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

[John Todd]

>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

Try exhaustively checking  combinations of codec permissions to see 
if that makes a difference.

Set disallow=all allow=ulaw  in both the [general] section, and in 
each end device.  Change those settings around a bit, trying a call 
each time you change a setting.  Both the GS and the x-lite phone 
have some quirks with how they handle codecs, especially with *.

JT