similar to: Problem transferring call to a meetme conference

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. 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

Hello, Is there a way to change the format from the default? 'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format ${MEETME_RECORDINGFORMAT}). Default filename is meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. - requires chan_zap.so Many thanks ******************************************************************** This email and any attachments

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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

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

Hi. There are two things in a meetme conference which I would like to be able to do -- one is to start recording somewhere in the conference without having to change my dialplan to use the r option -- sort of like the one-touch in the features.conf -- and some way to play some specifically named file fromwithin the conference. I have an extension which uses ControlPlayback to play a file, but

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 am trying to set-up the Open SSH. I want to connect to my Unix systems from Windows or from any other Unix system by using DSA public and private key without password. With this I can run the scripts on the remove system and use the scp etc on remote system. Can you please help me to configure this system. I don't know how to set-up this. I am getting the following error

Hi. I am using asterisk 1.4 (latest as of today) and zaptel 1.4 (latest as of today) and I cannot dial out using my 400P card with one fxs module and one fxo module. I am using kernel 2.6.24 and get the following log entries: [Feb 25 17:28:13] VERBOSE[25071] logger.c: -- Executing [s at macro-dialout-trunk:23] Dial("Zap/1-1", "ZAP/4/www411|300|wW") in new stack [Feb 25

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

?? Hello.? We are very successfully using asterisk (in the form of trixbox 2.2/asterisk 1.2).? We run a few conference lines for customer teleconferences which mostly work well but they seem to operate at half duplex.? If a person starts talking they will cut off others on the call.? Is this normal behavior?? Are there any options I can change to change this? ?? Thanks! James -------------- next

Hi Guys, I need some help. I''ve been using shorewall for a while now and it''s been running beautifully, but I''m now experiencing some problems. It seems that connections are getting dropped much like the behavior described by the NEWNOTSYN=no option in the shorewall.conf file, but I have NEWNOTSYN=Yes in my file. The messages I see in my logs are things like:

Hi , I am new to this puppet. I am implementing a network where my cisco switch will contact the puppet server for getting the configuration. I tried installing open source puppet and was successful in pushing down the configurations. I wanted then to try the same exercise with puppet enterprise 3.1. I installed puppet enterprise in a different server and changed my puppet agent (switch) to

I would us Meetme for conferance SIP-->SIP fist. my Meetme.conf: [rooms] conf => 9999 my extensions.conf: exten => 9999,1,MeetMe(9999) But : == Parsing '/etc/asterisk/meetme.conf': Found Jun 16 10:33:22 WARNING[12100]: chan_zap.c:916 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Jun 16 10:33:22 ERROR[12100]: chan_zap.c:6969 chandup: Unable

Dear Help, I am having a very odd problem. For some reason, I am able to browse to my Samba share by IP address but not by host name from Windows 2008 servers in a particular domain (the same domain the Samba server belongs to). However, I am able to browse by host name from XP clients as well as Windows 2008 servers in a different domain. Even stranger, if I add an alias (using the netbios

New server, running 1.2.9.1 with a Sangoma 4 port T1 card, wanpipe rev 2.8.4. I get this message after using a MeetMe room a few times: Jun 27 18:37:23 WARNING[1054] chan_zap.c: Unable to open '/dev/zap/pseudo': No such file or directory Jun 27 18:37:23 ERROR[1054] chan_zap.c: Unable to dup channel: No such file or directory Jun 27 18:37:23 WARNING[1054] app_meetme.c: Unable to open

Hello subscribers, I have a very strange question regarding SSL setup on gluster storage. I have create a common CA and sign certificate for my gluster nodes, placed host certificate, key and common CA certificate into /etc/ssl/, create a file called secure-access into /var/lib/glusterd/ Then, I start glusterd on all nodes, system work fine, I see with peer status all of my nodes. No problem.