Summary:	ASTERISK-06244: Grandstream Devices don't hangup with Asterisk
Reporter:	Dan Journo (kesher)	Labels:
Date Opened:	2006-02-02 14:46:55.000-0600	Date Closed:	2011-06-07 14:02:53
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Core/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	When connected to Asterisk, a grandstream device will not disconnect a call while a Dial command in in operation. Please see steps to reproduce. Seems to only happen with grandstream devices. X-Lite seems to work fine. ****** STEPS TO REPRODUCE ****** Create the following dialplan: exten => 666,1,Ringing() exten => 666,2,Answer() exten => 666,3,Wait(1) exten => 666,4,Playback(pcfirst) exten => 666,5,ForkCDR() exten => 666,6,Dial(SIP/OUTBOUND_PROVIDER/PSTN_NUMBER,,m(default)) Then using a grandstream, dial 666. The call answers and plays back the "pcfirst" file. Then the music starts, and the outbound call is started. If you then hangup, the call does not terminate until the outbound call is answered. The call then instantly drops. However this can be annoying for the Callee as the caller has already hungup the phone.
Comments:	By: Matt O'Gorman (mogorman) 2006-02-02 15:11:57.000-0600 where are the steps/info to reproduce? By: Dan Journo (kesher) 2006-02-02 17:06:49.000-0600 Listed under Steps to Reproduce. Make sure you are looking at the Advanced view and not simple view. Here is a paste of the steps just in case you cant see them. Create the following dialplan: exten => 666,1,Ringing() exten => 666,2,Answer() exten => 666,3,Wait(1) exten => 666,4,Playback(pcfirst) exten => 666,5,ForkCDR() exten => 666,6,Dial(SIP/OUTBOUND_PROVIDER/PSTN_NUMBER,,m(default)) Then using a grandstream, dial 666. The call answers and plays back the "pcfirst" file. Then the music starts, and the outbound call is started. If you then hangup, the call does not terminate until the outbound call is answered. The call then instantly drops. However this can be annoying for the Callee as the caller has already hungup the phone. By: Olle Johansson (oej) 2006-02-03 01:13:06.000-0600 Please read the bug guidelines 1) This was reported in the wrong category 2) You did not add a SIP debug output We do need SIP debugs with debug set to 4, verbose set to 4 attached as separate files. Thank you. By: Olle Johansson (oej) 2006-02-09 08:35:10.000-0600 Can we get any more information on this bug? By: Mark Spencer (markster) 2006-02-20 12:18:42.000-0600 Suspending due to lack of debug information.