similar to: Trying to add MoH to conference bridge

Hi all, I'm trying to configure a few conference bridges. I've started with the very basic: [general] [default_bridge] type=bridge [default_user] type=user [default_bridge] type=bridge [5340] type=bridge However: confbridge list Conference Bridge Name Users Marked Locked? ================================ ====== ====== ======== *CLI> It doesn't seem to be

Hi all, I'm looking for some serious help. :) I couldn't find a better description for my problem... I think it is quite complex! Here's what I would like to achieve: A SIP caller dials into to my Asterisk 10. He will automatically listen to a specific MP3 stream. Other SIP callers dial also into my Asterisk. They all will automatically listen to the same MP3 stream. All

I am trying to get the "Mega Phone" demo working on my office PBX but there seems to be a problem when trying to set the default bridge to sfu mode. I have the following configuration in confbridge.conf in the default_bridge section: video_mode = sfu but when I do a "confbridge show profile bridge default_bridge" I see: Video Mode: no video I can change it

All, Thank you in advance for any help. I have a customer in need of a conferencing system. A requirement is for users to each have their own PIN for the same bridge. So, I put the list of users, PINs bridges into a MYSQL DB and used an ODBC connector to parse the table. Asterisk is connected and reads the rows as expected. The problem is that if a user enters a PIN that is NOT in the table,

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

Deep in the discussion of this bug, unfortunately mostly in private comments: https://bugzilla.redhat.com/show_bug.cgi?id=1594515 we decided it'd be more flexible for RHV if we had a way to map individual NICs to target networks and bridges. This can be done by adding a new --mac option so you can specify the exact mapping you need: $ virt-v2v [...] \ --mac

We upgraded from asterisk 11 to asterisk 13. Recordings were working fine in 11 but nothing is being written on 13. Here is the dialplan segment same => n,ExecIF($["${TL_PHONE_CALL_RECORD}"="TRUE"]?SET(CONFBRIDGE(bridge,record_conference)=yes)) same =>

To all: No matter what I try, I cannot get the system to wait for the admin to join. It just dumps users into the bridge directly. I do not have a pin for users, does that matter? What am I missing? Another issue the absolute timeout is not working ? ... have recordings that last for over 24 hours... and this should not happen... All calls should hangup after 4 ? Any ideas ? Any help is much

On 11/14/17 3:38 PM, Joshua Colp wrote: > On Tue, Nov 14, 2017, at 05:23 PM, Carlos Chavez wrote: >> I am trying to get the "Mega Phone" demo working on my office PBX >> but there seems to be a problem when trying to set the default bridge to >> sfu mode. I have the following configuration in confbridge.conf in the >> default_bridge section: video_mode

On 4/26/20 10:48 AM, Dovid Bender wrote: > 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

Using asterisk 16.1.1. I'm setting up a test using the cmp2k (Cyber Mega Phone 2K Ultimate Dynamic Edition). I have noticed Chrome 72 had some issues with video streams. I just upgraded to Chrome 73 and see they still have some issues. If I have 2 calls in a confbridge with video set to none. I then set the video source to a Chrome browser and the Remote Video shown to both calls from

I am doing dynamic conference bridges using confbridge in asterisk 11. Is there a way to toggle off an on recording of an ongoing conference call I have figured out how to record a conference if it is turned on when someone enters. Also I have noticed that when setting music_on_hold_class dynamically it does not override what is set on the channel. exten =>

/**************************************************************************** * * * Always Be Conferencing (ABC) * * * * Creator: chris @ Penguin PBX Solutions * *

Hi Joshua If i put the default_user option per endpoint would it work?? So what exactly does the contact_user option do? I know that in freeswitch there is the option extension-in-contact.We ?basically need to achieve the same functionality? Thanks<div> </div><div> </div><!-- originalMessage --><div>-------- Original message --------</div><div>From:

Hi I have two models . Dbase -------- name vendor type port defuser - default user name Users ------- username password email I need to build an association between these two in that the dbase model''s defuser needs to be present in the users table. I am from a relational database background so am trying hard to understand rails associations. What should I do to associate the

Hi Guys We are using the wizard to configure our pjsip trunk(see below) How do we get this setting to work contact_user=username We want to change the contact field in the sip invite to display the username of the trunk [trunk_defaults](!) type = wizard transport = transport-udp endpoint/allow_subscribe = no endpoint/allow = !all,g729 aor/qualify_frequency = 30

Hello, Anyone see this issue, I have a conference bridge setup for a church with a Barix unit that streams audio into the bridge. The bridge is started by calling in to a number that executes a call file and the system calls the Barix unit starting the broadcast. Users then call in and can listen to the sermons live. The system works flawless with 1 issue I can't get accurate cdr's. Every

Hi all, A new version (0.9.6) of DIAX is available for download at: http://www.laser.com/dante or http://www.geocities.com/tdanro There are no new functions, but some bugs fixed: What's new in version 0.9.6: - add Default_user locales as new language. The program language can be automatically selected based on default user locales on your system. You still can manually select the language,

I have used every asterisk 11.8.X version. Have not had an issue till 11.8.0 and 11.8.1 When I use ConfBridge I am attempting to put all participants in MUTE mode and just one talker... However, since 11.8.0 I am hearing feedback in the announcement like the channel is not really muted. I dropped back to 11.7.0 and I hear no feedback. Has something changed - or - am I not correctly setting up

Hello, when I call my conference, I see this error in my logs: ERROR: Function DENOISE not registered here is snippet from extensions.conf ... same => n,Set(CONFBRIDGE(user,announce_join_leave)=yes) same => n,Set(CONFBRIDGE(bridge,record_conference)=yes) same => n,Set(CONFBRIDGE(bridge,record_file)=/home/asterisk/file.wav) same => n,ConfBridge(1000) same =>