similar to: How to kick/mute using ConfBridge application

Hi All, I am working on Asterisk 12.6.0 with ConfBridge module, when there are multiple user like admin and normal participant running with conference. When I try to kicked 2 user (Normal User), it play file "conf-kicked" and again join conference My scenario in confbridge like. 1] Admin User (e.g. SIP/8484-00000000) 2] Normal User (e.g. SIP/8484-00000001) 3] Admin User (e.g.

A very nice feature of another conferencing system that I've used is that the admin/moderator can press a star code to MUTE ALL OTHER USERS on the conference. This is great if you have several people on the call and one of the people puts the call on hold (and so the music/advertisement/your call is important/etc) message starts, or someone's cellphone handsfree unit in their car is

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, We use Asterisk extensively for conferencing - for the last 8 years or so this has been the 1.4/1.6/1.8 releases running chan_sip and meetme for up to around 350 concurrent users. Right around that number DAHDI hit's a hard coded memory limit and kicks allocation errors in the log. [Jun 22 10:04:13] WARNING[9095] app_meetme.c: Unable to open DAHDI pseudo channel: Cannot allocate

I have a confbridge setup that feeds the conference from the ALSA microphone input (this is the conference leader) and then uses app_ices to send the conference audio to icecast. I start the conference leader like this: console dial 1000_admin at conferences I join the ices user to the confbridge with a call file: Channel: Local/1000 at conferences MaxRetries: 2 RetryTime: 60 WaitTime: 30

Is there a way of temporarily suspending participants in a conference? Say I have 5 users, A,B,C,D,E and I wish A, B and C to have a discussion in the confbridge session that D and E can't hear, is there a way to suspend D and E for a while (whilst they are played music or whatever) and later join them back in? Failing that, I was considering kicking them and using an AGI script to rejoin

Hi All, The issue appearing at the random for confbridge module i.e. in some cases if a participant joins the confbridge, he/she unable to hear others which make him/her to hangup the call and redial the bridge again. By joining the bridge second time, participant able to hear the other participants. Any ideas which may causing this issue? As Asterisk version I'm using is 11.2.1. Is it a

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

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

Dear R-helpers, I am using confint() within a function, and I want to turn off the message it prints: x <- rnorm(100) y <- x^1.1+rnorm(100) nlsfit <- nls(y ~ g0*x^g1, start=list(g0=1,g1=1)) > confint(nlsfit) Waiting for profiling to be done... 2.5% 97.5% g0 0.4484198 1.143761 g1 1.0380479 2.370057 I cannot find any way to turn off 'Waiting for. .." I tried

I have a CyberPower PR3000LCDRTXL2U with a BP48V75ART2U expansion chassis, which I am monitoring using NUT 2.8.0 (on Gentoo Linux). TThe UPS appears to be telling me that the batteries need replacement due to age. CyberPower support has confirmed that for me and told me how i should be able to mute the alarm from the front panel until I can replace the batteries, but it doesn't appear

On Thu, Mar 21, 2024 at 12:23?PM Phil Stracchino via Nut-upsuser < nut-upsuser at alioth-lists.debian.net> wrote: > I have a CyberPower PR3000LCDRTXL2U with a BP48V75ART2U expansion > chassis, which I am monitoring using NUT 2.8.0 (on Gentoo Linux). TThe > UPS appears to be telling me that the batteries need replacement due to > age. CyberPower support has confirmed that for

Is there a way to make ConfBridge hang up on the final participant in a conference (obviously after some sort of initial grace period)? Background - I have just moved all of the phones in my house to separate extensions. As a replacement for the POTS-style shared line, I have implemented a "barge in" feature; any internal extension is able to join the call of any other internal

Hello, I have a really irritating issue that I haven't had time to investigate much - I hope someone has encountered it and can tell me a solution. I didn't see anything in searching archives / sites.. When my Nextel i90c phone gets a page (2 way text message via the internet option) it has an irritating tone to get me to hear it. However this tone seems to mute asterisk (reproducible).

A final note: http://lists.alioth.debian.org/pipermail/nut-upsuser/2014-April/008977.html That message talks about the Eaton Nova AVR which on the Eaton 5E product page it says the 5E replaces the Nova. Do they perhaps share hardware/firmware? Thanks, Leith Bade leith at leithalweapon.geek.nz On 10 August 2014 10:16, Charles Lepple <clepple at gmail.com> wrote: > On Jul 26, 2014, at

Hi, I know this is a little OT but there are many Asterisk users of the excellent Siemens DECT/VOIP phones like the S685IP and 475IP and this is probably newsworthy for them. One of the biggest bug bears has been no mute function on the handset. When I woke up this morning, the handset told me there was a firmware update. I updated and then visited the web site to find out what had been

11.7 is working fine for me. I put on 11.8 and my "confbridge" that should be "muted" to users now sounds un-muted or like I am getting feedback. Did something change there??? Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140310/7006af9c/attachment.html>

I'm considering implementing an Asterisk PBX for conferencing. Before I get started, I wanted to make sure that it supports the features that I need. I plan to use Asterisk as a conference bridge only. I want people to be able to use my conference to listen live to lectures/etc, without having to listen to others in the conference. I'm using the FreePBX web interface, and I can't

CentOS-6.6 Does anyone know of an application that would provide a 'mute' buttons for video and audio that could be used from a desktop panel? Ideally this would show the status (muted/cloaked or open mic/recording) of each. Presently, turning audio and video pick-ups off and on is rather cumbersome. And I often forget what state I have left them in. Which occasionally makes for some

Hi, Any suggestions to below problem? Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jamie Heckford Sent: 17 October 2006 21:48 To: asterisk-users@lists.digium.com Subject: [asterisk-users] FW: Why does One-Touch record mute/disconnect callif not dialed quick enough? Hi List, Have an odd problem