similar to: How to define an extension for chan_h323

Hi all, I try to ask helps again about chan_h323 extensions. I define this in h323.conf: [general] port = 1720 bindaddr = 0.0.0.0 allow=gsm allow=ulaw gatekeeper = DISABLE context=default [gm1] type=friend host=192.168.1.20 context=default [gm2] type=friend host=192.168.1.25 context=default and I have in extensions.conf : [demo]

Hello all, I'am trying to use 2 gnophones on my LAN. But I can't get any sound. Here is my configuration: 1. on PC1, I have: - Asterisk compiled from CVS (CVS-05/15/03) and it runs. - Gnophone binary version from Debian: 0.2.4+cvs.20020624-3 - a sound card working ( module es1371 for /dev/dsp) - I registered it as "alice" in extensions.conf 2. on PC2, I

Hello all, It's me again. I would like play with calls between a H.323 client and a SIP client through * inside my LAN. For that, on host 192.168.1.20, I use GnomeMeeting (GM20) and Asterisk; on host 192.168.1.25, I use SJphone (SJ25) as SIP client on Windows and I register into * with a username, no password. The 3 files oh323.conf, sip.conf, extensions.conf are in attachment. In the same

Hello all and guest@misery.digium.com , I was surprised to play the "demo" extension from my Asterisk CVS. It was around "Fri May 23 22:18:37 CEST 2003". While trying to make a test with chan_h323, I got a "wrong" number and fell down on someone at this address IAX2/guest@misery.digium.com/s@default. The voice was clear but unfortunately I don't understand

Hi list, I'm working with gamm models of this sort, using Simon Wood's mgcv library: gm<- gamm(Z~te(x,y),data=DATA,random=list(Group=~1)) gm1<-gamm(Z~te(x,y,by=Factor)+Factor,data=DATA,random=list(Group=~1)) with a dataset of about 70000 rows and 110 levels for Group in order to test whether tensor product smooths vary across factor levels. I was wondering if comparing those two

Hello all, I tried to compile Asterisk from CVS yesterday. I would like to try gnophone as IAX client on the same Asterisk server with a sound card. I didn't have any problem of compilation. Here are the config files in /etc/asterisk: /etc/asterisk/asterisk.conf /etc/asterisk/extensions.conf /etc/asterisk/iax.conf /etc/asterisk/indications.conf /etc/asterisk/logger.conf

Hello I am not really a statistic person, so it's possible i did something completely wrong... if this is the case: sorry... I try to get the best GLM model (with the lowest AIC) for my dataset. Therefore I run a stepAIC (in the "MASS" package) for my GLM allowing only two-variable-interactions. For the output (summary) I got a model with 7 (of 8) variabels and 5 interactions and

Dear All Sorry for what appears a trivial matter - I'm new to R and am stumbling ahead. I have a table of numerical data (36 rows by 12 columns) such as below: GM1 GM2 GM3 GM4 GM5 ...etc GM12 Run1 1 2 1 2 3 ... Run2 2 1 3 2 1 ... ... Run36 2 1 1 1 1 I would like to plot simple line graphs of some of the runs or

I have been working on integrating some FXS ports into my dial plan delivered via a channel bank and testing with an analog handset. The receptionist is on Extension 700. All other SIP phones are 7XX. >From a SIP phone I can dial 700 and all other extensions. >From the analog handset I can dial any other extension but not the 700 number. Weird? Yep. The CLI does not show any dialing when I

Has anyone here done this? I got it compiled just fine but when I make a call I do not get any audio going either way. The * box is not behind any sort of firewall or nat. My H323 client (gnomemeeting) is behind NAT but I have it set up properly to work through NAT and it will talk correctly with my other regular x86 box running H323. One odd thing I note is that when looking at the UDP traffic

-----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

Hi all, I want to make extra columns in my datafile where the id of every groupmember is mentioned in separate columns. To explain it better see the example: id<-c(1,2,3,4,5,6,7,8,9,10,11,12) group<-c(1,1,1,1,2,2,3,3,3,3,3,3) a<-as.data.frame(cbind(id,group)) a id group 1 1 1 2 2 1 3 3 1 4 4 1 5 5

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

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'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 list, For months everything worked super here in our setup. This week I implemented some new idea in our webbased calendar system. I thought it would be nice to have an option that tells asterisk you are not available for calls during an appointment. For this to work I could no longer use the ringgroup setup: Dial(SIP/10&SIP/11&SIP/12,40,tr) So I thought, why not use the Local channel

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 -

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

Send CentOS-announce mailing list submissions to centos-announce at centos.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.centos.org/mailman/listinfo/centos-announce or, via email, send a message with subject or body 'help' to centos-announce-request at centos.org You can reach the person managing the list at centos-announce-owner at centos.org When

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>