[Asterisk-Users] SIP Calls Don't Properly Connect (Continue Ringing) After CVS Update

[Steven M. Sokol]

Hi,

I just updated my image from CVS, compiled and reinstalled it.  Now
whenever I make calls from my Grandstream phone to my X-Lite soft-phone,
the call does not complete correctly.

Scenario:

1.	I take the GS off hook and dial 1100 (the extension of the
x-lite phone).
2.	The x-lite phone rings properly.
3.	The user at the x-lite site answers the call.
4.	The GS phone continues to "ringback" and does not detect that
the call is complete.
5.	After about 10 seconds the GS plays busy and the x-lite detects
hangup.
6.    The x-lite goes back on hook.

This scenario was working properly (the call completed as expected)
prior to the CVS update.  Oddly, calls from x-lite to the GS complete
properly and without incident.  The big difference is that there is a
"precursor" script on the GS extension that answers and plays the use's
name using the name file in the voicemail folder.  THEN it uses Dial to
send the call to the SIP device.

I swear there was a thread about this last week but I can't find it for
the life of me.  Perhaps it was in the error log at Digium.

Any thoughts?

Thanks - Steve