there is this thing called sip trace. on the wiki. generate 1 and post it. On Sep 16, 2011 8:32 AM, "Nils Adolfsson" <[email protected]> wrote: > 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/ >>
_______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
