[asterisk-bugs] [Asterisk 0017273]: atxfer *2 channel dahdi FXS no hangup
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Jan 18 12:07:20 CST 2011
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: closed
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:
Resolution: fixed
Fixed in Version:
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Date Submitted: 2010-05-01 17:06 CDT
Last Modified: 2011-01-18 12:07 CST
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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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Relationships ID Summary
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related to 0018395 C should not receive request call again...
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(0130626) svnbot (reporter) - 2011-01-18 12:07
https://issues.asterisk.org/view.php?id=17273#c130626
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Repository: asterisk
Revision: 302173
_U branches/1.6.2/
U branches/1.6.2/main/features.c
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r302173 | rmudgett | 2011-01-18 12:07:16 -0600 (Tue, 18 Jan 2011) | 95
lines
Merged revisions 302172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r302172 | rmudgett | 2011-01-18 12:04:36 -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/
........
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http://svn.digium.com/view/asterisk?view=rev&revision=302173
Issue History
Date Modified Username Field Change
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2011-01-18 12:07 svnbot Checkin
2011-01-18 12:07 svnbot Note Added: 0130626
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