[asterisk-bugs] [Asterisk 0018916]: Flooding with chan_sip.c:6047 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Apr 5 14:29:52 CDT 2011
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=18916
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Reported By: yarns
Assigned To:
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Project: Asterisk
Issue ID: 18916
Category: Channels/chan_sip/CodecHandling
Reproducibility: always
Severity: minor
Priority: normal
Status: new
Asterisk Version: 1.8.3
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: 2011-03-03 05:55 CST
Last Modified: 2011-04-05 14:29 CDT
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Summary: Flooding with chan_sip.c:6047 sip_write: Asked to
transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x
Description:
Astersik flooding with message :
chan_sip.c:6047 sip_write: Asked to transmit frame type alaw, while native
formats is 0x100 (g729) read/write = 0x100 (g729)/0x100 (g729)
When user test dial at outtrunk:
Dial(SIP/somenumberhere at outtrunk,40,m);
i got many warninig messages like above. When we remove m(music on hold)
parameter everything works fine:
Dial(SIP/somenumberhere at outtrunk,40);
We have installed moh files in alaw and g729 formats, and we using g729
codec from: <link removed by lmadsen>
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(0133387) lmadsen (administrator) - 2011-04-05 14:29
https://issues.asterisk.org/view.php?id=18916#c133387
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<pabelanger> leifmadsen: I usually see that when I don't have all the
required codec modules loaded
<pabelanger> leifmadsen: what if he tried directmedia=no
Issue History
Date Modified Username Field Change
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2011-04-05 14:29 lmadsen Note Added: 0133387
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