[asterisk-bugs] [Asterisk 0010476]: RFC2833 DTMF digits seem to be recognized, but the resulting sequence missing some digits
noreply at bugs.digium.com
noreply at bugs.digium.com
Fri Aug 31 09:43:53 CDT 2007
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=10476
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Reported By: mdu113
Assigned To:
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Project: Asterisk
Issue ID: 10476
Category: Core/Channels
Reproducibility: sometimes
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.4.10.1
SVN Branch (only for SVN checkouts, not tarball releases): 1.4
SVN Revision (number only!): 79690
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 08-16-2007 16:43 CDT
Last Modified: 08-31-2007 09:43 CDT
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Summary: RFC2833 DTMF digits seem to be recognized, but the
resulting sequence missing some digits
Description:
This is very strange situation. DTMF digits seem to be recognized, but the
resulting sequence missing some digits.
Scenario: in VoiceMailMain i'm trying to dial a password. As you can see
in uploaded log.txt all digits of the password '0501' were recognized, at
least there's DTMF begin and DTMF end lines for them. Nonetheless it says
"-- Incorrect password '001' for user '13' (context = ace)"
It missed digit '5' when it checked the password.
This is intermittent, but usually I can replicate it once every 4/5
attempts.
This happens only with rfc2833 DTMF mode. Inband DTMF seems to be fine.
I'm not sure this is the right category I placed this report in as I'm not
sure what it is. I apologize if this is placement is incorrect.
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dtyoo - 08-31-07 09:43
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We are also experiencing something very similar to the problem reported by
mdu113 with polycom ip 430 and 501 phones connecting to asterisk 1.4.9. We
recently upgraded from 1.2 to 1.4 and did not have this issue previously.
As best I can tell the DTMF (rfc2833) is making it all the way from the
phone to the asterisk server, but then we see the voicemail application
report "Incorrect password" with typically the first digit of either the
mailbox or the password missing. Switching to inband improves but does not
eliminate this issue for us. Unfortunately, the polycoms don't support sip
info.
I'll post some of my trace / debug info in case anyone can get anything
out of it. The one other strange thing I have noticed a couple times is
when this issue happens, my asterisk console session hangs for a moment,
then comes back.
Issue History
Date Modified Username Field Change
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08-31-07 09:43 dtyoo Note Added: 0069780
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