On Fri, 2009-02-27 at 16:24 -0800, Chris St Denis wrote: > Scott Lawrence wrote: > > On Fri, 2009-02-27 at 14:37 -0800, Chris St Denis wrote: > > > > > SipX: 3.10 > > > Phones: Polycom 501/601. 3.x firmware > > > Gateway: AudioCodes Mediant PRI gateway > > > > > > I have configured a call forward to a pstn number (my cell) for one of > > > my extensions from the sipx user login. > > > > > > * Calls to that extension from other internal phones correctly forward > > > to the cell phone. > > > * Incoming calls to the general number (hunt group) that are then > > > forwarded by a person to that extension correctly forward to the cell > > > phone. > > > * Calls to that phone's direct number (set as an alias) go to voicemail > > > instead of forwarding. > > > > > > How can I fix this? > > > > > > > Can you more specifically describe the failure case? Exactly what > > values are put in exactly what configuration fields, and exactly what > > address is dialed for the call that goes to voicemail? > > > > sorry to seem pedantic, but there are lots of places to configure things > > and your description is not specific enough to know what you've really > > done > I had thought this was clear, but I'll try to be more detailed. I was > hoping there was just something simple/obvious I was missing.
That all looks correct. To debug this, you'll need to trace the message flow and see what's going wrong. See: http://sipx-wiki.calivia.com/index.php/Display_SIP_message_flow_using_Sipviewer#Getting_SIP_Messages_to_display when you get the trace data, take a look at it using sipviewer and/or post the trace with a description of your configuration (identify components by IP address), what you were doing, and which call in the trace you're talking about (by call-id or frame number in the trace, preferably). _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
