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] >>> <mailto: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]>;tag=784036913\r\nTo: \"Joegen >>> Baclor\" <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. _______________________________________________ sipx-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-dev/
