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
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