Thanks for all the replies. I've tried entering all kinds of number combinations as aliases now, so if it "should" be as easy as just entering the DID number there it should have worked by now. After reading what Michael Picher wrote I've also tried to use dial plans. After all, it does make sense to be able to use a rule that can be applied to a group of people to lower the amount of repetitive labour.
I've also spoken to a co-worker of mine to see if we can set up an old asterisk based system to connect to the SIP trunk. That way we can see if the messages sent between the trunk and the two different systems differ in any way. I'm beginning to suspect my ITSP a bit because I need to set the "INVITE >From ITSP Account" flag in the gateway (which apparently does not follow the RFC - btw, is that RFC 3261?). If I don't, I get a "604 does not exist anywhere" message back from the ITSP when I try to call out from a soft phone, that is connected to my SipX server, to a regular phone. I'm not saying that it works with it checked though. Because what happens is that the SipX server goes to the SIP trunk and successfully starts the call. The soft phone then starts sending a lot of RTP packets to the SipX server. However, my phone does not ring and no RTP packets are sent from the SipX server to the SIP trunk, nor does the soft phone receive any RTP packets. I will come back later when I've tested this more thoroughly and know more about the "604 does not exist anywhere" error message. And yes, mr Graziano, I know that all of this can be found in the wiki, which you can see in my original post. Basically, the reason why I started this thread was because the tips found here http://wiki.sipfoundry.org/display/sipXecs/Trouble+shooting+and+problem+reporting did not help when I was trouble shooting after following the guide in the wiki. Therefore I turned to the people in this mailing list who have considerable more experience of sipx than me. In that way someone might have been able point out where I could go next in my hunt for solutions of my troubles. I've got some pointers now, and I'm very thankful for that Regards Nils 2011/9/17 Todd Hodgen <[email protected]> > Are you including incoming calls when you say “all”**** > > ** ** > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Michael Picher > *Sent:* Friday, September 16, 2011 4:31 PM > > *To:* Discussion list for users of sipXecs software > *Subject:* Re: [sipx-users] Trouble with setting up SIP trunking**** > > ** ** > > Fyi, dial plans process all calls after user extensions/aliases... not > just outbound calls.**** > > On Sep 16, 2011 12:42 PM, "Todd Hodgen" <[email protected]> wrote: > > Tony brings up a good point here, that there really isn't a great > document > > describing in detail when you look at the overall architecture of > sipxecs. > > The dial plans in sipXecs are only used for Outgoing calls, and have no > > effect on incoming calls. And, just to re-iterate what I've seen answered > > here many times on this list, when the dial plans are used on a call, it > > looks for the first match, and then takes that route - so if you are > having > > issues with outgoing calls, start at the top of our plans and start > looking > > for matches until you find one - and your problem. > > > > > > > > For your incoming calls, ensure the number is in an alias somewhere, and > > ensure it is the right length in characters. Putting the entire number in > > that field when the carrier provides the last 4 digits will be of no > value. > > If your alias doesn't work with an extension in the system, try it on an > > auto attendant. > > > > > > > > If it works on one, and not the other, you need to start looking for > issues > > between the system and the carrier. > > > > > > > > From: [email protected] > > [mailto:[email protected]] On Behalf Of Tony > Graziano > > Sent: Friday, September 16, 2011 3:39 AM > > To: Discussion list for users of sipXecs software > > Subject: Re: [sipx-users] Trouble with setting up SIP trunking > > > > > > > > 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-0:00000 > > 000: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\nM > > ax-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] <mailto: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:PipelineThrea > > d-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\nS > > erver: > >>>>> 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/ >
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