I am having a hard time following you because the dial plan is not used for incoming calls.
the user alias or the service alias for d I d number is were you put the incoming number in the format it's provided. no 1 can provide you with that because you manually edit did that out of your log file. everything in this thread so far is in the wiki. On Sep 16, 2011 5:34 AM, "Nils Adolfsson" <[email protected]> wrote: > Sorry for reviving a dead thread. > For those who find this conversation on any search engine: > > The problem I had was solved by changing the To and From parameters from > <sip:username@ITSP_provider_domain> to <sip:username@my_domain>. > > The problem I'm having now is to call in from a regular phone on the public > phone net to through the SIP trunk and to the SipX server. > Before saying any more, the server is located OUTSIDE the company's firewall > using a public address (I will deal with possible firewall problems later > when the SipX configuration works). > > As it looks right now, the SipX server has a SIP trunk gateway that > successfully registers with the ITSP's SIP trunk. > However, when calling to the SipX server from a regular phone it fails. > Every phone extension has its own public phone number, i.e. I'm using DID's > if I understand everything correctly. > I have tried to set up dial plans to route incoming calls from the DID > number to the local extension, enter the DID as the alias of the user and > make a user who's extension is its DID. > And none of this works. > > It seems as if the calls come into the SipX bridge and to the proxy, however > then it says "404 Not found" and sends an error message back to the SIP > trunk. > > I have followed several guides showing how to set things up but it doesn't > seem to work when I do as they do. > > The short version of the logs of both the proxy and the bridge is the > following: > > IN: Invite > OUT: Trying > OUT: Invite > IN: Trying > IN: 404 Not found > OUT: ACK > OUT: 404 Not found > IN: ACK > > The whole log entries can be found in the attached .txt file. > > Regards > Nils Adolfsson > > 2011/9/9 Tony Graziano <[email protected]> > >> Oh, juniper needs to have the SIP ALG function turned off, and 1:1 nat >> enabled. Im guessing you don't have the SIP off in it. it's intentionally >> mangling the ports. >> >> >> On Fri, Sep 9, 2011 at 8:03 AM, Tony Graziano < >> [email protected]> wrote: >> >>> I don't think you get it. >>> >>> provide the ITSp name. Maybe someone has done this before. Some are >>> quirky. >>> >>> State the firewall. If you dont have symmetric nat ebaled it WILL NOT >>> WORK. >>> >>> Make sure your intranet subnets is stated properly. >>> >>> it times out/cant be found because it cant resolve the name or your IP >>> address you are entering for the itsp can't be reached. >>> >>> you are being asked question, repeatedly, but you are avoiding answering >>> them. >>> >>> On Fri, Sep 9, 2011 at 7:48 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 I have tried to open >>>> both port 5060 and port 5080 both to and from the SipX server. >>>> Another reason why it should not be problem with the firewall is because >>>> that my SipX server is the one who registers to the ITSP server, i.e. the >>>> SipX server opens a connection. >>>> >>>> I find it a bit interesting that it writes the local address in the SIP >>>> messages (as you can see in the log message below). >>>> The SipX server knows that it is under NAT and that it should use NAT >>>> traversal, as well as that it knows its public IP address. >>>> I also find it interesting that the source and destination addresses are >>>> identical saying "username@ITSP_provider_domain", 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 have tried to use port 5080 on my server and send authentication >>>> requests to port 5060 on the ITSP's server. >>>> I have also tried to use port 5060 on my SipX server while using the >>>> option where SipX listens to that port for SIP trunking messages. >>>> >>>> The two main guides I've followed are: >>>> http://wiki.sipfoundry.org/display/sipXecs/SIP+Trunking >>>> http://blog.myitdepartment.net/?p=191 >>>> >>>> Other info >>>> -------------- >>>> The firewall is a Juniper SSG5 (firmware 6.3.0R8.0) >>>> ITSP: DGC (http://www.dgc.se/sv/om-dgc/About-DGC-ENG/) >>>> Name of the SipX server: sipx1.prod.sipx >>>> >>>> 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-users mailing list >>>> [email protected] >>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> >>> >>> >>> >>> -- >>> ====================== >>> 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! >> >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>
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