similar to: Conference Meetme

Hi Guys, I am looking to use System() function along with some bash scripting to determine if a Trunk is being used during certain time of the day or not. Here is what I have in mind. Please guide me if you know a better way: exten => s,1,answer exten => s,n,System(/tmp/check.sh) check.sh: check EPOCH time => do an IF for certain times => Allow mutiple calls in certain times and

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

I am playing around with dynamic meetme conferences, and wanted to have one person constantly in the conference, with calls "popping in and out". Is there an option / any way of playing enter / leave sounds to the person who created the conference only, and not the people leaving / joining ? TIA Julian.

Hello, We would like to have a conference with 15 users aprox. We think that Pentium 4 3GHz and 1GB of RAM should be enough. Only Asterisk running. We wonder if somebody has some other experience, good or bad. We will use Asterisk 1.2 (it is a small and short project for only this). Thanks! -- Carles Pina i Estany GPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona

In the past, there have been reports of problems with Asterisk with multiple processors and/or HyperThreading. I'm having a !@#$ of a problem with an HPDL380 with 2 3.4gHz Xeon processors, 2 gb RAM -- if I got 24 hours I'd think I had died and gone to heaven :) Am I missing something obvious like "Asterisk is single CPU, single core?" I can't access the ILO so I

Hey Guy! I have following dialplan for meetme and i want if someone type wrong meetme extension it should say invalid extension. But look like following doesn't work. its just hangup if i type wrong number. how to fix this code.. ;Conference rooms/lines: exten => 7580,1,Goto(ivr-meetme,s,1) [ivr-meetme] include => meetme exten => s,1,Answer() exten => s,n,Wait(1) exten =>

hi, I am trying to get the usernum of a user when dialing in to a MeetMe conference. Is there somehow a possibility to save the usernum of a MeetMe participant into a variable? Everything should be done through the DialPlan, no manager and no *cli. Thanks for your help, Emrah

Hey, i forgot to ask, how can i get the user number from a caller he is in a conference, i don't find a variable to us this for the current channel. Only the command "meetme list <roomnr>" shows the usernumber, but i can't use this output. Thanks. Daniel

I will be using a dell R320 Xeon E5-2420 2G and 4G RAM. also using a SIP trunk with ulaw/alaw codec. How many calls could I expect to make at the same time? no transcoding or anything. Just call a number and play a gsm file. Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

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,

The meetmeadmin() dialplan function lets you specify a user to mute, un-mute or kick. But how do you get a list of users in your dialplan? When a user joins a conference, the user number assigned is "the last user number +1." If you have a long running conference with callers joining and leaving all the time, this can grow to be a large number. I want to be able to

I want to be able to allow a caller to dial a ddi, system to verify identity etc (this is all done) I then want them to sit listening to music, until an event happens. When this (external) event happens, I want to play a certain file to the caller, using playback (so that they have ff / rw etc), and when finished, go back to the music. 1) I thought of redirecting to an extension that played the

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

We're running Asterisk 1.8 on a 32-bit Debian machine, and it has been fine for some time now. But! We've got such a incoming call volume over the few weeks that we'll have Asterisk occasionally restart itself. My hunch is that it is in part memory pressure. I can't add RAM and have it help, because it's 32-bit. I intend to move to a 64-bit machine, but I was hoping to

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

We have two Linksys 942 phones which sound great when they call each other directly through Asterisk. But when they both dial in to a meetme conference room, the sound is very jittery. Other phones like Polycom 501 and Snom 360 sound fine when using meetme. Both Linksys phones are set to use the default g711u (ulaw) codecs. Adjusting the jitter buffer and jitter level settings to various values

Please don't top post. > Am 29. Mai 2015 09:42:55 MESZ, schrieb Luca Bertoncello > <lucabert at lucabert.de>: >> Zitat von jg <webaccounts173 at jgoettgens.de>: >>> Yes, it is called "core set verbose 42", the other options is "core >>> set debug 42". Enjoy the show! I know you can specify a level to the verbose application,

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

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

Greetings, I have 4 asterisk boxes in 4 different locations, works perfect [over ip with ipsec + asteriskes use IAX to talk to each other]. Users can call each other and talk no problem, sound quality is fine. asterisk is great. I am to create the conferencing facility. and I am stuck. The TDM400P card is installed in one of the asterisk boxes [zaptel, wcfxs and crc_ccitt modules installed