similar to: Project Critique

Hi all I have a problem using the Speex voice codecs when using GnomeMeeting on one side and NetMeeting on the other side. I use GnomeMeeting under Suse Linux 9.0 to communicate with a friend working under Windows XP and using NetMeeting 3.0. Under Windows XP / NetMeeting we have installed and registered the Speex voice codec. (You can find more information how we have registered the Speex codec

Hi all, I have seen many posts on the Shorewalllists dealing with H323. Although lots of them indicated that this is difficult process with kernelrecompilation etc. I just tried what seemed to be logical for me. Surprisingly it worked. Configuration: WS1 ----- FW ------ Internet ------- WS2/Shorewall WS1, FW and WS2 run Redhat9 with its standardkernel 2.4.20. FW and WS2 run Shorewall

Hi! I am newbie as Debian user as Shorewall and as GnomeMeeting. I try to configure Shorewall but i have still problem with GnomeMeeting. I have Debian Sarge, Gnome and Gnomemeeting, standalone computer and dsl internet. Thanks, Mitja

In the company I work for, quite a few people use Netmeeting to share desktops during training. Anyone know of a way to either connect to their Netmeeting, or an alternative that will work on both Windows & Linux, and not require a server to host the meeting. Matt

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, i tried to configure Gnomemeeting for Asterisk, because its, how it looks, the only tool which gifes me all i want for the use in linux... I have allready installed and running h323 support in asterisk and edited the h323.conf. But i have no chance to configure Gnomemeeting that it connects with Asterisk! I found also nothing useful in the

> http://www.speakfreely.org/ > > I think this would be one of the best real-world tests of the speex codec. > This software doesnt use ACM or directsound api's but uses straight C code. > I was thinking the speexenc/speexdec should be easy enough to add. The last time I looked at this it was still very much old news - mostly half duplex audio, does not adhere to any

Hi to everyone, I am dealing with my primer Asterisk installation and we are trying to set up a H323 server in order to use Asterisk to place calls between NM clients (also Gnomemeeting). I have a basic extensions.conf file: [general] static=yes writeprotect=no [default] exten =>

X-Fixer wrote: > Hi- > > is someone working on ACM codec at the moment? > if not, I'll start writting it one of these days. > > -XF <p>I was going to; asked on this list about it back in December. But the documentation on the MS website stinks ... it basically says, "Look at the example GSM 6.10 codec provided in the Microsoft Driver Development Kit and

Hi all! I'm newie to asterisk and I've been trying to make it work in order to use it with Linux softphones (H.323, SIP or IAX, I don't mind) and none hardware phone. I'm using asterisk packages from Debian SID (my distribution), asterisk, asterisk-config, asterisk-sounds, asterisk-h323. I've still not tried with any IAX softphone (gnophone?) but with linphone (SIP) I've

Hi all, i've installed the a new pwlib (1.5.0) / oh323lib (1.12.0) on my *. Then i've installed the new chan_oh323 (0.5.6). when i try to make a call with "netmeeting" through * ( * dial out with "Dial,OH323/${EXTEN}@xx.xxx.xxx.xx" ) the call will be blocked. Before, there was chan_oh323 0.5.5 and pwlib(1.4.11) and openh323(1.11.7) installed, and it worked. Is here

HI, I'm using netmeeting to connect to an asterisk server and dial out. my extension looks like this exten => s,1,Dial,Zap/1/ Unfortunatelly the number that I have dialed in Netmeeting is lost ;-( If I hardcode the number on the line above, like ... exten => s,1,Dial,Zap/1/6642794 ... everything works fine What am I missing?

Hello all, Encouraged by the successful "demo", I'am getting on with Asterisk CVS. I added 2 H.323 extensions in extensions.conf [default] include => demo exten => 701,1,Dial(H323/gm2@192.168.1.20/s) exten => 702,1,Dial(H323/gm2@192.168.1.25/s) With: - [demo] is defined by default in sample.extensions.conf - Asterisk server is running on host 192.168.1.20, on the

The gnome2 port is broken? I updated the ports tree two time today, but the result is: rss@DaeMoN:/usr/ports/x11/gnome2> sudo make install clean ===> Installing for gnome2-2.4.0 ===> gnome2-2.4.0 depends on file: /usr/X11R6/libexec/cdplayer_applet2 - found ===> gnome2-2.4.0 depends on executable: gnome-cd - found ===> gnome2-2.4.0 depends on executable: gnome-dictionary -

Hi, I'm still struggling to make netmeeting work with asterisk and oh323. I'm dialing from netmeeting into a regular phone, connected to my TDM10B. everything looks great, except that I cannot hear my voice at the FXS side, just static that increases when I speak on netmeeting's mike. Nevertheless, if I speak on the telephone I do can hear my voice on my headsets. I configured

Hi- is someone working on ACM codec at the moment? if not, I'll start writting it one of these days. -XF --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed. Unsubscribe

hello is there a windows plug in and a linux plug in for net meeting to use speaks codes for voice chatting? there is a linux version of net meeting. thanks hank --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-request@xiph.org' containing only the word

Hello All, Finally I realised that the Asterisk demo setup didn't include support for h323. (Maybe it should have been obvious) so I went to work out how to get the h323 channel running. I had openh323 and pwlib installed as I'd been playing with vocal so it didn't take long to do cd asterisk/channels/h323; make; make install; make samples, copy the pwlib and h323 libraries to

I have hit a problem where chan_h323 sometimes doesn't hang up properly and stays stuck in the "Up" state, with asterisk consuming 100% of CPU: *CLI> show channels Channel (Context Extension Pri ) State Appl. Data H323/ip$127.0.0.1:30008/21552 (local 123 1 ) Up (None) (None) 1 active channel(s) *CLI>

I'm trying to set-up H.323 support under Asterisk. I built a recent CVS release and the ooh323c code from the asterisk-addons. Everything built and installed and the H.323 stuff loads OK when asterisk starts. What is the easiest way to check if the H.323 code is working? I've edited the h323.conf and extensions.conf files but I'm sure that things aren't right. I've

Hi there ! I searched the whole web to find some helping information about H323 Control Protocol, but there is no way to find that information. We compiled and installed asterisk_0.9.0 + pwlib 1.5.2 + openh323_1.12.2 + 'asterisk-oh323_1.5 channel driver + wrapper' and configured the dialplan for using our H323 Endpoints which are ip200 Innovaphones. Besides, we also use Gnomemeeting but