similar to: ACM codec

Hi Brian- BT> I was going to; asked on this list about it back in December. But the BT> documentation on the MS website stinks ... it basically says, "Look at BT> the example GSM 6.10 codec provided in the Microsoft Driver Development BT> Kit and replace the application-specific bits" ... but, while I own a BT> student edition of MS Dev Studio, I don't have an

Hi, Speex is getting close to beta4, which I'd like to be feature-complete (or as close as possible). That's why I'd like to ask if anyone here has needs for a feature that hasn't been implemented yet. If so, please let me know. For those interested, here's what's going to be in beta4. First, the VBR code has been greatly improved and now works good with wideband too.

X-Fixer wrote: > Hi- > > is someone working on ACM codec at the moment? > if not, I'll start writting it one of these days. > > -XF <p>I was going to; asked on this list about it back in December. But the documentation on the MS website stinks ... it basically says, "Look at the example GSM 6.10 codec provided in the Microsoft Driver Development Kit and

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

> 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

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'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've oh323 chan installed and working to make calls from SIP to H323 devices. The problem is can no hear sound with the H323 device. I think this is some related with codecs o nat, because the H323 have one public IP from a different subnet from the asterisk box. If I use netmeeting in gateway mode, the call can be completed and I can talk with a SIP device, but in gateway mode I can not

Hi Every time I hangup on my AGI script Asterisk crashes if it is not running in console mode. (happens when using python and perl AGI scripts) I'm desparatly trying to get my employer to let me use Asterisk. So I must get this to work. I've posted about this before, I'm sorry, but I'm desperate. I'm running RedHat 9.0 (kernel 2.4.20-8 everything else updated) I'm

Hello all, Please forgive me if this sounds a little (or a lot) ignorant as I am very new to asterisk. Right now I have two pc's connected back to back through an E100 card running asterisk. I have openh323 running as well and I am able to route calls through the E1 line. Next up I would like to be able to register asterisk with a gatekeeper. On another computer is running openGK. Using

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 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 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 guys, i'm encountering one way audio on cvs using netmeeting and chan_h323.so is there a quick fix or workaround for this? compiled using openh323 1.12 pwlib 1.5 i also saw this in earlier version of openh323 and pwlib.... thanks for any info ~kelvin -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

> 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

Mr. Blum, I had to make some assumptions writing the driver, because I had no previous experience writing ACM drivers. I did what I could based in Microsoft's documentation but, of course, I made same mistakes. I had no time to test the code I wrote yet and will not in the next 2 weeks. All I can tell you now is about what I did and about my sources. So, considering your enthusiasm, I suggest

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 Friends, Hope you would help me out here, I have searched the asterisk user list for hours and also read the readme and test files that comes with the driver. I need a very simple scenario. I have SIP clients and want to use oh323 to dial out to PSTN using a h323 gateway. a)If I set the extention.conf like this: exten => _87.,1,Dial(OH323/16.52.153.206) oh323 dials out (I can ring 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