[asterisk-bugs] [Asterisk 0017372]: [patch] [regression] Progress in band error (don't send RTP packets)
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Jul 6 12:42:00 CDT 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=17372
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Reported By: tech_admin
Assigned To: jpeeler
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Project: Asterisk
Issue ID: 17372
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: assigned
Asterisk Version: 1.6.2.7
JIRA: SWP-1526
Regression: Yes
Reviewboard Link: https://reviewboard.asterisk.org/r/705/
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2010-05-21 09:31 CDT
Last Modified: 2010-07-06 12:41 CDT
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Summary: [patch] [regression] Progress in band error (don't
send RTP packets)
Description:
Dear all,
on a updated redhat rhel5 (kernel: 2.6.18-194.3.1.el5xen), I installed the
last asterisk release (1.6.2.7) from the source code.
The problem is: SIP caller doesn't hear ringing tone.
We use progressinband=yes with ch (Switzerland) tones and voice is G729
encoded using TC400B Digium card.
After answer, all is ok: both side RTP packets go trough asterisk without
translation.
The Digium card work fine with others service like IVR, Queues, Voicemail
... all using G729.
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Relationships ID Summary
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related to 0017185 [patch] [regression] Using Local channe...
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(0124264) jpeeler (administrator) - 2010-07-06 12:41
https://issues.asterisk.org/view.php?id=17372#c124264
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The call to ast_prod can not be moved back as it was intentionally moved to
prevent a deadlock scenario. Please test the latest patch as I believe it
solves the issue.
Issue History
Date Modified Username Field Change
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2010-07-06 12:41 jpeeler Note Added: 0124264
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