[asterisk-bugs] [Asterisk 0016086]: No streaming musiconhold when using dial command

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=16086 
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Reported By:                jongerenchaos
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16086
Category:                   Resources/res_musiconhold
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
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:             2009-10-16 13:11 CDT
Last Modified:              2009-10-18 15:27 CDT
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Summary:                    No streaming musiconhold when using dial command
Description: 
I use a shoutcast server incombination with asterisk. Asterisk can play the
shoutcast stream with the musiconhold application.

I use the application as following:

file /etc/asterisk/stream.sh
#!/bin/bash
mpg123 -q -r 8000 -f 8192 -s --mono http://91.121.208.150:8080

file /etc/asterisk/musiconhold.conf
[default]
mode=custom
dir=/var/lib/asterisk/mohmp3-empty
application=/etc/asterisk/stream.sh

In the dialplan i use:
exten => 1,1,Musiconhold(default)

This works great and i hear the shoutcast streaming solution.

With the command:
exten => 1,1,Dial(SIP/NUMBER at outbound,260,tm(default)) or
exten => 1,1,Dial(SIP/NUMBER at outbound,260,tm)
I get the error (no audio):
[Oct 16 20:02:15] WARNING[17784]: chan_sip.c:6308 sip_write: Asked to
transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write =
0x4 (ulaw)(4)/0x4 (ulaw)(4)

Is there a solution for it?
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---------------------------------------------------------------------- 
 (0112399) ebroad (manager) - 2009-10-18 15:27
 https://issues.asterisk.org/view.php?id=16086#c112399 
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I am unable to reproduce this. Can you please post a sip debug and a
verbose trace. You can accomplish this by issuing:

sip set debug on
core set verbose 3

in the Asterisk CLI.

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-10-18 15:27 ebroad         Note Added: 0112399                          
2009-10-18 15:27 ebroad         Status                   new => feedback     
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