similar to: Asterisk Meetme & MeetMeAdmin cmd info-use

Hello users, I have been working with a conference setup. My setup includes: 1)There will be an interface number provided to the user which might be a DID number or A Toll free number When user calls the number it asks for the conference room number and the user pin . on successfull authentication he will be participated in the conference 2)by didaling the same DID number the

Would anybody mind if the the following command options where added to MeetMeAdmin? 0 - 9, * and # I'm considering hacking the code to add these commands to play the DTMFs to the specified user as tones and hope that the SIP or IAX channels then work with these correctly. -HJC

hi! i'm on debian lenny and trying to run ocfs2 on a dual primary drbd device. the drbd device is already set up as msDRBD0. to get dlm_controld.pcmk i installed it from source (from cluster-suite-3.0.10) now i configured a resource "resDLM" with 2 clones: primitive resDLM ocf:pacemaker:controld op monitor interval="120s" clone cloneDLM resDLM meta

If I start ocfs2_controld.cman in parallel on a few nodes, only one of them starts up, the others exit with one of these errors: call_section_read at 370: Reading from section "daemon_protocol" on checkpoint "ocfs2:controld" (try 1) call_section_read at 387: Checkpoint "ocfs2:controld" does not have a section named "daemon_protocol" call_section_read at

built-in; This doesn=92t matter because the moderator would have to use meetmeadmin or the confbridge equivalent to control the other functions. The moderator would either need two phones or a phone and a web = interface. Let=92s say Yves=92 =93special conference=94 is 5555. The moderator = would start using this command Exten =3D> s,1,meetme(5555) The participants would do Exten =3D>

http://bugzilla.mindrot.org/show_bug.cgi?id=273 ------- Additional Comments From sam at neutrino.phy.uct.ac.za 2002-07-13 01:22 ------- i also have this exact same problem using OpenSSH_3.4p1, SSH protocols 1.5/2.0, OpenSSL 0x0090603f i start a program with "nohup <program> &" and upon returning to my native machine, the terminal is balnk, frozen with no bash prompt. i

hi, we are experiecing the following probem, if anybody have come across such a problem or a solution to this please let us know. our set up is, an Asterisk server equipped with, 4 port station interface card ,single port fxo card and several soft sip phones we have found problems with the following scenarios, outside caller (calling through fxo interface) <------------------------------>

Hi, I'm new to asterisk and have a little trouble in developing my first more complex dialplan. The basic task is a click to call solution: - call one number via sip, play some announcements, do cdr etc. and put the callee into an conference room with music on hold - call a second number via sip, play some announcements, do cdr etc. put the callee into the same conference - have a nice chat

Is there any way that I can have an admin user hit * and then Mute all other users in a meetme conference? Sort of a moderator function? I know it can be done with MeetMeAdmin, but as I see it that requires a separate extension to dial, unless I've got the logic wrong? If it can be done in a single extension please show examples. Thanks. ________________________________ This e-mail,

Under Advanced make sure this is set: Call join on Xfer (2 calls): to off ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tommaso Calosi Sent: Thursday, May 04, 2006 4:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Unwanted conference with snom320 and asterisk

These is my scenario. Asterisk 1.4.16 Zaptel 1.4.8 Grandstream BT200 Grandstream GXP2020 Grandstream GXP2000 For some reason the end user ask to configurate son direct access like *01,*02,*03 thru *78. After they began to use these direct access, I cant place a 3 way CONFERENCE. I remove the direct access, but i dont know if one of them block the CONFERNCE. Do you know if i can make

I have 13 Snom 320 with asterisk 1.07 bristuffed. The problem is that sometimes on random basis, when one customer is placed on hold and another call arrives, the customers are put in conference with each other. This look very strange to me, but I've disabled the confernce button on the snom phones to prevent the human errors, but it still occurs. Investigating I've discovered that a

OK, I got my Cisco 7960's to auto-answer on the second line but I can't get the Asterisk to call all the lines at one time. I have 4 phones I would like all of then to answer when I dial x300. Any help would be great Thanks Tuska extensions.conf [conference] exten => 300,1,AGI(callall) exten => 300,2,MeetMe(300,dtqp) ; press # to exit the conference exten =>

Hello Folks, I am testing Asterisk 1.4.2 running on Fedora Core 5 (version 2.6-17). I have loaded the app_meet.so module in order to activate the MeetMe, MeetMeCount and MeetMeAdmin applications. While I have been successful in loading the app_meet.so module, I am experiencing an immediate kernel panic every time I try to make a call to a room conference. Is this story unique to me? How can

Has anyone here done any work on clustering Meetme conferences over multiple Asterisk boxes? The scenario I am thinking of is where there are two or more boxes connected to a set of PRIs that all answer to the same PSTN number, and where it's not possible to know in advance on which box a call would arrive. So it would be possible to have some calls on one box and some on another, that should

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

Hi people, I want to make a meetme between 2 numbers. First I enter the number1 into the meetme. It is waiting for the other number, but the other number never entered, so, how can I finish the meetme from the dialplan?. Is it posible by using MeetmeAdmin and kick all the users? Thanks, Anahi Ludue?a _________________________________________________________________ Descubre

hello i want to kick participant in a meeting by pressing the digit on sip phone.when i entry the meeting ,no matter how i press the button,the dtmf does not work. here is my dialplan and my agi script,and sip.conf [from-internal] exten =>121,1,MeetMeCount(900,CONFCOUNT) exten =>121,2,GotoIF($[${CONFCOUNT}<10]?3:100) exten =>121,3,Authenticate(123456) exten

Hello fellow asterisk users, I've been facing a problem when using MeetMe's admin functionality to unmute users in a conference using *Asterisk 1.6.2.11*. I've tried: 1) MeetMeUnmute (AMI) 2) MeetMeAdmin(AMI) 3) MeetMeChannelAdmin(AMI) and also tried via console : "asterisk -rx 'meetme unmute conf_no user_no'" and the available AGI functions. but all of this to no

Hi All, I am in desperate need of this feature. I want to play background music during a call while the 2 parties are having some lovely conversation (or maybe give them a sort of cursing background if they are cursing each other). I found this post which talks about creating a ghost call with the help of queues and putting that queue in a meetme room where queue will play the song/curse and the