tony Graziano <[email protected] > wrote: > Its the same. If you register to port 5080, that's where they will > send them. The default template is 5080 and the port number is not > exposed, I think if you change it to 5060 you can verify it registers > to port 5060 at the itsp, then verify the invite is sent to 5060. > > If that does not happen let me know and ill see if I can configure a > manual template and make it do this on 5060, but I don't think it will > be necessary. > > On 9/21/10, Joegen Baclor <[email protected]> wrote: > > I was under the impression that "ITSP Registrar Port" will be the port > > where the ITSP is expecting to receive REGISTERs and not where the > > bridge will be sending them from. Can you verify? > > > > On Wednesday, 22 September, 2010 09:06 AM, Tony Graziano wrote: > >> OK. The "registration status" page at the ITSP will show you what port > >> you are registered on. I think that would be a good enhancement for > >> sipxconfig (to show the registration port), but editing the gateway > >> itsp account to reflect "ITSP Registrar Port" and manually set it to > >> 5060 should show you after re-initialization at the voip.ms > >> <http://voip.ms> portal that the registration is at port 5060, at > >> which point your test can proceed. > >> > >> Logically, an ITSP that requires registration will send the calls to > >> the same port you are registered on. > >> > >> On Tue, Sep 21, 2010 at 6:25 PM, Joegen Baclor <[email protected] > >> <mailto:[email protected]>> wrote: > >> > >> Hi Everyone, > >> > >> I had a chance to test this patch using an ITSP. I used voip.ms > >> <http://voip.ms> to test. The call worked with bidirectional > >> audio and I was satisfied. The logs however shows that voip.ms > >> <http://voip.ms> did not send the call to port 5060 but via the > >> registration port 5080. Can you point me to an ITSP that insists > >> on sending to 5060? Or better yet, if you have an account with an > >> ITSP that behaves this way, would you be able to throw my test > >> server an inbound call via sip-trunk reg? Any help would be > >> appreciated. Thanks. > >> > >> Joegen > >> > >> On Monday, 20 September, 2010 05:49 PM, Tony Graziano wrote: > >>> Looking at the sipxbridge log, I see the by from 201, then the > >>> errors start. I think the question is where the bye should be > >>> sent and ack'd. Without seeing the sipxproxy.log its kind of hard > >>> to say. The error implies a listening error, but that is a bit of > >>> a long message and can simply be a result of "not knowing" what > >>> to do... the twinkle log looks plain, it shows sending the bye to > >>> bridge on port 5080. Is sipxbridge still listening locally on > >>> port 5080 in this environment?I just don;t know how to read it > >>> because the sipxbridge log shows the BYE on port 5060 and the > >>> twinkle log shows 5080. > >>> > >>> nBYE sip:[email protected] <sip%[email protected]> > >>> <mailto:sip%[email protected] > >>> <sip%[email protected]>>;x-sipX-nonat > SIP/2.0\r\nVia: > >>> SIP/2.0/UDP 112.201.137.176:5080 <http://112.201.137.176:5080>; > >>> > >>> Can you explain how the call flow for a bye should work (which > >>> service/port) and where it should be sent (directly to sipxbridge > >>> is my guess from the client and vice versa)? > >>> On Mon, Sep 20, 2010 at 2:19 AM, Joegen Baclor <[email protected] > >>> <mailto:[email protected]>> wrote: > >>> > >>> Hi Folks, > >>> > >>> For those of you who are following the development on this > >>> thread, I have attached a new set of twinkle log that > >>> demonstrates a complete call that passes through 5060 coming > >>> from a dummy ITSP. The previous log I have sent contained a > >>> glitch that is now corrected. I needed to modify contact > >>> creation in sipXbridge a bit so that it sends the internal IP > >>> address when talking to the proxy. This glitch is now > >>> corrected. > >>> > >>> However, I am now facing a new issue. When the BYE is coming > >>> from the called extension, sipXproxy sends a 407 for the BYE > >>> and sipXbridge suddenly barfs an exception > >>> > >>> > >>> "2010-09-20T05:48:59.328000Z":1188:sipxbridge:ERR:c2.ossapp.com: > Thread-88:00000000:SipListenerImpl:"Unexpected > >>> error processing response >>>> SIP/2.0 407 Proxy > >>> Authentication Required\r\nFrom: > >>> <sip:[email protected] > >>> <sip%[email protected]>>;tag=784036913\r\nTo: > \"Joegen > >>> Baclor\" <sip:[email protected] <sip%[email protected]> > >;tag=bjome\r\nCall-ID: > >>> kteensdeyxos...@bravia-c4\r\ncseq: 1 BYE\r\nVia: SIP/2.0/UDP > >>> > >>> 112.201.137.176:5080 > ;branch=z9hG4bK5fe25839220b0a96ff883192d4d6e60a373835;received=192.168.1.11;rport=5080\r\nProxy-Authenticate: > >>> Digest realm=\"c2.ossapp.com > >>> > >>> <http://c2.ossapp.com > >\",nonce=\"e669226f7847e446773d4cceeddd161a4c96f5cb\",qop=\"auth\"\r\nServer: > >>> sipXecs/4.3.0 sipXecs/sipXproxy (Linux)\r\nDate: Mon, 20 Sep > >>> 2010 05:48:59 GMT\r\nContent-Length: 0\r\n\r\n" > >>> javax.sip.SipException: Unexpected exception > >>> > >>> Newbie Questions: > >>> 1. Is the proxy suppose to authenticate mid-dialog requests > >>> from the bridge? Is this how it behaves currently? > >>> 2. What could be causing the bridge to barf? Isn't it > >>> suppose to just relay the response to the callee since it > >>> would know how to construct the authentication? Or is this > >>> something I have introduced by messing around with contact? > >>> > >>> Joegen > >>> > >>> > >>> > >>> On Thursday, 16 September, 2010 11:51 PM, Tony Graziano wrote: > >>>> I feel a little left out because they won't approve my > >>>> openscs registration request. > >>>> > >>>> On Thu, Sep 16, 2010 at 11:35 AM, Matt White > >>>> <[email protected] <mailto:[email protected]>> > >>>> wrote: > >>>> > >>>> Sweet!....thats what I was hoping to hear. > >>>> > >>>> I didn't think Avaya has released any code for it's new > >>>> openscs project as the website still has nothing new > >>>> from June. But was wondering if I missed something if > >>>> Avaya had actually released openscs code. > >>>> > >>>> I do think its funny the openscs webpage notes that > >>>> "/*As of July Avaya no longer participates in > >>>> SIPFoundry. SIPFoundry has forked the code base and is > >>>> being maintained by a new startup company.*/" > >>>> > >>>> Rather than Avaya being the one that forked it into a > >>>> new openscs project ;-) > >>>> > >>>> -M > >>>> > >>>> >>> Joegen Baclor 09/16/10 10:00 AM >>> > >>>> > >>>> Hi Matt, > >>>> > >>>> I've heard that Avaya is trying to solve this as well. > >>>> But this one is completely community/ezuce code. > >>>> > >>>> Joegen > >>>> > >>>> On Thursday, 16 September, 2010 09:04 PM, Matt White > wrote: > >>>>> Great news. This will go a long way towards increased > >>>>> interop. > >>>>> > >>>>> Out of curiosity, is any of this based on the work that > >>>>> avaya was doing before the fork? Or is this 100% > >>>>> community/ezuce code? > >>>>> > >>>>> -M > >>>>> > >>>>> >>> Joegen Baclor <[email protected]> > >>>>> <mailto:[email protected]> 09/16/10 4:37 AM >>> > >>>>> Hi Folks, > >>>>> > >>>>> I just thought you'd be interested in knowing that I > >>>>> have already > >>>>> successfully sent a call to port 5060 and forwarded to > >>>>> the bridge by > >>>>> redirection. See attached log from twinkle. > >>>>> Unfortutely, I am behind > >>>>> a firewall controlled by the ITSP so this is not yet > >>>>> tested in an actual > >>>>> environment. If you take a look at the log I have > >>>>> attached, the 200 OK > >>>>> is now coming from sipXbridge event if the call passed > >>>>> through the main > >>>>> sipXproxy listener. The ACK in this case will be > >>>>> misrouted because > >>>>> sipXbridge is sending the external IP as its contact. > >>>>> This however > >>>>> should not be an issue if the actual test call came > >>>>> from an entity > >>>>> outside my firewall. Hopefully we will have some more > >>>>> good news in the > >>>>> days to come. > >>>>> > >>>>> Joegen > >>>>> > >>>>> > >>>>> _______________________________________________ > >>>>> sipx-dev mailing list > >>>>> [email protected] > >>>>> <mailto:[email protected]> > >>>>> List Archive: > http://list.sipfoundry.org/archive/sipx-dev/ > >>>> > >>>> > >>>> _______________________________________________ > >>>> sipx-dev mailing list > >>>> [email protected] > >>>> <mailto:[email protected]> > >>>> List Archive: > http://list.sipfoundry.org/archive/sipx-dev/ > >>>> > >>>> > >>>> > >>>> > >>>> -- > >>>> ====================== > >>>> Tony Graziano, Manager > >>>> Telephone: 434.984.8430 > >>>> sip: [email protected] > >>>> <mailto:[email protected]> > >>>> Fax: 434.984.8431 > >>>> > >>>> Email: [email protected] > >>>> <mailto:[email protected]> > >>>> > >>>> LAN/Telephony/Security and Control Systems Helpdesk: > >>>> Telephone: 434.984.8426 > >>>> sip: [email protected] > >>>> <mailto:[email protected]> > >>>> Fax: 434.984.8427 > >>>> > >>>> Helpdesk Contract Customers: > >>>> http://www.myitdepartment.net/gethelp/ > >>>> > >>>> Why do mathematicians always confuse Halloween and Christmas? > >>>> Because 31 Oct = 25 Dec. > >>>> > >>>> > >>>> _______________________________________________ > >>>> sipx-dev mailing list > >>>> [email protected] > >>>> <mailto:[email protected]> > >>>> List Archive:http://list.sipfoundry.org/archive/sipx-dev/ > >>> > >>> > >>> > >>> > >>> -- > >>> ====================== > >>> Tony Graziano, Manager > >>> Telephone: 434.984.8430 > >>> sip: [email protected] > >>> <mailto:[email protected]> > >>> Fax: 434.984.8431 > >>> > >>> Email: [email protected] > >>> <mailto:[email protected]> > >>> > >>> LAN/Telephony/Security and Control Systems Helpdesk: > >>> Telephone: 434.984.8426 > >>> sip: [email protected] > >>> <mailto:[email protected]> > >>> Fax: 434.984.8427 > >>> > >>> Helpdesk Contract Customers: > >>> http://www.myitdepartment.net/gethelp/ > >>> > >>> Why do mathematicians always confuse Halloween and Christmas? > >>> Because 31 Oct = 25 Dec. > >>> > >> > >> > >> _______________________________________________ > >> sipx-dev mailing list > >> [email protected] <mailto:[email protected]> > >> List Archive: http://list.sipfoundry.org/archive/sipx-dev/ > >> > >> > >> > >> > >> -- > >> ====================== > >> Tony Graziano, Manager > >> Telephone: 434.984.8430 > >> sip: [email protected] > >> <mailto:[email protected]> > >> Fax: 434.984.8431 > >> > >> Email: [email protected] <mailto: > [email protected]> > >> > >> LAN/Telephony/Security and Control Systems Helpdesk: > >> Telephone: 434.984.8426 > >> sip: [email protected] > >> <mailto:[email protected]> > >> Fax: 434.984.8427 > >> > >> Helpdesk Contract Customers: > >> http://www.myitdepartment.net/gethelp/ > >> > >> Why do mathematicians always confuse Halloween and Christmas? > >> Because 31 Oct = 25 Dec. > >> > >> > >> _______________________________________________ > >> sipx-dev mailing list > >> [email protected] > >> List Archive: http://list.sipfoundry.org/archive/sipx-dev/ > > > > > > -- > Sent from my mobile device > > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: [email protected] > Fax: 434.984.8431 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: [email protected] > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > Why do mathematicians always confuse Halloween and Christmas? > Because 31 Oct = 25 Dec. > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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