Re: [asterisk-users] How to MessageSend to a SIP from AMI Or CLI?
I haven't used send text before in the ami but I would expect the channel to be the fully qualified one. When you do core show channels in the cli you get a longer channel name that's unique. I'm not 100% on this as I haven't used it but when in the ami it says to send channel, it usually means the unique current channel. The Asterisk wiki at wiki.asterisk.org is pretty good at showing you what's required, although their examples could be better, Hope this helps, Dan Jenkins On Nov 18, 2012, at 6:36, Face falaz...@gmail.com wrote: Hello all, I am running Asterisk 10.10.0 and I can send Message between SIP's no problem. However, I would like to be able to send send Message to a SIP from AMI Or CLI. I check the ListCommands On the AMI and it don't have MessageSend. Therefore, I try the SendText. AMI: Action: SendText Channel: SIP/600 Body: This is a test. Message: This is a test. Extension: 600; Response: Error Message: No such channel Any help would be much appreciated. -- Sincerely, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conf into a call in progress
Gentlemen,
So, from your answers I understand that I have 2 options:
1. AMI Redirect command
2. Asterisk command ChannelRedirect
I'm inclined to prefer the 2nd option, as we've never used AMI, but I don't
know if it can be web-initiated.
Basically, we have the following php code:
?php
if( $_POST[ext] || $_POST[dest] )
{
echo Selected extension: . $_POST['ext'] . br /;
echo Destination Number: . $_POST['dest'] . br /;
// Enter the selected extension's active call to a conference room
// Establish a new call to the entered number
// Connect the new call to the selected extension's conference room
// Put a disconnect/split button
echo connecting...br /;
exit();
}
?
html
body
form action=?php $_PHP_SELF ? method=POST
Extension:
select name=ext
option value=50015001/option
option value=50025002/option
option value=50035003/option
option value=50045004/option
/select
Destination Number: input type=text name=dest /
input type=Conf /
/form
/body
/html
To this code, we need to add the relevant functions to perform the tasks
specified above, i.e. enter the active call to a conf room, establish the
call to the new number and then, link them all together and give an option
to quit the conference.
What do you suggest?
Thanks,
Michael
On Fri, Nov 16, 2012 at 1:17 PM, Aldo Bergamini aabe...@gmail.com wrote:
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Re: [asterisk-users] How to MessageSend to a SIP from AMI Or CLI?
On 11/18/2012 05:17 AM, Dan Jenkins wrote: snip Hello all, I am running Asterisk 10.10.0 and I can send Message between SIP's no problem. However, I would like to be able to send send Message to a SIP from AMI Or CLI. I check the ListCommands On the AMI and it don't have MessageSend. Therefore, I try the SendText. snip In addition to what Dan said, the MessageSend AMI action was added in Asterisk 11: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_MessageSend While that doesn't help you if you're on Asterisk 10, it will help you if you choose to migrate to Asterisk 11. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] meetme race condition
I think I have a race condition. I am running something like this in my dialplan call agi to bring my list of devices into my MeetMe Playback beep start MeetMe() So in fact the meetme is not started before I bring the list of devices into the meetme. How can I do this differently so the MeetMe is started first or how can I wait in my AGI on the MeetMe to start because the MeetMe wont start until I exit the AGI... - or how do I in the dialplan wait for for the Meetme because I do have a stage where I redirect the Call into the MeetMe. so how do I inject a line that waits there for the MeetMe to be active??? THanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
