[asterisk-bugs] [Asterisk 0014815]: DTMF Appears to be broken from certain sources on asterisk 1.4.24 - double digit.
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Apr 28 23:26:42 CDT 2009
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=14815
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Reported By: geoff2010
Assigned To: file
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Project: Asterisk
Issue ID: 14815
Category: Core/RTP
Reproducibility: always
Severity: block
Priority: normal
Status: assigned
Target Version: 1.4.25
Asterisk Version: 1.4.24
Regression: Yes
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-04-02 15:38 CDT
Last Modified: 2009-04-28 23:26 CDT
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Summary: DTMF Appears to be broken from certain sources on
asterisk 1.4.24 - double digit.
Description:
I am having some trouble with 1.4.24. This appears to be a regression
since 1.4.21.1 as I have just recently upgraded and started to experience
the problem.
When calling my asterisk box from a cell phone the system detects double
DTMF digits. When calling from a landline or pure SIP device DTMF
detection works fine. I have determined that the cell phone is sending the
RTP DTMF payload about 2x more than the other devices, but as far as I can
tell it should still be considered a single digit as per the RFC.
Attached are two TXT files of pcap exports which show the dump for a
working DTMF interaction, and one for a failure situation.
Please let me know if I can provide any further information.
Thanks,
Geoff
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Relationships ID Summary
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related to 0014460 Asterisk plays a continuous tone foreve...
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(0103912) dimas (reporter) - 2009-04-28 23:26
http://bugs.digium.com/view.php?id=14815#c103912
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After looking more carefully at rtp.c I think it should be patched a little
bit different: dtmfcount should be replaced with field like "dtmf_end_ts"
which should be initialized with current timestamp + duration from DTMF
start packet (or some default if none). It should also be updated with
following DTMF packets. When current timestamp exceeds dtmf_end_ts,
Asterisk should forcibly end DTMF. Something like that.... But have to look
at least once into RTP specs though ;)
moliveras,
can you provide log similar to the one geoff2010 did? (containing both rtp
debug and dtmf debug)
If you you are hitting exactly the same issue, I could try creating a
patch. Wil take couple of days - kind of busy now...
Issue History
Date Modified Username Field Change
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2009-04-28 23:26 dimas Note Added: 0103912
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