Summary:ASTERISK-15516: [regression] Attended transfer broken in
Reporter:Morten Tryfoss (mtryfoss)Labels:
Date Opened:2010-01-26 01:08:42.000-0600Date Closed:2010-03-11 09:58:08.000-0600
Versions:Frequency of
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 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

   -- 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 (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