[asterisk-bugs] [Asterisk 0017338]: No ringtone when going from queue to dial-command

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17338 
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Reported By:                jonaskellens
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17338
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.4.30 
JIRA:                       SWP-1469 
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:             2010-05-14 03:09 CDT
Last Modified:              2010-05-24 10:49 CDT
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Summary:                    No ringtone when going from queue to dial-command
Description: 
Asterisk 1.4.25.1 on CentOS 5.4

3 steps in my dialplan :

exten => _9,1,Dial(SIP/testcorp1,10)
exten => _9,n,Queue(testcorpq1,,,,10)
exten => _9,n,Dial(SIP/testcorp2,10)

In step 1, while dialing testcorp1, the caller hears a ringtone.
In step 2, while inside the queue, the caller hears music on hold.
In step 3, while dialing testcorp2, the caller hears silence.

Every time again, when going from a queue with musiconhold to a
dial-command, the caller hears no dialtone but silence...

The only time when the caller also hears a dialtone in step 3 is when
calling the Queue-application with 'r'-option : Queue(testcorpq1,r,,,10)

Don't know how to get the ringtone back...
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 (0122302) jonaskellens (reporter) - 2010-05-24 10:49
 https://issues.asterisk.org/view.php?id=17338#c122302 
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The Dial()-command does not have the 'r'-option. As I said earlier, it
makes no difference if the 'r'-option is used or not.

This issue should be kept open as Answer() should not be the solution to
this ringtone-problem. It means that even if there's no answer for minutes,
the caller will have to pay for these minutes as he is 'connected'.
This can not be how it should be, it's not clean.

jtodd, you say the first ringtone is generated by my SIP-phone
(Grandstream GXP2010) and the second by Asterisk. This can be through, but
then there's definitely a problem with Asterisk as the second ringtone is
not coming correctly from Asterisk.

Are there some parameters that I can tune for the ringtone that is coming
from Asterisk via a SIP-channel to my SIP-phone ? 

Issue History 
Date Modified    Username       Field                    Change               
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2010-05-24 10:49 jonaskellens   Note Added: 0122302                          
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