Summary:	ASTERISK-15516: [regression] Attended transfer broken in 1.6.1.13
Reporter:	Morten Tryfoss (mtryfoss)	Labels:
Date Opened:	2010-01-26 01:08:42.000-0600	Date Closed:	2010-03-11 09:58:08.000-0600
Priority:	Blocker	Regression?	No
Status:	Closed/Complete	Components:	PBX/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	Error description: 1. A calls B 2. B picks up and talks to A 3. B does attended transfer to C 4. C picks up, but B still hears ringing 5. A and B are connected again (AT timeout exceeded on console) https://issues.asterisk.org/view.php?id=16513 is a similar problem, but I was told to file a different ticket since this is version 1.6.1.
Comments:	By: Walter Doekes (wdoekes) 2010-01-26 01:40:05.000-0600 I'm running 1.6.1.13 and it works like a charm over here:    -- Called +316xxx@trunk    -- Called +316xxx@world_out    -- SIP/trunk-00000071 is ringing    -- Local/+316xxx@world_out-666c;1 is ringing    -- SIP/trunk-00000071 answered Local/+316xxx@world_out-666c;2    -- Local/+316xxx@world_out-666c;1 answered SIP/phone2-00000070    -- Packet2Packet bridging SIP/phone2-00000070 and SIP/trunk-00000071  == Spawn extension (world_out, +316xxx, 1) exited non-zero on 'Local/+316xxx@world_out-666c;2'    -- Stopped music on hold on SIP/phone1-0000006e    -- Packet2Packet bridging SIP/phone1-0000006e and SIP/trunk-00000071 (this was the let-B-pickup-C-first case, the transfer-and-let-A-pickup-C works fine as well) By: Leif Madsen (lmadsen) 2010-01-26 07:58:43.000-0600 1) Is this a built-in transfer via features.conf, or is this a SIP transfer? 2) you need to provide console output showing the problem, along with debugging enabled, and the SIP console trace By: Morten Tryfoss (mtryfoss) 2010-01-26 12:08:53.000-0600 1) It's built-in transfer 2)    -- Started music on hold, class 'default', on IAX2/pstn2-1473    -- <Local/48999303@outgoing-85f0;1> Playing 'pbx-transfer.gsm' (language 'en')    -- Executing [111@transfertest:1] Goto("Local/111@transfertest-c6d5;2", "pstn1,41410986,1") in new stack    -- Goto (pstn1,41410986,1)    -- Executing [41410986@pstn1:1] UserEvent("Local/111@transfertest-c6d5;2", "dialednum,Dialednum: 41410986") in new stack    -- Executing [41410986@pstn1:2] Dial("Local/111@transfertest-c6d5;2", "IAX2/pstn1/41410986") in new stack    -- Called pstn1/41410986    -- Call accepted by 85.19.69.8 (format alaw)    -- Format for call is alaw    -- IAX2/pstn1-7354 is proceeding passing it to Local/111@transfertest-c6d5;2    -- IAX2/pstn1-7354 is ringing    -- IAX2/pstn1-7354 stopped sounds    -- IAX2/pstn1-7354 answered Local/111@transfertest-c6d5;2 [Jan 26 19:08:14] NOTICE[11822]: features.c:2161 ast_feature_request_and_dial: We exceeded our AT-timeout    -- Stopped music on hold on IAX2/pstn2-1473 By: Marcelo Terres (mhterres) 2010-01-27 09:18:54.000-0600 I have the same problem in asterisk 1.4.29. By: Tilghman Lesher (tilghman) 2010-02-02 11:41:20.000-0600 Please upgrade to the SVN version of the appropriate release branch.  I believe this may already be fixed. By: thefinch (thefinch) 2010-02-03 18:33:32.000-0600 Confirmed, fixed for svn 1.6.2 branch By: Leif Madsen (lmadsen) 2010-02-04 14:41:51.000-0600 Closed per the commit done for issue ASTERISK-15367