search for: unmuting

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

One of the features that I thought would be popular with the Web-MeetMe suite is the ability to start all non-admin callers in a muted state and selectively unmute them. For example any large conference that is of an announcment nature with a Q&A session. It's probably a feature I should have tested better, but I just discovered that a caller that is joined to a MeetMe with the |m flag

...ng the same DID number the administrator of the conference enters the conference by entering the admin password on dialing * (for menu option) he only can reduce/increse the volume and can lock and unlock the conference but no provision for the 1) kicking out a particular participant 2)Muting/unmuting of a particular participant so for the above things i should go for the MeetMeAdmin() but with a DID inward dialing if i want to call a particular command of MeetMeAdmin() exten => DID ,1,MeetMe(confnumber,command,user) This is an unfeasible solution Then how can i do that as i cannot cre...

Hi, Looking at https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration there is an option for admin_toggle_mute_participants however the non admin users can still toggle toggle_mute. Is there any option for the admin to disallow non admins from using toggle_mute to unmute themselves? If there isn't such an option on there any devs here that can ping me off line what it would

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

I tried using the iLBC codec, and whlie I like it, I ran into a strange issue. I did a few searches on Google and haven't found anyone with the same issue as this. Anyhow, I was using a Plantronics analog headset and box plugged into a Digium TDM card, dialed out over my VoIP provider's IAX channel to the PSTN. I was in a conference call which is running on an Avaya PBX (which

Hello All, Started out awhile ago trying to select columns in a dataframe whose names contain some variation of the word "mutant" using code like: names(KRASyn)[grep("muta", names(KRASyn))] The idea then would be to add together the various columns using code like: KRASyn$Mutant_comb <- rowSums(KRASyn[grep("muta", names(KRASyn))]) What I discovered though, is

Greetings, I have a somewhat unique (I think) configuration that I am testing involving MeetMe conferencing and have encountered a problem that I'm not quite sure how to solve. Here is a brief description of my setup for the background. I wanted to offer the ability for users to mute and unmute themselves while in a conference. If they enter a conference as monitor only, they are

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

here at wkdu.org our minds are blown trying to figure out what our problem is, we are using a dell optiplex gx1 with a soundblaster live card. we are getting a nasty hum on the line in. the thing is we plug a cd player directly into the soundcard the hum goes away and when we plug the line in into anything else there is no hum, so the question i ask where is the hum coming from? we are

I have a Debian Sarge box that I am trying to get icecast2 and ices2 up and running on. I'm having problems, but do not know where to look to identify the problem. I have my audio working with ALSA - microphone in plays through my audio out. I'm running a basic configuration using the defaults provided, and the ALSA example config file for ices. When I try to connect to the server using

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

...long with darkice.7.....if this > wasnt enough i have a a lit paper due tomm that has made my head hurt > on top of this...... Are you using ALSA? I had a weird problem with ALSA and hum on my SB Live. I finally figured out I had to unmute the last emu10k1 channel in alsamixer. Try muting/unmuting some of them if you are indeed using ALSA. HTH Alex. -- Mausoleum: The final and funniest folly of the rich. -- Ambrose Bierce --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to '...

I used to have sound when i first installed wine, it stopped after a crash, then i noticed that it auto mutes the sound when i run any program with wine, i unmute it but still cant get no sound. Does anyone know how to fix this? I have tried pasdp but i get the following error: bash: pasdp: command not found

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?Danny, Thank you for your prompt response. The way you are suggesting is great .?Infect?asterisk have its own functionality that if user presses *1 during meetme conferencing asterisk automatically unmute that user and user comes in talking mode.But it is not?fulfill my need. There is and issue that if 3-4 user presses *1 at the same time than how can i decide that who is asking the question

Hi, I have released AppKonference 2.5 today. This release fixes a bug that can cause audio problems when conference frame caching is enabled. It also fixes the spy feature so that more than one spyer can spy on a channel at the same time. If more than one spyer is unmuted, their audio is mixed and whispered to the spyee. -- Paul Albrecht

Putting a whole bunch of people into a listen-only/muted Confbridge conference or getting the broadcaster audio into a MOH class and then just having callers attach to that MOH class? Does the the muted side of a Confbridge Room still try to mix in audio from the muted channels or does it just disregard those channels and only run mixes against unmuted channels? Now, if the answer is

Cool! I'm still away from the office, but I was starting to work towards syching meetme2 up to the version of meetme in * 1.0.7. It is over a 2000 line diff, ignoring the database integration code, so it was looking like a not too trivial task. One question though, how difficult will it be to extend your latest version with the scheduling features I built on top of you previous version? I