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/ >>> >
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