similar to: meetme option 'e'

Hi all, Is there a way to connect two live calls through the manager api without directing them to a meeting room? Currently, I can connect them by sending them to a meeting room. However, I don't know what the overhead is, and I kind of think that if I can connect them or link them up, the overhead would be minimum. -------------- next part -------------- An HTML attachment was scrubbed...

I have noticed that when I try to connect multiple G729 VoIP devices into a MeetMe conference that I can only add up to the number of G729 licenses I have. Now I would think that because all the devices are G729, this wouldn't be the case and the only license that would ever be used would be if a non G729 device or Zap channel was a part of the Meetme conference. This is apparently note the

Hi all, Anyone know how to dynamically create meetme conference rooms on the flight? I remembered a while ago there was a switch that tell meetme to create the conference room is the room is not defined in the meetme.conf. It doen't seem to be working for me anymore. Thnx

I'm really confused on how to use these two options together: A while back: JustRumours edited this page: http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe and added a little section about dynamic conferences. the 'e' option is repeated all over the page as the savior of dynamic conferences, maybe I'm just dumb, but can someone tell me if a conference is created with the e

It sounds like Naren and company has their own CRM application. They need a predictive dialer that allows third party app integration. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Matt Florell Sent: Monday, March 13, 2006 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

My TE411p does not seem to like to have two PRIs from different telcos (span 1 and span 2). I can get one working, but not the other. However, if I use vpmsupport=0 when loading the wct4xxp module, they both work. But here is the problem, vpmsupport=0 disables the on board echo cancellation. Any ideas? BTW, here is zaptel.conf span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs bchan=1-23 dchan=24

I want to have a "personal meetme conference", so when on a call I can transfer the other party to my personal conference with "#7". (I can then make other calls, and dump them into the conference using "#7" as well, then join myself by dialing "7"). Using: exten => 7,1,MeetMe(${CALLERIDNUM}|Mpd) this works as long as I originate the call. However,

NVWhisper. Justin ------------------------------ Date: Thu, 08 Mar 2007 16:25:28 -0500 From: Wai Wu <wkwu@calltrol.com> Subject: [asterisk-users] Coaching in asterisk Is there a way to setup a conference where party A can coach another Party B, at the same time, all other parties cannot hear party A? In order words, partis A and B can hear every one, and party A can only be heard by

Hello, I'm running Asterisk 1.8.10 on Linux box, when I'm in a conference(meetme) with another person, and a third person join our conference when the third person leave the conference I get disconnected from the original conference with a second party. I hope this clear. This does not happen often, is random, anybody experience something similar? or any idea how to fix this problem?

Hello! I am having problems figuring out how to do something, and any help would be much appreciated. I would like to use the manager API to take an existing call on a specific SIP extension, dial and conference in a third party. From what I can tell, the way to do this would be to take the two original parties on the call and stick them in a meetme room using Redirect with ExtraChannel,

Hi, I am playing with conferencing, but might have hit a bug... Any use who wants to hang up or leave the conference should press the # key, after which they get a "goodbye" message and the call gets disconnected. However, this does not happen. whatever keys are pressed by whichever party gets heard on every other party. I am using Zap channels (Digium T405p) My extensions.conf

We've started to use Asterisk for conferencing and have been getting some complaints. Our configuration is that some people call in from home, but we have a physical conference room with a Polycom. When somebody was giving a presentation in the physical conference room, we were told that the remote people kept hearing him cut in and our. To me, this sounds like the talking optimization was

Anyone tried the digium TC400B transcoding card? What are your opinions? Thnx

hi,all i install MeetMe module on Asterisk 1.6.2.10. when i use MeetMe to open a conference. even without 'r' option .it will record too. is this the bug of this module? my dialplan is : [95040] exten => 95040263007,1,MeetMe(95040,sM,123) the CLI output is : *CLI> == Using SIP RTP CoS mark 5 -- Executing [95040263007 at 95040:1] MeetMe("SIP/999-00000021",

I'm running 2.4.5 and app_meetme never plays conf-hasleft or conf-hasjoined with user names. I looked at app_meetme.c, but couldn't determine the cause. Any suggestions are greatly appreciated. exten => 600,1,MeetMe(600|i) I get the following: -- Executing MeetMe("SIP/jon-21f8", "600|aciMps") in new stack == Parsing '/etc/asterisk/meetme.conf': Found

Hello, Asterisk version 11.13.1 I'm trying use realtime meetme application with c and q option. If both options are used together in meetme table under 'opts' field, c option (Announce user(s) count on joining a conference.) doesn't work i.e. system doesn't play user counting to caller. Is it bug or some configuration problem. Thanks, Kamlesh --------------

Hi all. I'm trying to limit the number of participants in a conference room with the realtime option "maxusers", but it doesn't work. Someone have this option working properly? -- thiagoc "O povo n?o deveria temer o governo. O governo ? quem deveria temer o povo." V de Vingan?a

Hello, I know there's a problem with Asterisk at the moment in that while it's easy for one caller to dial another (using the dial command), it's tricky to connect two calls that are already in progress. I've been using MeetMe to achieve this (with each caller's call being directed to a dynamically created conference room programatically), and this is working - kind of -

Hi all, I am handed a project to setup *. The requirement is that it can handle 8 T1s. Half of the calls coming into the system will be routed to SIP extensions (with transcoding). The machine we have in our disposal is a new dual Xeon 3.2gHz server with 2g of ram and an dual 1000mb nic. Voice will be coming in from the PSTN (through 2 quad digium cards) in g711ulaw, and most of the time will

Hi all, I am setting up a script to record all the call. There are two app for recording. "Monitor" and "Mixmonitor", one mixing the audio on the fly and one mixing it at the end but also allow a option not to mixing the audio at all. If mixing the audio on the fly is not that taxing on the CPU, I would like to use 'mixmonitor' app. My question is, what is penalty on