[asterisk-bugs] [Asterisk 0015405]: Early media causes "Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8)......"
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Jan 6 03:48:58 CST 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=15405
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Reported By: alecdavis
Assigned To:
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Project: Asterisk
Issue ID: 15405
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: acknowledged
Asterisk Version: SVN
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!): 203569
Request Review:
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Date Submitted: 2009-06-26 07:19 CDT
Last Modified: 2010-01-06 03:48 CST
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Summary: Early media causes "Asked to transmit frame type 64,
while native formats is 0x8 (alaw)(8)......"
Description:
Calling from a SIP phone to a local device or over an IAX trunk that sends
Early Media causes many messages on screen, and no audio.
Console Screen fills with, and no audio is heard.
[Jun 27 12:59:25] WARNING[2765]: chan_sip.c:6157 sip_write: Asked to
transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write =
0x8 (alaw)(8)/0x8 (alaw)(8)
[Jun 27 12:59:25] WARNING[2765]: chan_sip.c:6157 sip_write: Asked to
transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write =
0x8 (alaw)(8)/0x8 (alaw)(8)
[Jun 27 12:59:25] WARNING[2765]: chan_sip.c:6157 sip_write: Asked to
transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write =
0x8 (alaw)(8)/0x8 (alaw)(8)
[Jun 27 12:59:25] WARNING[2765]: chan_sip.c:6157 sip_write: Asked to
transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write =
0x8 (alaw)(8)/0x8 (alaw)(8)
Calling from FXS port to FXS port works great, a triple ring is heard at
callers handset.
This used to work when I originally submitted
https://issues.asterisk.org/view.php?id=14504
Another recent bug that may be related:
https://issues.asterisk.org/view.php?id=14310
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Relationships ID Summary
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related to 0014504 [patch] Allow app_dial to provide 'tone...
related to 0014310 No voice (ringing tone) after call was ...
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(0116105) alecdavis (reporter) - 2010-01-06 03:48
https://issues.asterisk.org/view.php?id=15405#c116105
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related: https://issues.asterisk.org/view.php?id=16086 No streaming musiconhold
when using dial command
Using trunk SVN-trunk-r237920M, and undoing the early
ast_channel_make_compatible() fix from
https://issues.asterisk.org/view.php?id=14504, I can confirm the problem
and get the same.
[Jan 6 22:40:20] WARNING[15914]: chan_sip.c:6526 sip_write: Asked to
transmit frame type slin, while native formats is 0x8 (alaw) read/write =
0x8 (alaw)/0x8 (alaw)
[Jan 6 22:40:21] WARNING[15914]: chan_sip.c:6526 sip_write: Asked to
transmit frame type slin, while native formats is 0x8 (alaw) read/write =
0x8 (alaw)/0x8 (alaw)
What I forgot to add in the previous note
https://issues.asterisk.org/view.php?id=15405#c116103, was when the called
party answers, ast_channel_make_compatible() is executed again.
Issue History
Date Modified Username Field Change
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2010-01-06 03:48 alecdavis Note Added: 0116105
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