[asterisk-bugs] [Asterisk 0016287]: One way audio after attended transfer
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Jul 6 10:53:46 CDT 2010
The issue 0017400 has been set as DUPLICATE OF the following issue.
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https://issues.asterisk.org/view.php?id=16287
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Reported By: cbkm
Assigned To:
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Project: Asterisk
Issue ID: 16287
Category: Channels/chan_sip/CodecHandling
Reproducibility: always
Severity: minor
Priority: normal
Status: acknowledged
Asterisk Version: Older 1.6.2 - please test a newer version
JIRA: SWP-442
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-11-19 11:58 CST
Last Modified: 2010-07-06 10:53 CDT
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Summary: One way audio after attended transfer
Description:
Possibly related to 14249 but I couldn't find a way to reopen that bug,
so...
Scenario:-
1. SIP/213 calls SIP/207
2. SIP/207 uses asterisk transfer functionality to transfer to SIP/214
3. SIP/207 hangs up
4. SIP/213 is now bridged to SIP/214
5. Console scrolls with:-
[Nov 19 17:31:59] WARNING[4263] chan_sip.c: Asked to transmit frame type
64, while native formats is 0x4 (ulaw)(4) read/write = 0x8 (alaw)(8)/0x4
(ulaw)(4)
6. If SIP/214 presses a key the message stops scrolling.
Attached is a noisy (verbose 9 / debug 9 / sip debug) log (on the
assumption I can attach a file once I hit submit).
This is with asterisk 1.6.2rc2 (with the patch from issue: 15848) NOT
1.6.2rc6 like it says in the version dropdown - sorry.
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Relationships ID Summary
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duplicate of 0017364 atxfer, one way sound, codecs
has duplicate 0017400 Writeformat Slin instead Alaw after att...
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Issue History
Date Modified Username Field Change
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2010-07-06 10:53 lmadsen Relationship added has duplicate 0017400
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