similar to: setting up ---- newbie

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 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, 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, 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

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

> 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'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?

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

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 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 am in a H.323 network with a gatekeeper and some terminals. Asterisk is a gateway between this network and the SIP network. Now I can do calls from de foreign network (SIP) to the locla (H.323) but I don't know how to do the inverse. The H.323 terminals use NetMeeting, and when I try to make a call, it says that the number dialed must be registered in the gatekeeper. How can I register

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 Just installed Asterisk on a Debian Woody/testing. I want to create a AGI script that is run after an outbound call is answered. I did this a while back (many versions ago). The problem is Asterisk does not seem to know the AGI application. I create a file test.call and place it in the outbound spool directory: the test.call file looks like this: #Simple test call script. #call my

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,

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 ~ I would like to : set up a server on Linux on which my friends can connect with msn or netmeeting, suporting at least sound conferance, and optionally video, but I dont want asterisk server to lock up the sound card; and then, I want to be able to connect that server with a free Linux tool; I had a look at http://www.voip-info.org/wiki-Asterisk but