Summary:	ASTERISK-09976: Transfer Function not working
Reporter:	kkiely (kkiely)	Labels:
Date Opened:	2007-07-28 12:34:37	Date Closed:	2008-01-11 11:25:51.000-0600
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Applications/app_followme
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	When a target has been reached by the followme application, the transfer function dosent seem to work and exits not completing the dial step. exten => 401,1,followme(101205001|) [101205001] context=>101205-followme number=>301,35 [101205-followme] include => 101205-internal include => 101205-ld
Comments:	By: kkiely (kkiely) 2007-07-28 12:36:18 -- Playback of name file appears to be done.    -- <SIP/101205001-08213af0> Playing 'followme/options' (language 'en')    -- Stopped music on hold on SIP/101205004-b7226b28    -- Started music on hold, class 'default', on SIP/101205004-b7226b28    -- <SIP/101205001-08213af0> Playing 'pbx-transfer' (language 'en')    -- Stopped music on hold on SIP/101205004-b7226b28  == Spawn extension (101205-followme, 305, 0) exited non-zero on 'SIP/101205004-b7226b28' By: Digium Subversion (svnbot) 2007-07-30 11:53:44 Repository: asterisk Revision: 77778 ------------------------------------------------------------------------ r77778 | file | 2007-07-30 11:53:43 -0500 (Mon, 30 Jul 2007) | 4 lines (closes issue ASTERISK-9976) Reported by: kkiely Instead of directly mucking with the extension/context/priority of the channel we are transferring when it has a PBX simply call ast_async_goto on it. This will ensure that the channel gets handled properly and sent to the right place. ------------------------------------------------------------------------ By: Digium Subversion (svnbot) 2007-07-30 11:55:39 Repository: asterisk Revision: 77779 ------------------------------------------------------------------------ r77779 | file | 2007-07-30 11:55:38 -0500 (Mon, 30 Jul 2007) | 12 lines Merged revisions 77778 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77778 | file | 2007-07-30 14:11:02 -0300 (Mon, 30 Jul 2007) | 4 lines (closes issue ASTERISK-9976) Reported by: kkiely Instead of directly mucking with the extension/context/priority of the channel we are transferring when it has a PBX simply call ast_async_goto on it. This will ensure that the channel gets handled properly and sent to the right place. ........ ------------------------------------------------------------------------ By: Digium Subversion (svnbot) 2007-08-14 10:09:27 Repository: asterisk Revision: 79397 ------------------------------------------------------------------------ r79397 | file | 2007-08-14 10:09:26 -0500 (Tue, 14 Aug 2007) | 4 lines (closes issue ASTERISK-10052) Reported by: atis Revert fix for ASTERISK-9976 as it causes more issues then it solves. ------------------------------------------------------------------------ By: Digium Subversion (svnbot) 2007-08-14 10:12:17 Repository: asterisk Revision: 79403 ------------------------------------------------------------------------ r79403 | file | 2007-08-14 10:12:16 -0500 (Tue, 14 Aug 2007) | 12 lines Merged revisions 79397 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79397 | file | 2007-08-14 12:27:13 -0300 (Tue, 14 Aug 2007) | 4 lines (closes issue ASTERISK-10052) Reported by: atis Revert fix for ASTERISK-9976 as it causes more issues then it solves. ........ ------------------------------------------------------------------------ By: Digium Subversion (svnbot) 2008-01-11 11:20:46.000-0600 Repository: asterisk Revision: 98219 U   branches/1.4/apps/app_followme.c ------------------------------------------------------------------------ r98219 | file | 2008-01-11 11:20:45 -0600 (Fri, 11 Jan 2008) | 4 lines Ensure the return value of ast_bridge_call is passed back up as the application return value. This is needed for transfers to function so the PBX core knows to continue execution. (closes issue ASTERISK-9976) Reported by: kkiely ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=98219 By: Digium Subversion (svnbot) 2008-01-11 11:25:51.000-0600 Repository: asterisk Revision: 98220 _U  trunk/ U   trunk/apps/app_followme.c ------------------------------------------------------------------------ r98220 | file | 2008-01-11 11:25:50 -0600 (Fri, 11 Jan 2008) | 12 lines Merged revisions 98219 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4 lines Ensure the return value of ast_bridge_call is passed back up as the application return value. This is needed for transfers to function so the PBX core knows to continue execution. (closes issue ASTERISK-9976) Reported by: kkiely ........ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=98220