similar to: ConfBridge details

Howdy, I am proud to announce that after a good bit of development, community feedback, testing, and code review, the brand new ConfBridge application has been officially merged into Asterisk Trunk!!! http://svnview.digium.com/svn/asterisk?view=revision&revision=314598 If you are already familiar with ConfBridge from Asterisk 1.6.X and 1.8, forget everything you know. This is a completely

Address 0xfffffffe out of bounds why and how to solve.MyConfbridgeCount(conferencenumber,variablename )return total number of user in conference given by conferencenumber otherwise zero.At runtime using MyConfbridgeCount(4000,count ).now app2: MyConfbridgeCount will call function count_exec(struct ast_channel *chan, const char *data).But at compile time char * data cause core dumped.

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

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

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.

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

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 ?

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 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

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

Hi guys, I'm trying to move away from meetme to loose the dependency on dahdi. ConfBridge seems to be a good fit but I can't get it going. The document sounds like an easy to use app. Am I missing any bridge_ modules? Asterisk 1.6.2.0~rc2-0ubuntu1.2 -- Executing [521 at outbound:1] Answer("SIP/109-b877a8c8", "") in new stack -- Executing [521 at

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've got some Asterisk 11 (I could 'upgrade' if needed) hosts using meetme and I'd like to switch to confbridge to service more callers. Can anyone reply with their experience along the lines of 'using meetme I was only getting x callers per server but with confbridge I now get y callers per server?' -- Thanks in advance,

I have a very old server that is used only for conferences on Meetme. To manage the conference rooms we use Web Meetme. Now it is time to upgrade everything but since Meetme is no longer available I need to find a replacement GUI to manage the conference rooms. Anyone know a solution that works with Confbridge? I found "Asterisk Confbridge Manager" from a russian company but it

On 11/14/17 3:55 PM, Joshua Colp wrote: > On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: >> I followed the blog post and I can get video from the conference if >> I configure the bridge as follow_talker so I know everything is working >> on the pjsip side. The only problem is that video_mode = sfu is >> apparently not valid in either confbridge.conf or

Hi How to set a threeway conference in asterisk only for VOIP (I am using only SIP channel). Thanks Nikhil

I believe I am running an AGI (to put users in a conf) before the confbridge is built. So the users are not really get in the conf... exten X,1,run agi to put users in conf exten X,n,ConfBridge() How do I have in the dial plan ConfBridge() and someplace run an AGI that brings the users I want into that Conf. I cannot delay in the AGI and wait for the conf because the conf is not built until I

We're encountering no audio issues in ConfBridge. Only the moderator and the 1st invited have audio. When the 2nd invited number picks up the phone, only the announcement is heard, then followed by silence. This occurs frequently. Only 1 out of 5 test calls do not have this problem. We were hoping that switching from MeetMe to ConfBridge will solve the no audio issue, since there's no

On 11/14/17 4:27 PM, Joshua Colp wrote: > On Tue, Nov 14, 2017, at 06:25 PM, Carlos Chavez wrote: >> On 11/14/17 3:55 PM, Joshua Colp wrote: >> >>> On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: >>>> I followed the blog post and I can get video from the conference if >>>> I configure the bridge as follow_talker so I know everything

Hello, using Asterisk 1.8.12.2 I am having trouble with exiting the conference room by entering a single digit. option X of the Meetme()-application should do this. I have following in extensions.conf : /exten => _1000X,n,Set(MEETME_EXIT_CONTEXT=dynamic-nway-invite)// //exten => _1000X,n,MeetMe(${CONFNO},dMX)// // // //[dynamic-nway-invite]// //exten => 0,1,NoOp(confno =