hey just change following<br><br><br>[status-one-en]<br>
exten => 100,1,Meetme (12345,qdM)<br>
<div id=":1op">
exten => 100,1,Hangup()<br>
</div><br><br><div id=":1op">
<br>
Channel: Local/100@status-one-en<br>
CallerID: Rick <5555555555><br>
MaxRetries: 0<br>
RetryTime: 15<br>
WaitTime: 45<br>
Application: Playback<br>
Data: my_status_message</div><br><br><div class="gmail_quote">On Mon, Apr 4, 2011 at 10:38 PM, D. Rick Anderson <span dir="ltr"><<a href="mailto:randerson@customteleconnect.com">randerson@customteleconnect.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">Ok, I've been running applications on 1.4 for quite some time using<br>
meetme to hold a person, while the person on the other end of the call<br>
accepts, etc. I was playing status messages to the calling party using a<br>
context like this:<br>
<br>
[status-one-en]<br>
exten => 100,1,Playback(my_status_message)<br>
exten => 100,1,Hangup()<br>
<br>
and then creating a call file like this:<br>
<br>
Channel: Local/100@status-one-en<br>
CallerID: Rick <5555555555><br>
MaxRetries: 0<br>
RetryTime: 15<br>
WaitTime: 45<br>
Application: MeetMe<br>
Data: 12345,qdM<br>
<br>
and it would hook into the meetme, play the message, then hangup and<br>
drop out.<br>
<br>
I've been building an application with 1.6, and this isn't working at<br>
all. In verbose mode, I see the message played, and the call hang up,<br>
but the music never even stops on the meetme. After about 20 seconds I<br>
get:<br>
<br>
Call failed to go through, reason (3) Remote end Ringing<br>
<br>
Is there some other way to do this in 1.6 that I'm unaware of? I've<br>
tried creating a context and extension for the meetme portion (rather<br>
than using the Application/Data in the call file, and switched the order<br>
around (which does cause the music to stop, but the announcement still<br>
doesn't get played, and I get the same call failed message). I've been<br>
googling on this for days now, and really just need to get it working.<br>
<br>
TIA<br>
<br>
Rick<br>
<br>
<br>
CONFIDENTIALITY / PRIVILEGE NOTICE: This transmission and any attachments are intended solely for the addressee. This transmission is covered by the Electronic Communications Privacy Act, 18 U.S.C §§ 2510-2521. The information contained in this transmission is confidential in nature and protected from further use or disclosure under U.S. Pub. L. 106-102, 113 U.S. Stat. 1338 (1999), and may be subject to attorney-client or other legal privilege. Your use or disclosure of this information for any purpose other than that intended by its transmittal is strictly prohibited, and may subject you to fines and/or penalties under federal and state law. If you are not the intended recipient of this transmission, please DESTROY ALL COPIES RECEIVED and confirm destruction to the sender via return transmittal.<br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a></blockquote></div><br>