similar to: meetme feature request (or maybe its there already?)

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. I am having a strange problem when using the 1.4 version of asterisk and zaptel. If I call from a pstn line into the asterisk box using a phone number which calls the box via sip, then once I am in the meetme conference nothing happens when I hit the star key -- I cannot get the user menu. There is nothing in the logs at all its as though asterisk never sees the digit at all. Now if I do

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

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 am using latest asterisk 1.2 and it would be nice in a meetme conference to be able to mute all but a particular user and then unmute all those users again with one command. Am I missing something or is this not available? Maybe I could write something, but I wanted to check first. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it?

Hi. After the latest upgrade of asterisk-1.8, I notice that meetme does not allow the menu, if musiconhold is active which only occurrs if a single user is in the conference. Sometimes I have to unmute someone because for various reasons they cannot do this and they are the first one in the conference. I looked at the code and its a one line change, and I wonder if it was done deliberately or

I am using CVS as of December 7 2005, and I am having the following problem -- I have an extension so I can transfer a call I have made (from a zap channel) to a meetme conferenceand what I have found is that if I am using the i option, I can transfer the call OK, but the zap channel I transferred from gets totally unusable --what I get in the log file is endless messages like this: Feb 22

Hi. I tried to record a meetme conference using the r option -- using asterisk 1.2.4 and the volume is so loud it clips. Any way to fix this -- using the monitor the volume is generally fine. Thanks. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici@ccs.covici.com

Hi. I have an extension to use ControlPlayback -- however the forward and rewind controls do not work if I am in a 3-way call with the pstn and that extension, however if I then hang up the controls work. Anything to do about this? Any assistance would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici

How did you install these packages -- make sure you do ./configure and if needed make menuselect in each one of these before the make and make install. This is the only thing I can think of -- check whether there are any built-in modules as well. on Monday 03/12/2007 Asterisk Asterisk(asteriskbunnies@yahoo.com) wrote > Hey! > > Thanks for your interest, i checked the modules and i

Oh, I forgot to mention that Asterisk 15 went End-Of-Life last Thursday. :) You should use Asterisk 16. On Mon, Oct 7, 2019 at 5:58 AM George Joseph <gjoseph at digium.com> wrote: > > > On Fri, Oct 4, 2019 at 1:19 PM John Covici <covici at ccs.covici.com> wrote: > >> Hi. I am trying to install asterisk 15.7.4 from git onto a Debian 10 >> system and I am

Hello. I have a strange problem with the MeetMe application. Configured is a misdn msn to go into a preconfigured MeetMe room. exten => 12,1,MeetMe(1234,pIM) The first caller gets the prompt to enter the pin and then gets connected to the MeetMe room. The second caller gets also the prompt but after entering the right key he hears a dialtone followed by the message: The number you have

Hi. I have an asterisk system under Debian Leni using asterisk 1.8 with no Digium hardware -- and when I go into a meetme conference the system either locks up or is 100% cpu utilized or something -- I can't type anything and I have to physically reboot the system. The dahdi module is loaded and the last log entry is the playing of you are the only person in this conference,. How would I

Hi. I am using asterisk-1.8 and I am having problems getting conferencing to work properly. I did modprobe on dahdi and did load => chan_dahdi.so in /etc/asterisk/modules.conf. Now I do get conferencing, but meetme says [Jan 15 11:38:56] WARNING[9214] app_meetme.c: No DAHDI channel available for conference, conference recording disabled (is chan_dahdi loaded?) Now chan_dahdi is indeed

Hi. I have a Debian Leni system with asterisk-1.8. I was trying to get meetme to work and it depends on dahdi, so I compiled dahdi-trunk and dahdi-tools-trunk, however, when trying to insert dahdi_dummy, it complained about symbol crc_ccitt_table, although the module was actually there in the kernel tree. So, I took the Debian source, and I had the config and I did make Bzimage, make modules

One thing that concerns me is that this post is from 2009, even though the newest version of the app on Sourceforge is up to date. I have a customer who has been using a conference server that I built for him using app_konference for several years now and he routinely runs conferences with anywhere from 10 ? 125 active users. The ultimate goal is several hundred concurrent users and I can see that

Hi, I've just observed something a bit odd - I'm wondering if this is the expected behaviour, a bug/feature, or something I'm doing stupid! 1st person gets into MeetMe. Nothing fancy, just: exten => 987,1,MeetMe(400,iM) They enter the passcode and their name, then listen to MoH. So-far so good. Now the 2nd person dials in. They enter the pin-code, and at that point, the MoH

Emrah wrote: >> This is an asterisk-users question, and would have been more appropriate to have >> asked there. >> >> Instead of setting up your conferences in meetme.conf, you could set them up >> dynamically in the dialplan, and then you can control whether the user is >> prompted for a pin or not when joining the conference, based on whatever logic

First off, not a dev question, redirecting to -user. On Sun, 19 Jul 2009, Emrah wrote: > Would it be possible to imagine an option that avoids the need of > entering the pin code of a conference room? Unless I'm misunderstanding you, just create the meetme without requiring a pin. See: [core] show application meetme or whatever random order of words makes sense to your version

There is one more thing to try: http://snapvoip.blogspot.com/2009/07/appkonference-asterikast-high.html I would appreciate if anyone can comment on the feasibility of playing an audio file to the caller and callee using ControlPlayBack and appkonference. Much of the reviews indicate that appkonference is an over-kill for an audio as its main functionality is with video. Going past that. Thanks