[asterisk-bugs] [Asterisk 0019372]: Attended transfer - transfering phone left connected

Asterisk Bug Tracker noreply at bugs.digium.com
Fri May 27 06:12:52 CDT 2011


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=19372 
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Reported By:                jamicque
Assigned To:                
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Project:                    Asterisk
Issue ID:                   19372
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.8.4.1 
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:             2011-05-26 09:52 CDT
Last Modified:              2011-05-27 06:12 CDT
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Summary:                    Attended transfer - transfering phone left connected
Description: 
The issue 0015833 still exists in newest version of Asterisk.

When doing a remote attended transfer in one of these 2 setups:

phones A,B,C --- proxy --- asterisks Z,X
when A->B call is on Z and B->C is on X, or:

phones A,B (with identity B1,B2), C --- asterisks Z,X
(A,B1 register on Z; B2,C on X)
when A->B1 call is on Z and B2->C is on X

In both scenarios Z and X are friends with no authentication needed.

The B phone doesn't get properly disconnected. asterisks invite/replace
each other properly and the audio channel is ok. B itself drops one of the
calls. But Z is not disconnecting B's call at all. You can replicate that
scenario with minimalistic dialplan - _X.,Dial(SIP/${EXTEN}) in default on
both sides.



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---------------------------------------------------------------------- 
 (0135480) davidw (reporter) - 2011-05-27 06:12
 https://issues.asterisk.org/view.php?id=19372#c135480 
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I can't understand the scenario here, but:

- at least up to 1.6.x asterisk cannot originate invite/replaces -
transfers are done as internal manipulations - the rationale is that it is
a back to back users agent, so you are transferring between the nearside
user agents on asterisk;
- if you have a proxy in this context, you have to make it translate any
replaces into the namespace understood by the asterisk to which you are
directing the request.

Things may have changed in 1.8, but I would have hoped to have become
aware of that, although we are locked into 1.6, at the moment. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-05-27 06:12 davidw         Note Added: 0135480                          
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