similar to: Leave and re-enter a conference

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

Title says it all - for the time being I am stuck deploying Asterisk in ESXi . We are also looking at Proxmox for our next round of servers.. Everything works fine except conference calls - very stuttery , have tried a few different codecs. I assume this is a granular clocking issue , and wondering if anyone has anything I could try to fix or mitigate the problem in ESXi environment . We have

On Wed, 25 Mar 2020 12:42:00 -0400, Doug Lytle wrote: > > > >>> he problem is that there is some sort of distortion in the audio > > Has been been going on for a while or is this a new setup? Do you have a timing source? > > I bit the bullet around a year ago and moved to CONFBRIDGE; it wasn't as horrible as I thought it would be to setup. Well, this has

On Thu, 26 Mar 2020 06:54:37 -0400, Doug Lytle wrote: > > >>> I never moved to confbridge because they don't have an option for controlling the volume of other > >>> participants audio > > I have menu options in my confbridge configs that has increase and decrease conference volume. > > I'd still configure a small confbridge and test if you still

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! As I see there is dsp_drop_silence switch in confbridge. Could you tell me how asterisk detects silence? Is it possible to change silence level, so, let's say some not loud enough background noises will be recognized as silence and only loud enough human voice will be recognized as sound? Thank you!

We have a customer who wants us to record anywhere from 2-4 participants on a call in stereo (as opposed to mono) quality audio. Some background.. We are using asterisk 16.6.1 We are also currently using AMI/AsyncAGI and ConfBridge to bring the parties together. I believe recording in the various file formats (based on extension), it's always recording in mono quality. My one thought is to

Hi list, I am in need of a "special" asterisk conference room with the following constraints: - there is one admin / moderator and several "normal" callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the

hi, group there are two module can used for meeting. MeetMe and Conference(which is a plugin) My question is : which is better for large conference(maybe above 100 people in a meeting)? -- Thanks & Regards Sucan

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

Hi folks, Just wondering if someone have already done something like that : SIP Client_A ---1)call---> ASTERISK ---2)outgoingcall-PSTN-->Client_B | | 3) Enter conference | MeetMe <----------------------------' with user A Make 2 user in conference, it's definitely easy, but call an other user and put the

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.

Hi. I have a problem with my audio in meetme conference under asterisk 13 using Debian buster compiled from source. The problem is that there is some sort of distortion in the audio -- a workaround is always to lower the listen volume (*4). I see nothing in the log and so I wonder what is happening. I have dahdi loaded so I can record the conferences. Thanks in advance for any suggestions and

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

On Tue, Apr 26, 2016 at 11:07 AM, Administrator TOOTAI <admin at tootai.net> wrote: > Le 26/04/2016 17:23, Mamadou NGOM a ?crit : > >> Hello, >> >> >> Having installed DAHDI to be able to use the meetme() application , when >> I start the dahdi service it generates me the following error: >> >> -bash: /etc/init.d/dahdi: No such file or

Hi all, Just wondering if someone have already done something like that : SIP Client_A ---> 1)call ---> ASTERISK ---> 2)outgoingcall-PSTN-->Client_B | | 3) Enter conference | MeetMe <----------------------------' with user A Make 2 user in conference (point 1 and 2), it's definitely easy, but call

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

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,

On Mon, Aug 26, 2019, at 2:00 PM, Dan Cropp wrote: > Thank you Joshua. > > Out of curiosity, what is the maximum capacity you have heard for > simultaneous ConfBridges in a single box? (Looking at 3-4 channels per > ConfBridge) with recording. I don't really remember any specific values. 100? 200? -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer

Hi all, I'm trying to get the MeetMe system to take a caller and announce to them they've joined the conference in addition to the other members of the conference assuming previous members of the conference >= 1. I can see where the meetme.c app actually processes it using the ast_pthread_create_background(&conf->announcethread, NULL, announce_thread, conf); function. The