similar to: no audio from meetme conference bridge

Hello folks, I'm running 1.8.11 on a Centos 6 system with an adjacent Hylafax server using softmodems: Noticed this in the Asterisk log when trying to send a fax from Hylafax to Asterisk: [Apr 3 01:53:09] WARNING[29184]: chan_sip.c:8926 process_sdp: Multiple audio streams are not supported I've googled a few asterisk tickets that may suggest that yes, multiple audio streams are not

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

Hi, Is it possible to change the default voice prompt for Asterisk meet me conference bridge. We have our own customized recordings for Welcome and PIN request and would like to use that instead of the default "Please enter your.. " . If I replace the default sound file with my custom file by using the same filename as the default message, will it affect any other applications..??

Is it possible to play and audio file into a meetme conference for both parties to hear? I thought I remembered reading something about it, but I can't find it now. Any help would be greatly appreciated. Paul

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

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

So I setup the MeetMe application in Asterisk Assigned an extension to it. When one of my SIP phone dials the conference extension, they get a message "you are the first one in the conference", so far so good. When the 2nd SIP phone dials the conference extension, they get a busy signal Now I know that you have to have a Zapta device to enable conference application. I have an X100P (1

>>> 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 have the issue, since meetme is no longer being developed. Doug

>>> 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. Doug

>>> Can I adjust the talk or listen volume for another user? I've never used the volume controls, but it would appear. https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration Doug

I have Asterisk running on a Linux 2.4.x box with ztdummy. Once I did a modprobe on ztdummy I was able to enter into a conference room using my softphone clients. I'm using SJphone and Firefly. I have noticed a significant delay (1 to 3 seconds) while talking within the conference room. I have tried both clients, SIP and IAX protocols and various codecs. I have also tried it from different

I'm trying to use the password entered with Authenticate to create dynamic meetme conferences with the following dial plan: exten => _XXXXXXXXXX18467,1,Authenticate(/etc/asterisk/meetme.pw|a) exten => _XXXXXXXXXX18467,n,MeetMe(CDR(accountcode)) ; 281-8467 However CDR(accountcode) is always being set to 1022 no matter what password is used. The passwords are stored in a file so they can

I have meetme working with BLF on polycom phones however when meetme is not actually being used by anyone the 'status' of meetme becomes "idle". Which the Polycom phone sees and produces a clock symbol and FLASHING red LED. Are there any 'tricks' or work-arounds to change this status to something that does not blink the phone's LED making it look busy when meetme is

Hi all, Anyone know how to dynamically create meetme conference rooms on the flight? I remembered a while ago there was a switch that tell meetme to create the conference room is the room is not defined in the meetme.conf. It doen't seem to be working for me anymore. Thnx

Hi, im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running meetme conference, when i try to call meetme i get this from the asterisk console Aug 24 00:14:12 WARNING[15466]: pbx.c:1720 pbx_extension_helper: No application 'MeetMe' for extension (sample, 65000, 1) i recompiled my zaptel and asterisk, but the app_meetme file still didn't install, what am i missing

Hi, I was just wondering how would the application be where the Asterisk calls a number and that number joins the conference as soon as the call connects. There would be only one conference already defined in meetme.conf and there is one person already joined the conference. Currently MeetMe requires a person dialing into it and the joining the conference. How could this be done using MeetMe or

Hi - I found the "maxusers" defined in meetme.c, but I'm not sure how this value is set. Does anybody know if one can limit the number of users permitted in a meetme conference? I know there's MeetmeCount(), but I'd rather avoid the dialplan logic and just set maxusers instead. Thanks, Noah

Hello; We need to have admin page, so the administrator can create passwords to be used to join the meetme conferences and can determine the allowed time ..? Well, the admin interface can be done easy (I do not know if there is something ready), and the password and the time limitation can be added to the database (or even text file), but how asterisk can use it? Do I need to use the AGI to

Hi there, I'm using asterisk meetme function like: exten => 9070,n,MeetMe(|dcM) and everything works pretty well. But I would like to add a review of the entered conference number before the user jumps into the conference. Somthing like: *:"Please enter the conference number followed by the hash key" (works) U: 123456# (works) *: "You are entering conference number