[asterisk-bugs] [Asterisk 0014310]: No voice (ringing tone) after call was diverted
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Jan 23 13:07:18 CST 2009
The following issue has been RESOLVED.
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http://bugs.digium.com/view.php?id=14310
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Reported By: RadicAlish
Assigned To: file
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Project: Asterisk
Issue ID: 14310
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: resolved
Asterisk Version: 1.4.21.2
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
Resolution: fixed
Fixed in Version:
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Date Submitted: 2009-01-22 09:39 CST
Last Modified: 2009-01-23 13:07 CST
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Summary: No voice (ringing tone) after call was diverted
Description:
We have problem with one way voice in next scenario:
101 calls to IVR (number 1234) that have Answer() application and then
call directed to 103, 103 doesn't want to answer and divert call (SIP 302
Moved Temporarily) to 102,
101 stops to hear ringing tones and we get in CLI bulk of messages:
WARNING[5475] chan_sip.c: Asked to transmit frame type 64, while native
formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8)
when 102 answers audio come back to 101
I attached another log full.issue_14249.2
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(0098600) svnbot (reporter) - 2009-01-23 13:07
http://bugs.digium.com/view.php?id=14310#c98600
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Repository: asterisk
Revision: 170568
U branches/1.4/apps/app_dial.c
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r170568 | file | 2009-01-23 13:07:18 -0600 (Fri, 23 Jan 2009) | 4 lines
When a call is forwarded stop any active indications. The new channel will
provide an indication, if need be, itself.
(closes issue http://bugs.digium.com/view.php?id=14310)
Reported by: RadicAlish
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http://svn.digium.com/view/asterisk?view=rev&revision=170568
Issue History
Date Modified Username Field Change
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2009-01-23 13:07 svnbot Checkin
2009-01-23 13:07 svnbot Note Added: 0098600
2009-01-23 13:07 svnbot Status assigned => resolved
2009-01-23 13:07 svnbot Resolution open => fixed
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