similar to: Voicemaster

What is the configuration of H323.conf and openphone in order to run openphone and asterisk together ?

> 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

Is there anyone with experience how to integrate Sysmaster's VoiceMaster? Please can you share your experience. Thanks. Habiyakare Aimable Voice Services Terracom Communications Tel :(250)08435550 SIP:04400104@voice.terracom.rw E-mail:aimable@terracom.rw MSN:aimable@terracom.rw -------------- next part -------------- An HTML attachment was scrubbed... URL:

Update on this problem: I gave up on the "native" h.323 because, like others, I couldn't get audio working. (yes, I tried disabling FastStart in ast_h323.cpp - no change) So I went and got the OH323 code from www.inaccessnetworks.com. Glad to say that everything seems to work so far. Not only does audio work, but even the handshaking is now working in both OpenPhone and even

Hello I am a newbie to Asterisk. We have set up Asterisk on a PC with Redhat 9.0. We have installed H323 openphone on our PC's. We are wondering what a gatekeeper does. It seems we need one but what I have seen in this group is that a gatekeeper must be installed on another box on the network. As all our PC's on the network use Microsoft OS is there a free gatekeeper software for

Hi, I am a newbie in asterisk, i could compile it and run it with no problem on a RedHat 9. In the same box, i got a linejack and a phonejack cards and i downloaded the CVS driver from quicknet. This 2 card were working in a openh323 (openphone and pstn) project with gnugk on a RedHat 9. I am using the default samples, and i tried /dev/phone0 and /dev/phone1, but when i run asterisk, i get this

I'm working on calls coming in to an asterisk box as H.323, and going out as SIP to a remote device (a VoiceMaster). The remote device is refusing the calls with SIP error 406 (Not Acceptable). I have attached the SIP debug output below. It looks like codecs overlaps - can anyone see why the call is being refused? (Note that I'm not registering with the remote SIP device, just

Hello- In order to satisfy a customer requirement, I've just build H.323 under asterisk (using the specified versions of OpenH323 & PWLib, and trying to follow the instructions religiously), and it seems to have come up fine. When testing with with OpenPhone (Windows version 1.8.1) and NetMeeting, I've gotten some intermittent results however. All my calls are from a PC to asterisk -

Hi, I just installed OH323 Plugin and im now tryin to make simple Configuration to connect Openphone and Xlite to my Asterisk-Server. All works fine, i just wanna know if there's a better way to do it? Is there anything wrong with my Config? OH323.conf [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=8000 udpEnd=8005 fastStart=no

I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711). But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message: -- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack --

I'm having a small issue with the oh323 implementation when it comes to codec selection. Version info: CVS Head 6/30/2004 OH323 0.6.3 OpenPhone for windows version 1.8.1 Asterisk is configured as a h323 endpoint which either terminates to the PSTN locally through a PRI or terminates the h323 call to an IAX provider remotely. Asterisk also has G729 licences installed. in oh323.conf we

Hi Mark Funny you should ask this question, I just spent yesterday integrating building asterisk with h323 support to connect to a Cisco call agent.....I cant say if it will work for you but it compiles and loads nicely ! I will be testing this evening.... # cd /root # wget http://www.voxgratia.org/releases/pwlib-Pandora_release-src-tar.gz # wget

In my test box I've installed chan_h323 and I've been testing it with Micro$oft netmeeting and openphone with success. I alos have in my installation a Cisco 1700 series router with an FXS card on it. On the router I places the g711-ulaw codec and it worked but I experienced one bad thing. When I made up more than three calls, in the first three calls I was able to transmit and

Hello I am trying to compile asterisk-oh323-0.7.0 with pwlib-Janus_patch4 and openh323-Janus_patch4 downloaded from inaccessnetworks so I did this: tar -zxvf openh323-Janus_patch4-src-tar.gz cd openh323 patch -p1 < /root/asterisk-oh323-0.7.0/openh323_1.13.5-make.patch ./configure make opt cd asterisk-oh323-0.7.0 vi Makefile (to set the paths and options according to my system...) NOW I

hello all i am having a problem in compiling openh323. [root@kamran openh323]# ./configure checking for g++... g++ checking for C++ compiler default output... a.out checking whether the C++ compiler works... yes checking whether we are cross compiling... no checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C++ compiler... yes

i ve downloaded asterisk-oh323-0.6.6.tar.gz I am getting this and anybody know howto fix this? #tar zxvf asterisk-oh323-0.6.6.tar.gz oh323]# cd asterisk-oh323-0.6.6 asterisk-oh323-0.6.6]# ls asterisk-driver CONFIGURATION Makefile rpm TESTS BUGS COPYING README rules.mak wrapper asterisk-oh323-0.6.6]# make for x in wrapper asterisk-driver; do make -C $x

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 all, i have one (teles) pbx with a BRI telephone and an outgoing E1 port. The outgoing E1 is connected to an pri_net port from my *. The incoming call will dail out to a h323 soft phone like openphone or sjphone or just netmeeting. The call will be conneted, but i don't hear any sound, from no one of the both sides. Can somebody help me? Thanks, Thomas.

Hi, I run asterisk with oh323 plugins.It runs correctly with sjphone H323 Gatekeeper. But When i run openphone it doesn't recognize my asterisk server like a gatekeeper !! What is the problem ? Thx

Hi all: I've got some core dumps when I use chan_h323. I dial an extension using h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes * hangs, sometimes not. The client used for test es SjPhone (http://www.sjlabs.com/). This is the data for one core dump: (gdb) bt #0 ast_rtp_get_us (rtp=0x0, us=0x5074759c) at rtp.c:790 #1 0x41f8879c in create_connection