Here you go. 2011/9/16 Tony Graziano <[email protected]>
> 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/ >
sipx-net-trace-110916T175117.pcap
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_______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
