btw - you need to state the firewall you are using. you also need to move this to the users list, this is not at all a dev issue.
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! > > -- ====================== 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!
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