similar to: MeetMeAdmin() working problem

Is there any way that I can have an admin user hit * and then Mute all other users in a meetme conference? Sort of a moderator function? I know it can be done with MeetMeAdmin, but as I see it that requires a separate extension to dial, unless I've got the logic wrong? If it can be done in a single extension please show examples. Thanks. ________________________________ This e-mail,

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

Hello all, I am struggling with sending voicemail as an attachement in Email. When i have given the email like someone at gmail.com it is delivering to my gamil account perfectly(of course to spam folder). But when i given the email like someone at mycompanymail.com it is not delivering to my company email account.. What should i do ? Actually my company is using a third party email server..

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

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

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

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

Hello users, Actually i wanted to implement Text-To-Speech engine from cepstral voice using swift application i tried the documentation of doing this and i was unsuccessful at doing this work with asterisk can anybody please help me out finding the solution to installation thanks in advacnce srinivas Antarvedi -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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

Hello all, Here is the requirement from my side to use Asterisk.NET API to generate an automated call (outgoing) from asterisk and then link to one of the extensions which plays a sound file for the callee. For this i have worked out in the follwing way 1)modified manager.conf to facilitate this API to talk to asterisk 2)used the command Originate to call a Registered user under

Hello all i have an asterisk setup integrated with mysql via odbc driver myproblem is: when i reading my voicemails, in advanced options the following functions not working with realtime asterisk but working with flat files. 1. Reply to the message(option no:1) 2.Leave a message(option no:5) i have following settings in my general section _ searchcontexts=yes _sendvoicemail=yes [test1] 1001

Hello users, i am planning to forward my skype calls from skype to the asterisk registerd skype. The scenario is as follows. i)SkypeUserA calls SkypeUserB ii)SkypeUserB forwards his calls to SkypeUserC iii)SkypeUserC sees he got call from SkypeUserA. do i have a way to extract the SkypeUserB's details so that i can control who can forward the calls to my asterisk box. Thanks in

Hello users, Recently i have installed the free version of FaxForAsterisk and trying to work with it by sending a fax on T38. My version information is as follows i)Asterisk 1.6.0.20 ii)res_fax-1.6.0.14_1.1.6-x86_32 iii)res_fax_digium-1.6.0.14_1.1.6-i686_32 sip.conf [general] t38pt_udptl=yes extensions.conf [default] exten => _XXXXXXXXXX,1,NoOp(Fax Incoming Call) exten =>

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>

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

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

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

Hi! I am playing around with Asterisk and have a problem :-) (Asterisk-version: 1.2.4, chan_capi-cm-version: 0.6.4) I have a sip-phone at my desk and an ISDN-phone (independent of the Asterisk-server) in my living room, when I'm not at my desk, the sip-phone is switched off. I would like to be able to accept calls at both phones (when available) and have Voicemail kick in if I don't