similar to: Start/Stop recording in confbridge

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 =>

Hi All, I want to set the video_mode of the confbridge dynamically in the dialplan. SO say if 5 users join the conference with follow_talker mode, it should work like that (and it does). But if 6th user changes the video_mode to first_marked and gets marked in the dial plan and joins the conference, he does not become the single video source of the conf. The video mode stays follow_talker. I

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

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

I'm running 10.7.1 (yes, I know it's old, but this may be a problem in later versions too) and had a conference being recorded via: Set(CONFBRIDGE(bridge,record_conference)=yes) The bridge started out at 8KHz despite one HD device. But when the second came in (G.722), it switched to 16KHz. At that point, the recording file had the bitrate change in the middle. That seems wrong.

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 =>

Hi everyone, I haven't used Asterisk in many years, but in searching for a good podcasting solution that will allow me to record three or four participants to individual tracks (which would allow me to go in and do noise removal on each participant individually, giving a higher quality), I came up with the idea to use Asterisk. Now I've installed it and got it all set up and did a test

Hi folks, I have one server with multiple companies (multi-tenant). >From AMI I get all events of all extensions so any one that connect can see other extensions, from different company (context). How can I limit specific user to get just specific context? Sam -------------- next part -------------- An HTML attachment was scrubbed... URL:

We have a system where two calls are in a ConfBridge with recording. This is Asterisk 16.3.0 Channel A seems to work perfectly. Wireshark is showing the RTP to/from working fine and having no jitter/lag issues. This call hears everything from channel B. Channel B we have more issues capturing a wireshark trace because their channel can be in the system for hours. When the two calls are in the

I want to set up a conference call to be recorded automatically, so I'd like the recording to start when the second caller joins the conference (one caller already there). The recording would continue until the last user hangs up. How can you determine how many are already in the conference bridge? [conferences] exten => 66,1,Answer exten => 66,n,Wait(1) exten =>

I'm trying to migrate from MeetMe to ConfBridge: [conferences] exten=>_8[1-9],1,Answer() ;;exten=>_8[1-9],1,Meetme(${EXTEN},1Ms,1234) exten=>_8[1-9],2,ConfBridge(${EXTEN},1Ms) exten=>_8[1-9],n,Hangup And that works. Also changed the hints: ;;exten => 81,hint,MeetMe:81 exten => 81,hint,ConfBridge:81 ;;exten => 82,hint,MeetMe:82 exten => 82,hint,ConfBridge:82 ;;exten

running Asterisk 1.8.9.0-rc2, what are the ways to interface with ConfBridge ? I see the CLI command 'confbridge' documented for asterisk 10, but i dont see how to interface with confbridge on 1.8 What I'm trying to do is keep track of conferences that are used. I tried something like the below, but not only does Confbridge not return, but i'd need something that erases the

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 =>

Hi folks, We are upgrading from AMI 1.0 to AMI 1.3 and looking for any documents or AMI 1.3 Specifications. I found AMI 1.4 Specification in wiki.asterisk.org but not for AMI 1.3. Can someone provide me the link for AMI 1.3 specification ? Thanks in advance Shishir -------------- next part -------------- An HTML attachment was scrubbed... URL:

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.

We have a product that uses Asterisk via AMI. I am relatively certain we used to be able to play prompts by actions like the following to make asterisk play the confbridge-join prompt when a new user joins the confbridge. However, that doesn't seem to work now. Action: SetVar ActionID: C58 Channel: PJSIP/1003-00000003 Variable: CONFBRIDGE(bridge,sound_join) Value: en/confbridge-join Does

Formerly on a thread called [asterisk-dev] Where to find the code of application Bridge On Wed, May 6, 2009 at 7:38 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote: >> Can someone please tell me in which file the code for the application to >> be found? I was not able to find a file named app_bridge.c in the folder >> apps. > > app_bridge.c ? app_confbridge.c ?

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

Asterisk Project Security Advisory - AST-2014-014 Product Asterisk Summary High call load may result in hung channels in ConfBridge. Nature of Advisory Denial of Service Susceptibility Remote

Asterisk Project Security Advisory - AST-2014-014 Product Asterisk Summary High call load may result in hung channels in ConfBridge. Nature of Advisory Denial of Service Susceptibility Remote