similar to: Meetme conferencing - large deployment SIP or ZAP?

I was able to get it to work with 2 extensions. One for the "host" and one for the "participants" Not the best way to set it up but it works. Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Savoy, Kevin - Williston, ND Sent: Wednesday, December 13, 2006 8:06 AM To: Asterisk Users

I'm currently writing some code to support conferencing in Asterisk without using the Meetme application. The conference runs in its own thread and every new inbound or outbound channel that is created is passed to it. This thread runs the conference loop reading and writing frames to each channel. I'm writing this as if it were a bridge with more than two channels, and I'm not

I am trying to set up a Conference room where users are put on hold until the host arrives. I have figured out that the A option activates marked mode and the w option is used to activate the waiting until the marked user arrives. This seems to be what I need. What I can't seem to find is how do I mark a user? Thanks _____________________ Kevin Savoy Business Unit Telecom Analyst 2218 4th

We are developing an on-demand teleconferencing solution. We will be billing per-minute/per-user. I've successfully gotten Asterisk to write CDR data to a postgres database, but with the way I've got things setup right now the CDR does not have the dialed conference number. We need this information in order to be able to bill. As teleconferencing is the only application of the

Hi all, I'm trying to get the MeetMe system to take a caller and announce to them they've joined the conference in addition to the other members of the conference assuming previous members of the conference >= 1. I can see where the meetme.c app actually processes it using the ast_pthread_create_background(&conf->announcethread, NULL, announce_thread, conf); function. The

I have a site running 1.4.17 with Zaptel. They want to add a TE420P for additional T1 capacity. I'm 99% sure this will work, anyone aware of a reason it wont? Thanks, James -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100806/b5f4bc0f/attachment.htm

Hi- Since the tru64 patch was designed for -current, I thought I would try to build it with a recent snapshot before backporting to 3.4p1. So I downloaded openssh-SNAP-20020826.tar.gz frpm the portable snapshots, but it does not contain the 'configure' script. I tried copying the 'configure' from 3.4p1, but that does not create a Makefile from the Makefile.in. Where are the

In my R learning I've come across a situation in which a piece of code that works on the work space outside a function does not work inside the function. WARNING THIS EMAIL CONTAINES THE CODE:#rm(list=ls()) THIS WILL CLEAR ALL OBJECTS FROM YOUR WORKSPACE! When I use rm(list=ls()) and then ls() it shows character(0) So I tried to make a quick function to speed this up as follows:

Hello R-user, sorry for this very off-topic question. But I shall present R to my dept. (pro's and con's and what it can do). The pro's and con's are easy but not what R can do (additional to the "normal" statistics). I looked through the packages, but the enormous amount of packages makes it very difficult for me to decide which one is worth mentioning. I used

Hi, I just read a pressrelease from VON that Digium will soon be releaseing a couple of new cards. What got me interested was: "The TE420P and TE415P support 128ms of G.168 (2002)-compliant echo cancellation across their entire 128 channels." Does anyone know when thease will be released and what they will cost when released? Thanks!

I was playing around with conferencing tonight. I was able to place a bunch of SIP phones and a couple of my Zap FXS phones into a conference. So I thought, "Let's see what it's like when people come in from outside." So I called a friend and had him call in on one of my Zap channels, WHICH IS CONNECTED TO MY POTS LINE THAT DOESN'T DO DISCONNECT SUPERVISION. When he

Dear R user, I asked for packages which should be mentioned in a ecological related presentation of R and would like to summarize the replies. First of all E. Paradis pointed out, that he would prefer to see R presented in a different way, please read his whole reply https:// www.stat.math.ethz.ch/pipermail/r-help/2003-December/041951.html. Packages which

Hi All I am trying to setup : PSTN E1 ---> Asterisk------>Legacy PBX------->Legacy Analog extensions. I've followed steps using : http://www.voipinfo.org/wiki/view/Asterisk-Panasonic i get the green light (sync) on both the 2nd span of digium TE420P (that is cnnected to the legacy pbx pri card) and the pri card of the legacy pbx. but when i try to make a call to asterisk so that

I'm putting together a PBX using a TE420P card configured for E1s that is connected to an Errickson MTS. successfully compiled and installed libpri 1.4.11.3, DAHDI 2.3.0.1+2.3.0 and Asterisk 1.6.2.9. everythings seems to be working. SIP phone to SIP phone (POLYCOM) calls work fine however, network calls do not. When I went to debug PRI, the only command that showed up when I did ">CLI

Hello I am trying to configure TE420P but i am confused what to give chan_dahdi :( Below is configuration i am using for TDM400P Please help what changes to make in it... Please provide a link as well [trunkgroups] [channels] ;default for channels switchtype=national rxwink=300 hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes

Hello Everybody I am using Trixbox 2.4 (with TE420P & PRI lines) .. my setup is like Calls -->Asterisk-->legacy pbx--->analog extensions(agents). Whenver a call comes in , asterisk dials the ACD number of the legacy pbx which in turn decides to route to appropriate agent.. for ex : s,1,Dial(ZAP/g4/5432) [g4 is the 4th span and 5432 is the ACD number of legacy pbx under which agents

Given a choice, and a green-field site, would you a) Have a separate network (switches etc) for your data and phone b) Use the same network, but use VLAN's ?? What are the pro's and con's of each ? TIA Julian

Can anyone point me to a sample or information on using MeetMe like this? Conference room is set up with 2 PINs, one for the moderator and one for the participants. Participants get music until the moderator joins (to avoid wild, un-moderated tangents). Call is ended and all participants are kicked out when the moderator leaves (or the moderator can kick everyone out via phone keypad).

HelloI have the following system Asterisk 1.6.1dahdi 2.2.0.2 TE420P card Centos I have noticed that all the four lights are blinking(ie coming red and then off so on)... Previously I also noted that when dahdi drivers are not installed lights blink but one by one in sequence(like in marriage cermonies :P) and after dahdi installation lights get off ... but this time all at same time

Hi all, Could someone please tell me what is the relative "cost" in using conf files oppose to the astdb? Basically I need to match a name to a phone number in order to have all users registered by name and not by number (which I understood is not a good practice). I have 2000 users and a complex dial-plan and server resources become an issue. I could implement this via a context in my