similar to: Server based audio merge

Hi Allen, > I tend to disagree. It normal human conversation it wouldn't make much > sense to have 2 people talking over each other at the same time. One of the problem is, that if the server doesn't distribute the stuff, then one entity must send the stream to every other entity. That could work fine with fast connections, but doesn't work with a modem connection. My

I tend to disagree. It normal human conversation it wouldn't make much sense to have 2 people talking over each other at the same time. Thus, it most scenarios you would have only one talker anyway. Additionally, encode->decode/mix/encode->decode isn't a very efficient CPU process for a server, it's complicated to keep timing correct and it has a negative impact on total

Hello: I would like to know if there's a way to request free chanels from remote asterisk servers ? My idea is to make an agi returning a dial to inter-asterisk connected servers when there's not enought chanels on local server, maybe like a ping to all of them or maybe requesting to a central server where all the *s send and request information about available chanels each 2 or 3

Hi Carsten, due to the ongoing discussion on both lists, i simply respond to both lists. it's hard crossposting, but it's for both roups relevant (i think). <p>+After having thought about control structures, it makes sense to me to do the extra work and merge this creamed cake into a jabber server component. Otherwise a control channel to the server component would have to be

Hi Ulrich, > Well, i don't know if we speak of the same ... The quality is ok, but > the encoding of a PCM frame on the tested machine took 2ms, which (if > the PCM frame is 7ms in length) means the machine can only encode three > streams in realtime ... Yes, but this is java. You don't use any threading models and i dont know if you encode from sound card. I also don't

> I tend to disagree. It normal human conversation it wouldn't make much > sense to have 2 people talking over each other at the same time. Thus, > it most scenarios you would have only one talker anyway. Additionally, > encode->decode/mix/encode->decode isn't a very efficient CPU process for > a server, it's complicated to keep timing correct and it has a

Hi all, <p>i have send a mail to the jabber mailing list and ask them how to send speex data with the jabber instant messaging protocol. I have added the mail adter this one. If someone here has any experience with jabber and speex please let me know. Thanks, <p><p>Carsten Breuer ====================================================== Hi all, im new on this list, so i want to

Hi all, <p>im new on the list and want to introduce my self. Im a software developer working on engine control software for a german car manufacturer. I want to use speex for a privat videoconferencing project and faces some problems with it. First of all the projects in Visual-C++ doesn't work. I don't mean the path an dependecy problems. That is easy. But there are some errors.

Hi Allen, > Speaking of video conferencing in conjunction with Speex, we are > currently beta testing a solution we developed that offers multi-point, > multi-party video and audio using the Speex engine for voice. I have visit your website and the solution looks interesting for big entities. Of course it is comercial, so it's not in my scope. I'm taregeting smal entities. Some

Hi all, I just installed an TDM02B. My system is a dell pc with linux 2.6.12-1.1456_FC4 asterisk-1.2.0-beta1 zaptel-1.2.0-beta1 libpri-1.2.0-beta1 in /etc/zaptel.conf I have (all others are default): fxsks=3-4 <--- I saw light in the ports channels=1-2 <--- change it to 3-4 has same result but... [root@nmsd0 asterisk]# /etc/rc.d/init.d/zaptel

Hi all, I want to check if this is a bug for which I should file a report. I am using R2.5.0 on OS X 10.4.9. When I invoke the data editor and when I change the values of individual cells, it seems to work as intended. However, when I try to delete/add a row/column, R.app crashes. I've attached the crash log. Best, -Nathan -------------- next part -------------- An embedded and

I would like to participate also. Tony T. ----- Original Message ----- From: "J.K. Lin" <jk@pageshare.com> To: <speex-dev@xiph.org> Sent: Wednesday, November 19, 2003 12:29 PM Subject: RE: [speex-dev] Videoconferencing with speex and jabber <p>> Me, me, me, me too. > I would love to use it if there is one ;-) > (I wish I can contribute more, but Windows is

Hi, please excuse my poor englisch. Is it possible to connect a (privat Test-Asterisk) to my privat ISDN and allow him to only answer one dialed number? We have 3 up to 10 Numbers on each (Euro-)ISDN (2 b-chanels), it cant't be done by the last Digits cause the numbers are completely different. For Example: I have 3 Numbers (641717, 928752....) Is it possible to tell Asterisk (in

There's no perfect solution to the multiple client problem. Each approach has advantages and drawbacks: 1) Mixing at the server - Allows a constant bandwidth for every client - Allows compatibility with regular VoIP prones - Requires transcoding, even when only on person is talking - Higher bit-rate required for the general case (one speaker is talking) 2) Sending multiple streams - Possible

Hello, I've got two streams, one for broadband, one for dialup. Well, having had occation to use a dialup connection recently i checked the dialup stream. Although it was streaming what the broadband stream was, the audio quality was audibly worse. It didn't buffer, but it didn't sound as clear as the broadband stream. I used lame to encode the tracks to mp3 and used it's

hey McFlow, this theme actually important for me now )) Me and my friend have the same problem in Command & Conquer 3 Kane's Wrath if you still there (or somebody else) please, tell us how did you fix it??? :( I have XP on my PC, my friend using Linux Gentoo. we make a chanel in hamachi, enter it. We pinging each other good but in the game I don't see him... and he sees me and my

Version: 2.2.9 & 2.2.15 ( http://ppa.launchpad.net/malte.swart/dovecot-2.2/ubuntu ) OS: Ubuntu 14.04 Server LTS I installed Ubuntu 14.04.2 LTS newly on my new server... And I try to configure the mail server considering both my system account and the virtual mailboxes, by seeing https://www.exratione.com/2014/05/a-mailserver-on-ubuntu-1404-postfix-dovecot-mysql/ But when I configured by

Hello, I'm about to install asterisk as the PBX at a location that my company has just moved into and I would like to get some comments and advice on the installation. I am new to * and don't want to make any big mistakes so I would love to hear whatever anyone has to say. Here is what I have so far Server: * 2.8Ghz P4 - 1G ram * T400P Tormenta II (is this as good as the

Hello, I am working on a trixbox based system with a TDM410P connected to 3 phone lines from the CO. The asterisk box is on a full duplex 100Mb LAN with some polycom and Aastra SIP phones. In general everything works. the problem I am trying to solve is that if both parties to a call speak at the same time one of the voices gets cut out such that the talker A cannot hear what talker B is

Hi, Currently streaming ogg isn't practical in this situation. That was one of the first things i checked into. WHen i looked i didn't see a streamer that did both ogg and mp3. Thanks. Dave. ----- Original Message ----- From: "Daniel Ballenger" <lpmusix@gmail.com> To: "Dave" <dmehler26@woh.rr.com> Cc: <icecast@xiph.org> Sent: Friday, December