would it matter how many codex you have enabled in your client/ua? On Nov 9, 2011 10:01 AM, "Joegen Baclor" <[email protected]> wrote:
> ** > Perhaps this would help. > http://wiki.sipfoundry.org/display/sipXecs/Call+Redirectors > > Each redirect is a hop. > > On 11/09/2011 08:14 PM, Paul Kramer wrote: > > 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/ >> >> >> > > _______________________________________________ > sipx-users mailing [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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