I've seen very similar results when the ITSP cannot accept SDP without options as well. Check with your ITSP.
-----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Tony Graziano Sent: Sunday, October 02, 2011 4:29 PM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] Calls dropped during transfer 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/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
