similar to: Shorewall Firewall and Cuseeme or Netmeeting

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

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

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

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, 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

> As you can see Q4 and Q5 are good candidates providing one > frame per block at rates of 12800 and 16800 bits/sec. > SO I'll try the 16kHz modes next, as they should really be an > improvement to quality and clarity. I'll sort out the sources now and > mail them to you. Thanks for your hacked up instcodec source code. I installed the 16.8 kbit/s 16kHz Q5 mode and it

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'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, 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

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

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,

> 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

Ok, I uploaded that tool including sources onto my site (darkav.de.vu). bye, D A --- >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 messages sent to the list

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 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

Hi there I am pretty close on giving up on Asterisk :-/ I am (still) trying to make a call from a H323 phone to an Asterisk provider using AIX. But H323 does not route the number to AIX. All it is transmitting is an "s". *CLI> -- Executing Dial("OH323/R27865", "IAX2/demo:demo@gw1.musimi.dk/s") in new stack -- Called demo:demo@gw1.musimi.dk/s Jul 2

Hello richard, Experts , Sorry for my offtopic post, During our maillinglist disscussions in the emac-list i thought of making an announcement. Long times ago, there was a smart video chat tool, called CuSeeMe. I am tied a little bit on the tool. Nowadays with the all this Streaming Servers the time for video chat tools is coming back with power. So, I thought, it would be a smart idea

Today, the Mandrake Store here in the US have made available for sale their "Multi Network Firewall" (MNF). MNF is built around Shorewall and has a lot of nice features including a cool GUI. MNF has been under development for almost a year and is Mandrake''s next-generation firewall product. For those of you who are using Shorewall in a commercial environment and who are