similar to: MeetMe VS. 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

hi, guys, when i create a manager account used for freepbx, the follow info produce all the time? do you know that's the reason? == Manager 'bitzsk' logged off from 127.0.0.1 == Manager 'bitzsk' logged on from 127.0.0.1 == Manager 'bitzsk' logged off from 127.0.0.1 == Manager 'bitzsk' logged on from 127.0.0.1 == Manager 'bitzsk' logged

hi, All one thing confused me a long time. when i change the extensions.conf file. why not take effects after restart the asterisk? details as follow: my dailplan is : [95040] exten => _95040XXXXX,1,Set(CALLINNUM=${CALLERID(dnid)}) exten => _95040XXXXX,n(start),Answer exten => _95040XXXXX,n(welcome),Background(${welcomefile},,123) ... exten => i,1,Playback(invalid) exten =>

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 ?

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, list i want to know what is the best OS for install Asterisk 1.6.2.9, which should work properly on working system. i want to use CentOS5.2 or CentOS 5.4. Which is better and stable? Thanks for your help. -- Thanks for your supporting, have a nice day. Sucan

hi, all i want to use g729 codec for set up a call. so i donwloaded the so file from web site: http://asterisk.hosting.lv/#bin and install it properly. *CLI> *CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin

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

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,

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 , all when i do the command "module load cdr_radius.so" ,error happens. i have installed radiusclient-ng , what's wrong with it? thanks! error message as follow: ZHANGSHUKUN*CLI> module load cdr_radius.so Unable to load module cdr_radius.so Command 'module load cdr_radius.so' failed. [Dec 23 17:55:41] WARNING[31072]: loader.c:380 load_dynamic_module: Error

All; What I want to do is create a way to easily send callers into a conference room to have an N-way conference call. I created an extension '100' that calls the MeetMe() command. Then all I need to do is transfer a caller using a blind transfer (*2 in my case) to extension 100. Then I can dial a feature code that sends me into that conference (*15 in my case). So far, a piece of

hello,all there is a AudioCodes Mediant 2000 out there. i want to realise ip to PSTN and PSTN to ip connection. after some configuration on AudioCodes Mediant 2000, PSTN to ip connecttion works. next ,i want to dial from asterisk to PSTN now. i have see the sample in the extensions.conf relevent to PSTN as follow: ; If you are freely delivering calls to the PSTN, list them here ; ;exten =>

hi,list Could Asterisk-addson-1.6.2.0 install in 64bit Cent-OS ? after i make and make install. i cant find the .so file. is this mean it can't install on 64bit Cent-OS. ps: it works fine on the 32 bit Cent-OS Thanks very much! -- Thanks for your supporting, have a nice day. Sucan

hi, Does A2Billing has mial list? -- Thanks, Sucan

Hi,all i want to hear the voicemail recorded, but when hear "if you want to play message , press 3", after i press 3 i only hear that that's the time the file recorded. not the content. do you know how to hear content of voicemail fle? debug message: == Parsing '/var/spool/asterisk/voicemail/95040654321/3/INBOX/msg0001.txt': Found -- <SIP/1003-00000058>

Dear all, I can't understand the diff between roundrobin and rrmemory strategy. Could you explain for me ? and is roundrobin means each available interface ring once or several times and ring another? ; A strategy may be specified. Valid strategies include: ; ; ringall - ring all available channels until one answers (default) ; roundrobin - take turns ringing each available interface ;

I'm trying to determine the capacity of my host running Asterisk 11.8.1 on CentOS 6.5. The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet. The primary application will be bridging groups of users using meetme(). I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1 behaving a bit more like a production box -- bridging calls (box2). The call

Hi, I know this question may have been asked before (although the archives don't seem to suggest it), but has anyone had any problems with Asterisk accepting a PIN number for a conference room. At this point in time I have established the conference definition in the meetme.conf file as well as specifying the appropriate lines in the extensions.conf file. meetme.conf file: conf =>

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