[asterisk-bugs] [Asterisk 0017096]: C keeps ringing when hanging A and B after blind transfer using atxfer

Tue Jan 18 12:04:42 CST 2011

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=17096 
====================================================================== 
Reported By:                gelo
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   17096
Category:                   Resources/res_features
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.4.30 
JIRA:                       SWP-1192 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-03-26 04:35 CDT
Last Modified:              2011-01-18 12:04 CST
====================================================================== 
Summary:                    C keeps ringing when hanging A and B after blind
transfer using atxfer
Description: 
The scenario is as follows:

1. A calls B
2. B answers, presses *2 for atxfer
3. While C is ringing, B hangs up

At this point, A and C are connected, so if C answers the call, A and C
speak without any problem.

But, if A hangs up while C is ringing, C keeps on ringing indefinitely
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0017999 Issues with DTMF triggered attended tra...
related to          0017192 [patch] Phone keeps ringing when hangup...
====================================================================== 

---------------------------------------------------------------------- 
 (0130620) svnbot (reporter) - 2011-01-18 12:04
 https://issues.asterisk.org/view.php?id=17096#c130620 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 302172

U   branches/1.4/res/res_features.c

------------------------------------------------------------------------
r302172 | rmudgett | 2011-01-18 12:04:37 -0600 (Tue, 18 Jan 2011) | 88
lines

Issues with DTMF triggered attended transfers.

Issue https://issues.asterisk.org/view.php?id=17999
1) A calls B. B answers.
2) B using DTMF dial *2 (code in features.conf for attended transfer).
3) A hears MOH. B dial number C
4) C ringing. A hears MOH.
5) B hangup. A still hears MOH. C ringing.
6) A hangup. C still ringing until "atxfernoanswertimeout" expires.
For v1.4 C will ring forever until C answers the dead line. (Issue
https://issues.asterisk.org/view.php?id=17096)

Problem: When A and B hangup, C is still ringing.

Issue https://issues.asterisk.org/view.php?id=18395
SIP call limit of B is 1
1. A call B, B answered
2. B *2(atxfer) call C
3. B hangup, C ringing
4. Timeout waiting for C to answer
5. Recall to B fails because B has reached its call limit.

Because B reached its call limit, it cannot do anything until the transfer
it started completes.

Issue https://issues.asterisk.org/view.php?id=17273
Same scenario as issue 18395 but party B is an FXS port.  Party B cannot
do anything until the transfer it started completes.  If B goes back off
hook before C answers, B hears ringback instead of the expected dialtone.

**********
Note for the issue https://issues.asterisk.org/view.php?id=17273 and
https://issues.asterisk.org/view.php?id=18395 fix:

DTMF attended transfer works within the channel bridge.  Unfortunately,
when either party A or B in the channel bridge hangs up, that channel is
not completely hung up until the transfer completes.  This is a real
problem depending upon the channel technology involved.

For chan_dahdi, the channel is crippled until the hangup is complete.
Either the channel is not useable (analog) or the protocol disconnect
messages are held up (PRI/BRI/SS7) and the media is not released.

For chan_sip, a call limit of one is going to block that endpoint from any
further calls until the hangup is complete.

For party A this is a minor problem.  The party A channel will only be in
this condition while party B is dialing and when party B and C are
conferring.  The conversation between party B and C is expected to be a
short one.  Party B is either asking a question of party C or announcing
party A.  Also party A does not have much incentive to hangup at this
point.

For party B this can be a major problem during a blonde transfer.  (A
blonde transfer is our term for an attended transfer that is converted
into a blind transfer.  :)) Party B could be the operator.  When party B
hangs up, he assumes that he is out of the original call entirely.  The
party B channel will be in this condition while party C is ringing, while
attempting to recall party B, and while waiting between call attempts.

WARNING:
The ATXFER_NULL_TECH conditional is a hack to fix the problem.  It will
replace the party B channel technology with a NULL channel driver to
complete hanging up the party B channel technology.  The consequences of
this code is that the 'h' extension will not be able to access any channel
technology specific information like SIP statistics for the call.

ATXFER_NULL_TECH is not defined by default.
**********

(closes issue https://issues.asterisk.org/view.php?id=17999)
Reported by: iskatel
Tested by: rmudgett
JIRA SWP-2246

(closes issue https://issues.asterisk.org/view.php?id=17096)
Reported by: gelo
Tested by: rmudgett
JIRA SWP-1192

(closes issue https://issues.asterisk.org/view.php?id=18395)
Reported by: shihchuan
Tested by: rmudgett

(closes issue https://issues.asterisk.org/view.php?id=17273)
Reported by: grecco
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/1047/

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=302172 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-01-18 12:04 svnbot         Checkin                                      
2011-01-18 12:04 svnbot         Note Added: 0130620                          
======================================================================