Yeah, I thought so too (about the formats of the numbers) so I entered both numbers as they were written in the log files.
Oh well, it is hard for anyone to help me trouble shooting without all the information required. The error is probably one of those things that is a bit tricky to find but becomes obvious when you see it. I just wonder where I go from here, I suppose I could look more into the format of the numbers and try to find out what it actually wants, as well as look into if there are any settings that I've entered incorrectly. Btw, are there any other parts involved in this except for the proxy and the bridge (maybe I could find some more log messages)? 2011/9/16 Tony Graziano <[email protected]> > I was speaking about things like the auto attendant. > > detachment does no good. > > the carrier is sending you the d I d number and it could be in several > formats which has to match the alias you have entered for the user etcetera. > On Sep 16, 2011 7:59 AM, "Nils Adolfsson" <[email protected]> wrote: > > Thanks for the clarification of the dial plans. > > As I mentioned above, I've entered the DID number in the user alias, but > > where do I find the service alias (or did you mean that it's the same > > thing)? > > I'll attach the same log file again but with the DID numbers, just in > case. > > > > > > 2011/9/16 Tony Graziano <[email protected]> > > > >> 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/ > >> >> > >> > >> _______________________________________________ > >> sipx-users mailing list > >> [email protected] > >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ > >> > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >
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