[asterisk-bugs] [Asterisk 0011421]: MeetMe conferences don't forward DTMF from SIP clients
noreply at bugs.digium.com
noreply at bugs.digium.com
Wed Feb 20 17:56:16 CST 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=11421
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Reported By: michael-fig
Assigned To:
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Project: Asterisk
Issue ID: 11421
Category: Applications/app_meetme
Reproducibility: always
Severity: major
Priority: normal
Status: acknowledged
Asterisk Version: 1.4.14
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 11-29-2007 16:32 CST
Last Modified: 02-20-2008 17:56 CST
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Summary: MeetMe conferences don't forward DTMF from SIP
clients
Description:
I'm using a MeetMe conference to connect an outbound call (over either SIP
or a Zap channel) with a SIP internal agent. I've turned on DTMF
forwarding for both users, but when I hit DTMF on any SIP client, it isn't
forwarded to the other end (the background noise is interrupted for dead
air for the duration of the keypress).
The reason I marked this as "major" is that my internal agents cannot
navigate voice menus on the outbound call, which is a big problem for us,
since many of the businesses we call have voice menus.
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dimas - 02-20-08 17:56
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It looks like a second half of your patch does not change anything really -
just moves code back and forth and changes formatting a bit.
Anyway this is one more reason of doing VDTMF emulation/normalization in
ast_write instead of ast_read. This is the question I emailed -dev list
(attn: Russell) yesterday. Asterisk core should handle this situation not
particular application.
I could try creating a patch to move emulation from ast_read to ast_write
if there is no objections on the approach.
Issue History
Date Modified Username Field Change
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02-20-08 17:56 dimas Note Added: 0082702
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