similar to: MeetMe Volume Issues

Hi, Matt and Asterisk gurus I encountered the same problem in my asterisk meetme. Whenever the 3rd person joins the meeting, it creates echo in the meeting, while 2 person meeting is fine. I am wondering if you can give me more hint on how to configure the mixer to have echo cancelled. We are using analog phones connected to asterisk TDM cards. Thanks a lot. Michael

I sent a query about this before, but have some further information and am hoping somebody has a suggestion as to what to try next to debug this. I'm using an Asterisk box primarily for MeetMe conferencing. There are two sources: TDM via two Q.SIG T1's and SIP phones. Conferencing works fine between TDM channels. But when a SIP phone calls the conference, there's no voice path *to*

I have encountered a frustrating problem with the meetme rooms and calls entering the system on the Digium analog cards. The typical scenario is: Callers on SIP phones, X-lite, Eyebeam, Cisco 7960, IAXy Callers entering the system from the PSTN via the digium Analog card (TDM400P) In the meetme room the SIP connections can all hear each other loud and clear. The PSTN people can hear

Asterisk People, I'm currently using Asterisk and with a SIP voip provider and I'm having problems where DTMF input in my IVR app is getting corrupted intermittently. For example, if someone enters 1025, it may come though correctly as 1025, or it may come trough as 10025, or 100255. DTMF digits will just double up. This doesn't happen all the time. Asterisk will just pick times

Hi, everyone: I am developing a system using Asterisk, TDM-400 analog cards, analog lines, and Polycom SIP phones for internal extensions. Initially there was bad echo but after a series of efforts, I've managed to reduce it to a negligible level (it only happens when both parties speak simultaneously, and even there, only for a few hundred milliseconds). From an echo standpoint, things are

Anyone ever seen MeetMe cause * to crash? Specifically, it happens consistantly if someone begins to enter a conference and then decides to hangup while Allison is introducing them - like playing back "conf-onlyperson". This has been seen with the MeetMe participant connecting via IAX and SIP (not saying it doesn't happen with Zap, just that I haven't seen it). The box is *

Hi. The UPGRADE.txt of asterisk distribution contains the following snippet under the MeetMe heading: /"MeetMe: * The conference application now allows users to increase/decrease their speaking volume and listening volume (independently of each other and other users); the 'admin' and 'user' menus have changed, and new sound files are included with this release.

Dollars to donuts it is related to these two posts, but no one seems to know where or why it happens - this issue doesn't seem to be related to one specific piece of hardware: Post 1) ********************************************************* Anyone ever seen MeetMe cause * to crash? Specifically, it happens consistantly if someone begins to enter a conference and then decides to hangup while

I've got a simple MeetMe conference configured using Asterisk 1.0.3 on Gentoo. I'm using zaprtc for timing from the bri-stuff package. extensions.conf exten => 37455,1,NoOp(Drill Squad Conference) exten => 37455,2,Monitor(wav,drillsquad-37455,mb) exten => 37455,3,MeetMe(37455,pMs) Now, when I enter the conference as the first call, the MusicOnHold plays, but it's blasting

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,

Asterisk intermittently does not send audio back to the callers in the meetme conference bridge. If the caller hangs up and calls back sometimes the audio will work and sometimes it does not. We have taken packet captures and reviewed the SIP and SDP, both are correct and you can actually hear the audio being transmitted from the callers to the conference bridge but no audio is sent back to the

One of the features that I thought would be popular with the Web-MeetMe suite is the ability to start all non-admin callers in a muted state and selectively unmute them. For example any large conference that is of an announcment nature with a Q&A session. It's probably a feature I should have tested better, but I just discovered that a caller that is joined to a MeetMe with the |m flag

System: RHv9 cvs-head-11/23/04 with TDM04B and T100P (isdn pri). We've all seen many posts over the last several months regarding the stability of the TDM card. Today, the above system's TDM04B card (4 fxo ports) failed at approx 15:54:49. [root@asterisk asterisk]# cat /proc/zaptel/2 Span 2: WCTDM/0 "Wildcard TDM400P REV E/F Board 1" 25 WCTDM/0/0 FXSKS (In use)

> On Mon, 26 Mar 2007 10:42:38 -0700, Stephen Uhler wrote > > >>>GDrayer@guesswho.com said: > > > I have been successful using ChanSpy on a standard Dial call but when > > > attempting to ChanSpy on an incoming call that has been added to a > > > MeetMe conference (attempting to coach an agent that is speaking to a > > > conference of callers)

I haven't seen this appear on the list, so I thought I would resend it... Sorry for the repost if it did appear before... ----- Forwarded message from Michael George <george> ----- Date: Wed, 3 May 2006 21:48:09 -0400 From: Michael George <george> Subject: meetme conference latency degrades... To: asterisk-users@lists.digium.com We have recently started making more frequent use

hello, I will do a server to do a lots of conferences (MeetMe). I want to know that if I dont use a digum card, the limit of simultaneous calls is harder without a card than with a card ?if, yes, how harder is the limit? thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part --------------

Hi Guys, sometimes if one caller or many callers are in a meetme Room and a new one join the room, then he or another caller into the same room where kickt from the room. It's very strange for me and in logs (full) I can't see anything. is it possible to log more from meetme.c ? can anyone help me and maybe someone has also the problem as i and have an solution. I use: asterisk-1.6.2.7

We have recently started making more frequent use of the meetme conference of our * system. We are using v1.0.8 with a 2.6.11 kernel on our system. We generally have 4 callers in it: two with the gsm codec and 2 with g729. Initially, the conference works fine and there is little latency. After about 15min., though, the latency is very noticable and by 25min it's unbearable. If we all leave

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,

I'm looking into deploying * for an internal conference call server (using MeetMe) and had a couple of quick questions for those of you who have used it. I checked the Wiki but there weren't a lot of details for MeetMe. - Can you limit the size of a conference "room", ie max 8 people, etc. - Is there a list somewhere (besides the source ;) that has all the commands availible to