Displaying 20 results from an estimated 7000 matches similar to: "requesting real world meetme capacity numbers"
2006 Jun 03
4
Meetme versus app_conference
As stated here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe
A Meetme room uses Ulaw as the audio codec, so if the other channels
use different codecs, then * will transcode.
Does the app_conference application works the same way?
Or if i have SIP/g729 users and i create a conference with other users
also at g729 asterisk will not transcode (when using app_conference)?
2006 Mar 02
1
IAX Video and Meetme
Hi
I'm browsing around the internet looking for signs that the IAX client
library and app_meetme support video.
I stumbled across this post by SteveK on the 27th of Feb 2006.
"My company is looking to hire a full-time developer, who will be working
about 25-50% of the time on iaxclient; in particular to finally integrate,
build, polish and enhance video in iaxclient, add video
2004 Dec 28
1
Meetme scalable to 300 people?
Hi everyone.
I am looking at providing a conference for up to 300 people and was
wondering if anyone has scaled meetme to 300 people.
Here are some points:
1) I am using an IAX2 gateway hosted on a VOIP service provider.
2) The machine is hosted at the providers site so one has to assume that
bandwidth is not going to be an issue.
3) Everything is coming in as ULAW so we won't need to
2005 Feb 07
1
Conferencing without Meetme
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
2008 Dec 11
2
MeetMe echo problems with more than two participants
Hi Asterisk Users,
we are using Asterisk 1.4.18.1 on debian 4.0 etch, pwlib 1.10 and openh323
1.18.
We are using MeetMe for conference calls and with two participants there is
no echo problems, but with more than two participants there is a lot of echo
that sometimes disappear for a short time and all function well.
Someone have some suggestions??
Do you ever used app_conference
2005 Feb 02
2
MeetMe & ztdummy
I'm running into a bit of a problem setting up conference calls. The box
I rent at a colo doesn't seem to have USB hardware.... When I try to load
usb-uhci I receive a "device does not exist" error. Which means I can't
load ztdummy....
The system has a rtc clock module, so zaprtc won't work... (which I'm
scared to unload rtc because I don't have physical access
2006 Jun 04
1
Compiling VD_app_conference for x86_64
Do anybody could compile app_conference on x86_64??? I tryied with two
versions of app_conference and got the same problem on compiling:
relocation R_X86_64_32 against `a local symbol' can not be used when
making a shared recompile with -fPIC
app_conference.o: could not read symbols: Bad value"
ENVIRONMENT:
2006 May 29
4
app_conference DTMFs?
We need to conference together a call center agent, a customer and a third
party IVR and send DTMF tones from the agent to the IVR.
MeetMe has been eating our DTMFs so we'd like to try app_conference.
Has anybody setup such a configuration in app_conference and how did you
configure it?
-HJC
2006 Mar 02
7
G729 and Meetme
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
2007 Feb 26
3
Asterisk -> Streaming Audio Bridge
Greetings,
Does anyone know of a tool that can act as a VoIP client and stream to a
streaming server such as shoutcast/icecast, etc.
I've got a client interested in doing basketball play by plays during
tourney season. They have * in place now and the bandwidth to burn for
streaming out. In the old world, I did an analog phone patch -> mixer ->
encoder -> streaming server. What
2005 Jul 06
2
app_conference and AGI
Hi,
i was successful in compiling app_conference and setting up an
conference was quite easy. :-)
Does anyone knows if it is possible to have an IVR accessable from
inside the conference. So, if i dialed into an conference i want to be
able to press '*' and then the actual discussion is muted for me and i
and menu is read to me. Something like the ${MEETME_AGI_BACKGROUND} in
MeetMe.
2005 Jun 29
1
App_conference in dial plan?
Hi all,
I've been trying to get meetme working for a while now (complie problems
- will probably try again later on another machine) but have given up
and started looking at alternatives.
I've managed to get app_conference compiled and installed - show modules
shows its there in asterisk, but I don't know how too actually use it in
the dial plan...
The info on voip-info
2005 Jul 05
1
app_conference, CVS HEAD, SIP and Xen
I have Asterisk running in Xen virtual machines. Unfortunately, this
kind of virtualization makes a real time clock impossible, which in turn
makes ztdummy or a Zaptel driver impossible to load, which also makes
MeetMe conferences impossible.
As an alternative, I have downloaded, patched, compiled and installed
the app_conference source code against the headers in Asterisk CVS HEAD.
I can load
2006 Jan 31
2
app_conference(Asterisk) with Speex
I'm using Linphone. I tested with Asterisk and Speex only, I created a
channel with echo and it worked. It seems to have problem when using
app_conference.
Jonathan
2006/1/31, Steve Kann <stevek@stevek.com>:
>
> jonathan blais wrote:
>
> > Hi,
> >
> > Does anyone ever used Speex with app_conference in Asterisk ? I'm
> > having a hard time to figure
2006 Jan 31
2
app_conference(Asterisk) with Speex
Hi,
Does anyone ever used Speex with app_conference in Asterisk ? I'm having a
hard time to figure why I always get this error "warning: Invalid mode
encountered: corrupted stream?".
Jonathan Blais
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2006 Jan 31
2
app_conference(Asterisk) with Speex
Just curious, how does Asterisk pack Speex frames in a packet. AFAIK,
Linphone just sends raw packets, as specified in the RTP draft.
Jean-Marc
Le mardi 31 janvier 2006 ? 10:43 -0500, Steve Kann a ?crit :
> jonathan blais wrote:
> > I'm using Linphone. I tested with Asterisk and Speex only, I created
> > a channel with echo and it worked. It seems to have problem when
>
2006 May 30
4
I guess my server capacity is ok
can someone overthere help?
the server specs are as follows
HP DL380G4 Dual Intel Xeon 3.2GHz processor with 4GB RAM,
running fedora core 3
asterisk-1.2.5
ss7-0.8.3d.
using sip as advised to receive calls from another gateway in US.
using g729 in transcoding way.
however, I noticed the call hit the 51 active calls which is 102channels, I
run "top" to check the system resources usage
2015 Jun 22
3
Bi-directional sync for Sysvol folder -- Osync?
Hello Min Wai Chan,
?
Can you explain more about,, the DC1 will remove any emptey directory on DC1.
tested it here, but that does not occure here.
i can create empty directories on DC1, and these are synced to DC2. empty or not.
?
?DC1 will overwrite any users/group change on DC2?
if setup correctly, your sysvol rights on DC1 and DC2 are the same..
?
i suggest you to the following.
get the
2015 Jun 23
2
Bi-directional sync for Sysvol folder -- Osync?
Dear Belle,
That produce the similar situation.
Thus I'm trying osync
And the result are much present according to my test case
And configuration are much streamline.
However, I'm not sure how it can work with 2 DC or more...
Thank You
On Mon, Jun 22, 2015 at 9:05 PM, L.P.H. van Belle <belle at bazuin.nl> wrote:
> Hai Min Wai Chan,
>
> I have tested it as shown in
2010 Feb 24
2
audio glitches in conference
I'm having a problem with conferences both meetme and app_conference,
though I've done most of the testing with meetme.
Essentially, I get little gaps in the audio - usually fewer than a dozen
or so samples, though it does vary. They seem to occur at random, but I
usually get one ever few seconds, on average. They also seem to delay
some buffer somewhere, so that if I start recording