crap... you need to confirm the nat and port forwarding in your home router is going to do what it needs to do. unless you can out your broadband into a bridged mode and control your own firewall properly configured, I would not waste your time.
On Sun, Oct 2, 2011 at 7:04 PM, Ewan McLean <[email protected]> wrote: > It's a SOHO installation, when I say home router I just mean it's not a > Cisco enterprise router or anything, it's a broadband router. Sipx is in the > home office. Each computer is setup with security and IPTABLES will be added > to sipx once it's working. > > Tony Graziano wrote: > > hold on a sec... what is the relevance of the home router type mentioning? > > is sipx at home or somewhere else? if so, what type of firewall is being > used? > > On Sun, Oct 2, 2011 at 6:47 PM, Ewan McLean > <[email protected]> wrote: > > It should be just one call - I rotated out the logs before I placed a test > call then merged them immediately afterwards? I dialled in on the ITSP > trunk, got connected to my auto attendant, pressed one to transfer to an > internal extension (201), heard the 'hold while i transfer you' blurb, then > got cut off. Meanwhile extension 201 started ringing even though the call > had been cut off, it was silence when the extension was picked up. > > Intranet domains *.servistech.co.uk > Intranet subnets 192.168.1.0/25 > > Sipx at 192.168.1.100 > Extension 201 on a polycom at 192.168.1.5 > > Home type ADSL router/switch at 192.168.1.100 with no port forwarding, > firewall etc. set up as far as I can tell. > > The ITSP is called Numbergroup at sip.numbergroup-services.com > > I tried an unmanaged gateway but it didn't seem to work. I plugged in the > same address as the destination but there was nowhere to plug in > authentication details and as far as I know it's setup to require those. > > I'm new to the raw SIP messages here, I used to use asterisk which hid all > of it. Happy to learn, but it's a bit overwhelming. I'm not sure what the > difference between the media server, bridge, proxy, registrar, trunking etc. > components are. > > Does any of this help? > > Tony Graziano wrote: > > I don't think anyone should make sense of this until you can explain > the call flow, or limit the call trace to a particular call. > > Can you do one or the other? > > The ITSP is syaing the options are not acceptable, stating: Reason: > Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" > > Which probably means they dont like the codec in the media server. > That really shouldn't matter since the destination should NOT be the > media server. The REFER is not being handled locally from what I can > see, otherwise they would be accepting the call. > > Explain what you have in intranets/subnets and what type of network > sipxecs is on locally. Then provide how you connect to the ITSP > (registration or IP based). > > In any case, I think I see the reinvite in frame 36, and the ITSP says > BYE at frame 47 saying an incompatible destination. > > So my question here is, since it appears that the ITSP (see frame 10) > supports REFER, why don't you set them up as an unmanaged gateway and > esentially leave sipxbridge out of the mix? > > > On Sun, Oct 2, 2011 at 2:30 PM, Tony Graziano > <[email protected]> wrote: > > I cantlook at thetrace until later. the itsp must support reinvite in order > to handle transfers. > > the are directions on the wiki to produce a trace for one particular call. > > On Oct 2, 2011 2:19 PM, "Ewan McLean" <[email protected]> wrote: > > Hi > > I have an autoattendant set up. When someone presses 1 to be > transferred, the call rings on the destination phone (and keeps ringing) > but the caller gets disconnected. I have isolated this to a particular > gateway as the same attendant works perfectly on another gateway > provided by a different ITSP. This is the trace file. I'm having > difficulty making sense of it as it's a much bigger trace than the ones > I've been getting used to recently. > > On an aside this is the same ITSP I have been having trouble getting > outbound calling working with, I haven't had time to look at that > properly yet. > > Can anyone help? > > Many thanks > > Ewan > > -- > <http://servistech.co.uk>Ewan McLean > > 02034684428 > servistech.co.uk <http://servistech.co.uk> > > > > _______________________________________________ > 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/ > -- ====================== 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 Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
