[asterisk-bugs] [Asterisk 0014365]: [patch] Add audio announce option to app_page.c

Asterisk Bug Tracker noreply at bugs.digium.com
Wed May 4 13:51:43 CDT 2011


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=14365 
====================================================================== 
Reported By:                dferrer
Assigned To:                jpeeler
====================================================================== 
Project:                    Asterisk
Issue ID:                   14365
Category:                   Applications/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     closed
Target Version:             1.8
Asterisk Version:           SVN 
JIRA:                       SWP-436 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 172132 
Request Review:              
Resolution:                 fixed
Fixed in Version:           
====================================================================== 
Date Submitted:             2009-01-29 09:21 CST
Last Modified:              2011-05-04 13:51 CDT
====================================================================== 
Summary:                    [patch] Add audio announce option to app_page.c
Description: 
It adds an extra option 'A(x)' to app_page (similar to Dial 'A(x)'), to
playback an announce simultaneously in all paged phones and (caller's one)
before conference bridge is activated.

Something like this:

exten => **,1,SIPAddHeader(Alert-Info: Ring Answer)
exten => **,n,Page(SIP/100&SIP/101,qA(chime))

where 'chime' is an audio file with a typical "airport" announce.

There's also another option 'n' that makes only to play announce in all
paged phones and not to play in caller's phone. If this was not specified
the announce were played simultaneously in SIP phone 100, 101 and in
caller's headset.
This is tipically used with Polycom phones, passing a SIP header that
makes the paged phones to autoanswer the call in hands-free speaker.

To acomplish this task, the only way that I found is to modify also
app_meetme, adding an option for play an arbitrary announce at start of
conference (G(x) option). The patch works for trunk, I also have a patch
for 1.4.23.

====================================================================== 

---------------------------------------------------------------------- 
 (0134447) svnbot (reporter) - 2011-05-04 13:51
 https://issues.asterisk.org/view.php?id=14365#c134447 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 316831

U   branches/1.8/apps/app_meetme.c

------------------------------------------------------------------------
r316831 | rmudgett | 2011-05-04 13:51:41 -0500 (Wed, 04 May 2011) | 9
lines

Wait for leader with Music On Hold allows crosstalk between participants.

Parenthesis in the wrong position.  Regression from issue
https://issues.asterisk.org/view.php?id=14365 when
expanding conference flags to use 64 bits.

(closes issue https://issues.asterisk.org/view.php?id=18418)
Reported by: MrHanMan
Tested by: rmudgett

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

http://svn.digium.com/view/asterisk?view=rev&revision=316831 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-05-04 13:51 svnbot         Checkin                                      
2011-05-04 13:51 svnbot         Note Added: 0134447                          
======================================================================




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