[asterisk-bugs] [Asterisk 0016542]: Dialout from Meetme conference

[Asterisk Bug Tracker]

The following issue has been SUBMITTED. 
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https://issues.asterisk.org/view.php?id=16542 
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Reported By:                SK
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16542
Category:                   Applications/app_meetme
Reproducibility:            always
Severity:                   block
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.28 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-01-04 13:37 CST
Last Modified:              2010-01-04 13:37 CST
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Summary:                    Dialout from Meetme conference
Description: 
I am implementing one dialer type of application.

In which i am first dialing one source number and sending it to conference
and then starting dialing the different destination numbers.

i have used meetme application of asterisk for this as i dont want to
disconnect the main source number.

Both numbers are being originated using AMI and in the context i have used
MeetMe with the specified room.
So, as soon as they got answered, they are being sent into the conference
and they can hear each other and talk.

Everything goes fine with the above mentioned settings.

But, what i am trying to achieve is, I want to dial the other number from
the meetme, so the source number can also hear the ringing sound of the
other phone. So, the call must be originated from conference and get into
conference as soon as the channel is originated.

For this i have used channel redirect function of phpagi-asmanager.php to
send the dialing channel of destination number to conference. By doing this
i can hear the ring sound, but when the call is being answered, the first
user of conference is not able to hear the voice of second user.

I checked the asterisk CLI and debug the issue, And i found that the
channel which we sent into conference without answer is having state
"Down". Can this be the issue of voice ?

Sometime during the test i get it working but on the next test it fails. 
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Issue History 
Date Modified    Username       Field                    Change               
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2010-01-04 13:37 SK             New Issue                                    
2010-01-04 13:37 SK             Asterisk Version          => 1.4.28          
2010-01-04 13:37 SK             Regression                => No              
2010-01-04 13:37 SK             SVN Branch (only for SVN checkouts, not tarball
releases) =>  1.4            
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