[asterisk-bugs] [Asterisk 0017273]: atxfer *2 channel dahdi FXS no hangup

Asterisk Bug Tracker noreply at bugs.digium.com
Mon May 3 09:29:28 CDT 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17273 
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Reported By:                grecco
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17273
Category:                   Channels/chan_dahdi
Reproducibility:            have not tried
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.2.7-rc3 
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:             2010-05-01 17:06 CDT
Last Modified:              2010-05-03 09:29 CDT
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Summary:                    atxfer *2 channel dahdi FXS no hangup
Description: 
Attended transfer *2 (feature atxfer=*2), channel dahdi port FXS.

A call B
B aswered A
B *2 (atxfer) call C 
C rining (no aswer)
B hangup 

B it keeps calling, getting blocked


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---------------------------------------------------------------------- 
 (0121288) pabelanger (manager) - 2010-05-03 09:29
 https://issues.asterisk.org/view.php?id=17273#c121288 
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We are lacking information here (see below).  Are all the phones DAHDI? We
will need traces, debug output and dialplans.

http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt

---
Thank you for taking the time to report this bug and helping to make
Asterisk better. 

Unfortunately, we cannot work on this bug because your description did not
include enough information. 

You may find it helpful to read the Asterisk Issue Guidelines
http://www.asterisk.org/developers/bug-guidelines. 

We\'d be grateful if you would then provide a more complete description of
the problem.

At a minimum, we need:
1. the specific steps or actions you took that caused you to encounter the
problem,
2. the behavior you expected, and
3. the behavior you actually encountered (in as much detail as possible).

This likely includes output from the console with debug level logging, a
SIP trace (if this is SIP related), and configuration information such as
dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf).

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-05-03 09:29 pabelanger     Note Added: 0121288                          
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