similar to: special conference room

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 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

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,

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

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

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

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

Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* *** -------------- next

sorry... typo.... the problematic phone has the 192.168.0.13 the asterisk has 192.168.1.211 when i connect a snom phone on the cable that was in the soundstation 6000 before and configure the phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP... it would be helpful if someone, that has a running soundstation ip 6000 could send the configuration... :-/ regards, yves Am

hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . ?service zaptel restart? or there is any other command Thanks and regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Mark, yes, you are right... these are different VLANs I configured the other phone to use the same IP (192.168.1.13)... and it worked flawlessly... on the SAME Networkcable in the same plug... so it must have something to do with the polycom phone config... remember... when I use tcp the phone tries to register, but does not even try with udp... thank you, yves Am 21.12.2016 um 13:34

hi, I have a really strange problem which is driving me crazy for days now. If I register my asterisk (tried all versions from 1.6 up to 13.x) with one sip registrar, everything works... calls go out and call come in... no 32 seconds limit. but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. as far as I know, there is no firewall in

Hi My System is: Asterisk 1.6 running on a Dell Server with two network interfaces. eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has the local ip 10.26.208.252 and the external ip 89.244.x.y eth0 of the server is configured to 10.26.192.107 The Problem: SIP registration works, phone rings in- and outbound, but there is no audio, nor the caller neither the callee can hear

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

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

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

Hello, We are creating a web-based conferencing application using Asterisk as the voice conferencing server. This as an open source project. We are trying to determine if there is interest of the community and perhaps work together to improve the application. Using the web application, you can upload your powerpoint presentation, manage the participants in the conference thru the web interface

Hi, I migrated from asterisk 1.6 to 11.3. The Server has a Sangoma A104 quadPri card installed. OS is a fresh installed Ubuntu 12.04 64bit libpri, dahdi etc. all latest releases.. Sangoma says... driver is compatible with ANY asterisk version... I tried driver 3.5.8... Setup ended with error. I tried (latest) driver 7.0.1 Setup went through, Asterisk is showing dahdi channels... all fine....

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

Hello, ? I am interested, too. ? Att, Welinghton Citando Mitul Limbani <mitul at enterux.in>: > Hey Helvio, > > Would like to check it out as well. > > Do email me, > > Mitul > On 22-Jun-2015 9:05 AM, "Helvio Junior" <helvio.listas at gmail.com> wrote: > >> Gentleman, >> >> Moderators, i don't know if this topic