Thank you both for your kind input. I'm very impressed at the promptness of replies on the list so far!
I've created a new issue at http://track.sipfoundry.org/browse/XX-9954, however this is the first issue I have created so there could be problems with it. Apart from the "Too many hops" error message in the sipxproxy log, was there any other indication that the problem could be related to max-forwards? In the pcap file I can see the original invite from the ITSP contains a max-forwards value of 10 which i'm guessing also helped in diagnosing the problem. Do you know if there's a resource available describinig the flow of data between each SipXecs component? I'm looking at the merged.xml file in sipviewer and must say i'm quite confused. Thanks again for your help. Paul On Wed, Nov 9, 2011 at 9:27 PM, Joegen Baclor <[email protected]> wrote: > ** > Max-Forwards strikes again. Your ITSP is sending a Max-Forwards of 10 > which is consumed by the hops required to eventually land the call to the > IVR. The reason why it does not show when not using aliases is because it > takes a lesser amount of hop and does not consume the max forwards. I > have discussed this with the developers and we agreed that we can introduce > a reset-max-forwards=value parameter in the ITSP account setting. This > would allow calls coming from these ITSPs with very low max forwards to be > rewritten. > > > Please create a tracker for this in jira and attach the logs oisted here. > > > > On 11/09/2011 04:19 PM, Paul Kramer wrote: > > Ok here is the log with enhanced debugging. I had a quick look at the > voicemail dialplan and everything seemed to be in order. > > On Tue, Nov 8, 2011 at 1:18 PM, Paul Kramer <[email protected]> wrote: > >> Hi all, this is my first post on the forums and I'm a little wet behind >> the ears - might have to be a bit patient with me [image: Laughing] >> >> I'm having an issue with calls made from the PSTN coming in through the >> Internode (Australian) ITSP trunk I have setup in sipxecs. When I forward >> calls to an extension via System -> Servers -> *server* -> Sip Trunking -> >> Incoming Calls Destination, calls forward to VM successfully. However when >> I apply my ITSP DID to any extension, the extension rings and a call can be >> established but when the call is forwarded to VM, it is dropped at the far >> end. VM forwarding also works as expected during internal calls. >> >> Attached are two sets of logs and traces: one with aliases configured and >> one with calls passed straight through to an extension. >> >> Thanks in advance for your help. >> >> Paul > > > > _______________________________________________ > sipx-users mailing [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > >
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