Thanks for your replies I'm sorry that I wrote the mail to the dev mail list, I must have copied the wrong address. Once I move the topic to the user list I will supply you with more information about the case
Thanks // Nils 2011/9/9 Michael Picher <[email protected]> > Let me try to clarify things here again... > > From what I have seen, there are generally 2 types of authentication used > for SIP trunks (don't complain if some oddball somewhere decided to do > something different)... What I call 'IP based' and l'ogin/registration > based'. > > For IP based trunks you need to setup 5080 udp inbound AND 30000 - 31000 > UDP inbound and have those NAT'd to the server. And you need to tell your > ITSP to send to you on 5080. You can send to them from whatever port you > want, to whatever port you want. > > For login/registration based SIP trunks you DO NOT need to map these ports > inbound!!!! Nor do you need to worry about 5080 vs. 5060!!!! sipXecs will > open a connection from inside your firewall to your ITSP and then the > keepalive keeps that connection open. This is automatic and a function of > firewalls. The ITSP will send to your across that established connection. > > In both cases the firewall must not do outbound port randomization for > NAT'd connections from sipXecs. > > Off soapbox. > > Mike > > On Fri, Sep 9, 2011 at 5:24 AM, Tony Graziano < > [email protected]> wrote: > >> i think it has everything to do with how you set up the registration. the >> asserted identity doesnt come into play until after you register. >> >> you need to provide more detail. the provider requires registration. who >> is the provider? >> >> when you register you register from port 5080 to port 5060 with the >> carrier. the carrier should see the FROM port and send you invites on that >> port (5080). when you send calls to the carrier, you send them to 5060. >> >> the carriers says register to them by domain, but it seems like maybe you >> are inputting an ip in the registration area. who is the itsp? >> >> On Fri, Sep 9, 2011 at 4:46 AM, Nils Adolfsson >> <[email protected]>wrote: >> >>> Hi, >>> >>> I am currently trying to set up a SIP trunk so that I can call to regular >>> phones through my SipX server. >>> I am having some problems though to authenticate with the ITSP's SIP >>> trunk service. >>> Log messages from sipxbridge.log shows that the request either times out >>> or that it is not found (errors 404 and 408). >>> I do not believe that it is the fire wall, because both port 5060 and >>> port 5080 should be open both to and from the SipX server. >>> >>> The SipX server knows that it is under NAT and that it should use NAT >>> traversal, so I do find it a bit interesting that it writes the local >>> address in the SIP messages. >>> I also find it interesting that the to and the from addresses are >>> identical saying "username@ITSP_provider_address", especially when the >>> ITSP (I called them to see if they had any logs of what was wring) said that >>> it should be "username@my_domain". >>> I've tried to change this by changing the "asserted identity" as well as >>> the "preferred identity" options in the ITSP account settings in the >>> gateway. >>> These settings are ignored so that the messages still >>> contain "username@ITSP_provider_address" instead of what is written in >>> those fields. >>> >>> By the way, is port 5060 the correct port to use? After looking at a >>> couple of guides I got the feeling that it should go through port 5080. >>> (I have tried to change it, but it didn't help. I also used the option >>> where SipX looks at port 5080 for SIP trunking messages as well, and I >>> didn't have much luck there either). >>> >>> The two main guides I've followed are: >>> http://wiki.sipfoundry.org/display/sipXecs/SIP+Trunking >>> http://blog.myitdepartment.net/?p=191 >>> >>> >>> Log messages from /var/log/sipxpbx/sipxbridge.log >>> >>> Outgoing message: >>> ---------------------------- >>> "2011-09-09T10:04:07.209000Z":20:OUTGOING:INFO:sipx1.prod.sipx:Timer-0:00000000:sipXbridge:"Sent >>> SIP Message :\n----Remote Host:192.168.10.12---- Port: 5060----\nREGISTER >>> sip:ITSP_provider_domain SIP/2.0\r\nCall-ID: >>> [email protected]\r\nCSeq: 2 >>> REGISTER\r\nFrom: >>> <sip:username@ITSP_provider_domain>;tag=892685948627891857\r\nTo: >>> <sip:username@ITSP_provider_domain>\r\nVia: SIP/2.0/TCP >>> 192.168.10.12:5080;branch=z9hG4bK65a6742857b86280cbfa7e40924e361e383035\r\nMax-Forwards: >>> 70\r\nUser-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)\r\nAllow: >>> INVITE,BYE,ACK,CANCEL,OPTIONS\r\nSupported: timer\r\nRoute: >>> <sip:192.168.10.12:5060;transport=tcp;lr>\r\nContact: < >>> sip:[email protected];transport=tcp>\r\nExpires: >>> 600\r\nContent-Length: >>> 0\r\n\r\n--------------------END--------------------\n" >>> >>> Incoming message: >>> ---------------------------- >>> "2011-09-09T10:04:12.336000Z":22:INCOMING:INFO:sipx1.prod.sipx:PipelineThread-0:00000000:sipXbridge:"Read >>> SIP Message:\n----Remote Host:192.168.10.12---- Port: 5060----\nSIP/2.0 408 >>> Request timeout\r\nFrom: >>> <sip:username@ITSP_provider_domain>;tag=892685948627891857\r\nTo: >>> <sip:username@ITSP_provider_domain>;tag=CHszxZ\r\nCall-ID: >>> [email protected]\r\nCSeq: 2 >>> REGISTER\r\nVia: SIP/2.0/TCP >>> 192.168.10.12:5080;branch=z9hG4bK65a6742857b86280cbfa7e40924e361e383035\r\nServer: >>> sipXecs/4.4.0 sipXecs/sipXproxy (Linux)\r\nContent-Length: >>> 0\r\n\r\n====================END====================\n" >>> >>> Sniffing with Wireshark shows pretty much the same thing as these logs. >>> >>> _______________________________________________ >>> sipx-dev mailing list >>> [email protected] >>> List Archive: http://list.sipfoundry.org/archive/sipx-dev/ >>> >> >> >> >> -- >> ====================== >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> sip: [email protected] >> Fax: 434.465.6833 >> >> Email: [email protected] >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> sip: [email protected] >> >> Helpdesk Contract Customers: >> http://support.myitdepartment.net >> >> <http://support.myitdepartment.net>Blog: >> http://blog.myitdepartment.net >> >> Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >> >> Ask about our Internet faxservices! >> >> >> _______________________________________________ >> sipx-dev mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-dev/ >> > > > > -- > Michael Picher > eZuce > Director of Technical Services > O.978-296-1005 X2015 > M.207-956-0262 > @mpicher <http://twitter.com/mpicher> > www.ezuce.com > > > _______________________________________________ > sipx-dev mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-dev/ >
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