[asterisk-bugs] [Asterisk 0014365]: [patch] Add audio announce option to app_page.c
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Nov 19 11:29:25 CST 2009
The following issue is now READY FOR TESTING.
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https://issues.asterisk.org/view.php?id=14365
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Reported By: dferrer
Assigned To:
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Project: Asterisk
Issue ID: 14365
Category: Applications/NewFeature
Reproducibility: N/A
Severity: feature
Priority: normal
Status: ready for testing
Target Version: Feature Tracker
Asterisk Version: SVN
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 172132
Request Review:
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Date Submitted: 2009-01-29 09:21 CST
Last Modified: 2009-11-19 11:29 CST
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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.
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Issue History
Date Modified Username Field Change
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2009-11-19 11:29 lmadsen Status confirmed => ready for
testing
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