similar to: MeetMe Admin Functions

Hello users, I have been working with a conference setup. My setup includes: 1)There will be an interface number provided to the user which might be a DID number or A Toll free number When user calls the number it asks for the conference room number and the user pin . on successfull authentication he will be participated in the conference 2)by didaling the same DID number the

The meetmeadmin() dialplan function lets you specify a user to mute, un-mute or kick. But how do you get a list of users in your dialplan? When a user joins a conference, the user number assigned is "the last user number +1." If you have a long running conference with callers joining and leaving all the time, this can grow to be a large number. I want to be able to

Has anyone here done any work on clustering Meetme conferences over multiple Asterisk boxes? The scenario I am thinking of is where there are two or more boxes connected to a set of PRIs that all answer to the same PSTN number, and where it's not possible to know in advance on which box a call would arrive. So it would be possible to have some calls on one box and some on another, that should

Hi list, I am in need of a "special" asterisk conference room with the following constraints: - there is one admin / moderator and several "normal" callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the

Hi All, I am in desperate need of this feature. I want to play background music during a call while the 2 parties are having some lovely conversation (or maybe give them a sort of cursing background if they are cursing each other). I found this post which talks about creating a ghost call with the help of queues and putting that queue in a meetme room where queue will play the song/curse and the

Hello fellow asterisk users, I've been facing a problem when using MeetMe's admin functionality to unmute users in a conference using *Asterisk 1.6.2.11*. I've tried: 1) MeetMeUnmute (AMI) 2) MeetMeAdmin(AMI) 3) MeetMeChannelAdmin(AMI) and also tried via console : "asterisk -rx 'meetme unmute conf_no user_no'" and the available AGI functions. but all of this to no

Hi, I setup an MeetMe conference. So, the admin user calls and enter the conference in talk/listen mode. (Options : dAaxs) Then other users call the same conference and enters in muted mode (options: dlmx) How can the admin user decide, when he is ready to let everybody speaks ? I didn't find such option in the admin menu. Thanks -------------- next part -------------- An HTML

I'm looking at what would be involved in converting from MeetMe to ConfBridge and there seems to be a lot of missing administrative things, but I hope I'm just missing it. We all know about the missing realtime linkage. That's a major nuisance, but can be worked around. More serious is that the CLI command to display users in a ConfBridge don't show the caller ID information, so

Hi All I need to set my Asterisk conference such way that , during confernce Admin Can kick 1 or all user , Same for mute fuction.As well as Admin can increase or decrease conf & user volume. for that i used MeetMeAdmin like this exten => 600,1,MeetMeAdmin(1111,ekKLmMNS,7010) where 1111 is conf number & 7010 is Admin user

Hi people, I want to make a meetme between 2 numbers. First I enter the number1 into the meetme. It is waiting for the other number, but the other number never entered, so, how can I finish the meetme from the dialplan?. Is it posible by using MeetmeAdmin and kick all the users? Thanks, Anahi Ludue?a _________________________________________________________________ Descubre

hello i want to kick participant in a meeting by pressing the digit on sip phone.when i entry the meeting ,no matter how i press the button,the dtmf does not work. here is my dialplan and my agi script,and sip.conf [from-internal] exten =>121,1,MeetMeCount(900,CONFCOUNT) exten =>121,2,GotoIF($[${CONFCOUNT}<10]?3:100) exten =>121,3,Authenticate(123456) exten

Just FYI: IP address 89.248.168.176 has been trying to use the recently release SIP vulnerability in Asterisk to make outbound calls via our box. They are running a bank account callback scam. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Helpdesk: 817-310-4999 x3 Fax: 817-310-4990 Email: jmann at txhmg.com

built-in; This doesn=92t matter because the moderator would have to use meetmeadmin or the confbridge equivalent to control the other functions. The moderator would either need two phones or a phone and a web = interface. Let=92s say Yves=92 =93special conference=94 is 5555. The moderator = would start using this command Exten =3D> s,1,meetme(5555) The participants would do Exten =3D>

Would anybody mind if the the following command options where added to MeetMeAdmin? 0 - 9, * and # I'm considering hacking the code to add these commands to play the DTMFs to the specified user as tones and hope that the SIP or IAX channels then work with these correctly. -HJC

Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? It's only going to support 4-5 users(the voice channels won't all be active obviously). ________________________________ This e-mail, facsimile, or letter and any files or attachments

Hi, I'm new to asterisk and have a little trouble in developing my first more complex dialplan. The basic task is a click to call solution: - call one number via sip, play some announcements, do cdr etc. and put the callee into an conference room with music on hold - call a second number via sip, play some announcements, do cdr etc. put the callee into the same conference - have a nice chat

I am having problems mounting a DFS share, even though it works in smbclient fine. After mounting, the mount point (in ls) has question marks for everything, and when ls'ing I just get permission denied. Turning up debug and verbose don't seem to help. Any ideas ? [tom@charles-compaq@1306 /home/tom/Projects/gbb-core-app ] smbclient //exchsvr/dfs -U tchiverton -W BLUEFINGER Password:

Gentleman, Moderators, i don't know if this topic if OFF-Topic, if yes, please tell me. I had some difficult looking for a Asterisk software that provide me some functions (For exemple: CDR, Queue control, MeetMe Control) all-in-one. So i decided to develop than. In a few weeks i'll deploy a Beta version of this software and i'd like to know if is somebody available to try this

OK, I got my Cisco 7960's to auto-answer on the second line but I can't get the Asterisk to call all the lines at one time. I have 4 phones I would like all of then to answer when I dial x300. Any help would be great Thanks Tuska extensions.conf [conference] exten => 300,1,AGI(callall) exten => 300,2,MeetMe(300,dtqp) ; press # to exit the conference exten =>

Hello Folks, I am testing Asterisk 1.4.2 running on Fedora Core 5 (version 2.6-17). I have loaded the app_meet.so module in order to activate the MeetMe, MeetMeCount and MeetMeAdmin applications. While I have been successful in loading the app_meet.so module, I am experiencing an immediate kernel panic every time I try to make a call to a room conference. Is this story unique to me? How can