[asterisk-bugs] [Asterisk 0015957]: Automatic progress indication breaks some scenarios
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Sep 24 14:44:44 CDT 2009
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=15957
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Reported By: lmadsen
Assigned To: tilghman
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Project: Asterisk
Issue ID: 15957
Category: Channels/General
Reproducibility: always
Severity: block
Priority: normal
Status: closed
Target Version: 1.6.1.7
Asterisk Version: 1.6.1.7-rc1
JIRA:
Regression: No
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: 2009-09-24 13:58 CDT
Last Modified: 2009-09-24 14:44 CDT
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Summary: Automatic progress indication breaks some scenarios
Description:
Per Pavel Troller on the Asterisk-Dev mailing list:
Hi!
I've noticed that in 1.6.1.7-rc1, an automatic sending of progress
indication
has been implemented for some applications, like Playback().
I have a problem with it. I was used to have full control over sending
any
backward signalling information, like Progress, Proceeding, Ringing,
Answer
etc. by my dialplan scripts. Now, this is over. Somebody decided that
Playback() will send progress, even if I explicitly don't want it, and my
scripts are not working anymore.
I have a real trouble with it. I'm using asterisk in a relatively
complex
heterogenous environment, full of various legacy devices like TDM PBXs
etc.
For example, Avaya Definity immediately stops transmitting dialling in
overlap
mode, when it receives a PROGRESS message over a PRI. Formerly, it was
normal
to send a tone/spoken messages during a dial phase. I still can send
tones
(because, thanks God, somebody forgot to add progress indication to the
PlayTones() app), but I cannot send a recording anymore, because Avaya
stops
dialling when it receives PROGRESS. Of course, there are channels, which
require sending progress info to successfully pass audio, like SIP, for
which
I was using the following in my macros:
exten => s,n,ExecIf($["${CUT(CHANNEL,/,1)}" = "SIP"]?Progress())
So, please, rethink this change if possible. In my opinion, it
substantially
reduces flexibility of Asterisk signalling possibilities.
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(0111337) svnbot (reporter) - 2009-09-24 14:44
https://issues.asterisk.org/view.php?id=15957#c111337
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Repository: asterisk
Revision: 220290
_U branches/1.6.0/
U branches/1.6.0/apps/app_disa.c
U branches/1.6.0/apps/app_playback.c
U branches/1.6.0/main/pbx.c
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r220290 | tilghman | 2009-09-24 14:44:44 -0500 (Thu, 24 Sep 2009) | 20
lines
Merged revisions 220289 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r220289 | tilghman | 2009-09-24 14:41:02 -0500 (Thu, 24 Sep 2009) | 13
lines
Merged revisions 220288 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) | 6
lines
Implicitly sending a progress signal breaks some applications.
Call Progress() in your dialplan if you explicitly want progress to be
sent.
(Reverts change 216430, closes issue
https://issues.asterisk.org/view.php?id=15957)
Reported by: Pavel Troller on the Asterisk-Dev mailing list
http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
........
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http://svn.digium.com/view/asterisk?view=rev&revision=220290
Issue History
Date Modified Username Field Change
======================================================================
2009-09-24 14:44 svnbot Checkin
2009-09-24 14:44 svnbot Note Added: 0111337
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