Summary:	ASTERISK-15395: Dialout from Meetme conference
Reporter:	shrikant (sk)	Labels:
Date Opened:	2010-01-04 13:37:43.000-0600	Date Closed:	2011-06-07 14:08:03
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Applications/app_meetme
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
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.
Comments:	By: Leif Madsen (lmadsen) 2010-01-04 14:01:18.000-0600 I don't believe this is a bug -- it seems more like either a feature request or a support issue. I'd suggest you utilize the asterisk-users mailing list for assistance in this matter. By: shrikant (sk) 2010-01-04 14:02:39.000-0600 Sometime during the test i get it working but on the next test it fails. So i think this might be a bug in the asterisk source code By: Leif Madsen (lmadsen) 2010-01-05 08:36:10.000-0600 You're going to have to provide a lot more information about why you think that is the case, and why you don't think it could be something in your custom script? Please provide console output, SIP debugging, SIP history, a list of the technologies you're using to connect, the PHP script you're using to trigger the connection, the output of the AMI connections and what is being sent, etc... Need copies of both working and non-working versions. Please attach as files to this issue, in plain text, and not in a compressed (ZIP) file. By: Leif Madsen (lmadsen) 2010-03-23 09:42:32 Closed due to lack of feedback from the reporter. They are welcome to reopen the issue if they have the requested information. Thanks!