similar to: Linux alternative to MS Netmeeting

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

I have just started laying out the plans for my first project using Asterisk. I am very interested at this stage in getting much needed feedback, critiquing my approach. What are the ups and downs going to be if I develop this project as follows: -The client wants to connect some phone reps in India through a VoIP to their clients. -There will be 3 phone lines, and 1 broadband internet

I've been banging my head on this one for a few days and am quite stuck. I've got a gatekeeper running and everything works there. Netmeeting works calling other netmeeting clients. Netmeeting calling asterisk connects, but netmeeting can't generate the signals to make the demo do anything other than talk. But connection from ohphone always disconnects straight away. I can't seem

Hi, I have a little question, I use asterisk with Netmeeting client. When I call netmeeting client with a phone. I don\'t have his ID in netmeeting window i have something like : ???;..dhz instead of 28. Someone know a way to display this ID ? Thanks you so much Rattana

has anyone managed to get these three beasties to work together ? we're using ooh323 from asterisk-addons-1.2.2, asterisk 1.2.6 and microsoft netmeeting default from windows xp. the symptoms are that calls from a SIP client to NetMeeting rings on NetMeeting, but upon answering the call in NetMeeting, no audio is passed between the two. eventually, the call times out and hangs up. on a

> Nice to hear! Do sou think you will be able to make the other modes also > compatible? But I guess these working modes are already OK for > netmeeting. I will try the Q4 16kHz mode today. Low bitrate was the design goal, not Netmeeting compatibility ;) Padding loss occurs because Speex encodes frames that do not end on byte boundaries. If you force each Speex frame to be byte

Hi, We have 2 offices interconnected with a VPN. This is the policy file in both of the Firewalls: fw loc ACCEPT loc fw ACCEPT #fw net DROP info fw net ACCEPT loc net DROP info loc vpn ACCEPT vpn loc

Hi, I'd kindly ask for any guidance how to setup Netmeeting to work with Asterisk. I've setup Asterisk as Gateway, selected GSM codec, and I'm able to call local extensions (no calls into PBX functions) but get no sound. Any hint, advice ? Anyone using Netmeeting (maybe also windows messenger) with Asterisk sucessfully ? Thanks in advance, regards, Robert.

Hi, I managed to get the codec into netmeeting. Unfortunately it doesn't properly work. I tried to talk vie net, but only erranous packets are decoded. Did I possibly register the codec wiht incorrecxt parameters or is this a problem of the acm codec? bye, D A --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from

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?

Well, the first day was interesting. We a variety of questions from why do we do this, what's going on with RH, how come this is the first I've heard of you. And hundreds more. The good thing is that we've recieve a very positive response from everyone that visited our booth. Such a positive response that all our CentOS CD's are gone, and most of the cAos CD's are gone, and

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

Hi, i want to control in my network, the netmeeting transfer of traffic, how can i control the audio or video transfer whether this services use dynamics ports? thanks -- Juan Felipe Botero Ingeniero de sistemas Universidad de Antioquia _______________________________________________ LARTC mailing list LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/cgi-bin/mailman/listinfo/lartc

Hi. After a small war with "underfined sybol" error and conflicts between h323 and oh323 I successfully install h323 channel. Now, I can connect from Netmeeting to SIP and ZAP channels, but I can't here anything. When I call at phone, and try to speak, on another end of line man said, that my voice very low. Microphone volume is maximum... Is there some parameters like rxgain,

Hi, > I can't tell why. Try to register mode 8kHz Mono mode Q3 or Q4 for Ok, I will try it. I tried 16kHz Q3 mono at 9,8kbit. > and upwards use 2 frames. All this is defined in codec.c in the > QualityInfo[] table. Oh OK. I looked a bit in the sources, but it was too confusing for me. ;-) > I would be interested in seeing the code for registering the Speex codec > with

Hello everybody, I wonder if anyone can help me in something I am trying to do but have no clue on how to do it: I have an Asterisk installation and I would like to be able to assign certain extensions to people with NetMeetings that take dynamic IP address. Does any one know how I can get the IP of an incoming channel in order to be able to dial back to that channel after the call is hangup?

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 =>

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

We have some custom software that is put out on our 6 clustered(balanced) webservers. Every 5 minutes that software checks into our server for a new version, if there is a new version it downloads it (20MB file). This happens about 2-3 times per month. Currently there are over 10000 clients that need to download the update. Up until this past week we were able to handle the load and this is