[asterisk-bugs] [Asterisk 0016856]: [regression] Blind transfers initiated from calling party aren't disconncted

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Mar 16 12:40:23 CDT 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16856 
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Reported By:                rsw686
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16856
Category:                   Applications/app_transfer
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.1.14 
JIRA:                       SWP-951 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-02-17 14:08 CST
Last Modified:              2010-03-16 12:40 CDT
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Summary:                    [regression] Blind transfers initiated from calling
party aren't disconncted
Description: 
Employee A calls a customer and talks for a few minutes. The customer asks
to talk to someone in another depart. Employee A does a blind transfer with
## to Employee B. The result is Employee B is connected to the customer.
However Employee A receives the all circuits are busy now message. What
should happen is Employee A is disconnected. I was originally using
Asterisk 1.6.1.4, but have tested Asterisk 1.6.1.14 with the same result.
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0014555 When i park a call after the slot annou...
====================================================================== 

---------------------------------------------------------------------- 
 (0119442) rsw686 (reporter) - 2010-03-16 12:40
 https://issues.asterisk.org/view.php?id=16856#c119442 
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I can reproduce this with the below basic dialplan on 1.6.1.18. 

8532 calls 8678
8532 then presses #https://issues.asterisk.org/view.php?id=8688 (## assigned as
the blind transfer code)
8678 is connected to 8688
8532 starts hearing busy tones as Asterisk didn't hangup the call

[default]
exten => 8678,1,Answer
exten => 8678,n,Dial(SIP/8678,,Ttr)
exten => 8678,n,Busy

exten => 8688,1,Answer
exten => 8688,n,Dial(SIP/8688,,Ttr)
exten => 8688,n,Busy

exten => 8532,1,Answer
exten => 8532,n,Dial(SIP/8532,,Ttr)
exten => 8532,n,Busy

Console output

  == Using SIP RTP CoS mark 5
    -- Executing [8678 at default:1] Answer("SIP/8532-0000001c", "") in new
stack
    -- Executing [8678 at default:2] Dial("SIP/8532-0000001c",
"SIP/8678,,Ttr") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 8678
    -- SIP/8678-0000001d is ringing
    -- SIP/8678-0000001d answered SIP/8532-0000001c
    -- Started music on hold, class 'default', on SIP/8678-0000001d
    -- <SIP/8532-0000001c> Playing 'pbx-transfer.gsm' (language 'en')
    -- Stopped music on hold on SIP/8678-0000001d
[Mar 16 13:37:51] DEBUG[28366]: features.c:1248 builtin_blindtransfer:
transferer=SIP/8532-0000001c; transferee=SIP/8678-0000001d; lastapp=Dial;
lastdata=SIP/8678,,Ttr; chan=SIP/8532-0000001c; dstchan=SIP/8678-0000001d
[Mar 16 13:37:51] DEBUG[28366]: features.c:1251 builtin_blindtransfer:
TRANSFEREE; lastapp=; lastdata=, chan=SIP/8678-0000001d; dstchan=
[Mar 16 13:37:51] DEBUG[28366]: features.c:1253 builtin_blindtransfer:
transferer_real_context=default; xferto=8688
    -- Transferring SIP/8678-0000001d to '8688' (context default) priority
1
    -- Executing [8678 at default:3] Busy("SIP/8532-0000001c", "") in new
stack
    -- Executing [8688 at default:1] Answer("SIP/8678-0000001d", "") in new
stack
    -- Executing [8688 at default:2] Dial("SIP/8678-0000001d",
"SIP/8688,,Ttr") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 8688
    -- SIP/8688-0000001e is ringing
    -- SIP/8688-0000001e answered SIP/8678-0000001d
  == Spawn extension (default, 8678, 3) exited non-zero on
'SIP/8532-0000001c'
  == Spawn extension (default, 8688, 2) exited non-zero on
'SIP/8678-0000001d' 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-03-16 12:40 rsw686         Note Added: 0119442                          
======================================================================




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