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/
