[asterisk-bugs] [Asterisk 0018379]: attended transfer weird behaviour

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Dec 7 11:13:47 CST 2010


The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=18379 
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Reported By:                gincantalupo
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18379
Category:                   Applications/app_dial
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.8.1-rc1 
JIRA:                        
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:             2010-11-25 11:35 CST
Last Modified:              2010-12-07 11:13 CST
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Summary:                    attended transfer weird behaviour
Description: 
Just installed 1.8.1-rc1 and tried the attended transfer function with 3
snoms (firmware 8.x), A,B and C. When A calls B and B transfers to C but C
is busy or does not answer, 'pbx-invalid.gsm' sound is played...but the
called number is right!

Another test: when I try to transfer the call to a wrong number I get this
message:
WARNING[31448]: features.c:1861 builtin_atxfer: Did not read data
and after that the call is bounced back to the transferrer (shouldn't
Asterisk say invalid extension???)

My test extensions:
exten => 12,1,Dial(SIP/81,5,tT)
exten => 12,2,NoOp(${DIALSTATUS})
exten => 12,3,Hangup

exten => 14,1,Dial(SIP/8,5,tT)
exten => 14,2,NoOp(${DIALSTATUS})
exten => 14,3,Hangup

exten => 17,1,Dial(SIP/70,5,tT)
exten => 17,2,NoOp(${DIALSTATUS})
exten => 17,3,Hangup

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Relationships       ID      Summary
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related to          0018254 Attended transfer failure
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---------------------------------------------------------------------- 
 (0129406) lmadsen (administrator) - 2010-12-07 11:13
 https://issues.asterisk.org/view.php?id=18379#c129406 
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Please provide a SIP trace, console trace (with debug level logging) and a
trace with SIP history enabled in sip.conf.

Without that information it is going to be difficult (if not impossible)
to reproduce this without your hardware. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-12-07 11:13 lmadsen        Note Added: 0129406                          
2010-12-07 11:13 lmadsen        Status                   new => feedback     
======================================================================




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