[asterisk-bugs] [Asterisk 0016287]: One way audio after attended transfer
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Nov 19 11:58:21 CST 2009
The following issue has been SUBMITTED.
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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: new
Asterisk Version: 1.6.2.0-rc6
JIRA:
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: 2009-11-19 11:58 CST
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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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Issue History
Date Modified Username Field Change
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2009-11-19 11:58 cbkm New Issue
2009-11-19 11:58 cbkm Asterisk Version => 1.6.2.0-rc6
2009-11-19 11:58 cbkm Regression => No
2009-11-19 11:58 cbkm SVN Branch (only for SVN checkouts, not tarball
releases) => N/A
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