[asterisk-bugs] [Asterisk 0015405]: Early media causes "Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8)......"

[Asterisk Bug Tracker]

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 04:26 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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 (0116108) alecdavis (reporter) - 2010-01-06 04:26
 https://issues.asterisk.org/view.php?id=15405#c116108 
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Became a regression with asterisk commit 186525

I take back what I suggested about branches, they are not affected.

This can now be closed. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-01-06 04:26 alecdavis      Note Added: 0116108                          
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