similar to: frame size

> Framesize always refers to the decoded data frame size in samples. > Framesize is dependent on the encoding mode > Narrowband (8kHz): framesize = 160 samples = 320 bytes of PCM > The size of the encoded data depends on the quality setting, so if you > know for instance that you are using quality 3 on narrowband, that is > 119 bits of encoded data per frame which is rounded to

> 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

All the clients that I'm aware of for IP telephony have drawbacks. Some won't work at all. KPHONE -- Kphone works best for me, but Kphone doesn't have a dialpad to dial tones during the middle of the call, so the demo that * comes with can't be run. Kphone (3.1, the latest) also has a habit of crashing if you do something even mildly stressful, such as hang up while Kphone is

Hello! I'm about to try to integrate SPEEX into the VOCAL project. If anyone has any pointers as to the best way to do this, please let me know. After reading the speex api man page, I have a few questions: 1) To encode, it appears I need an array of floats. If I am playing wav files, what is the best way to turn these into something speex can encode? 2) Are there any commercially

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

just to check that I've got everything right. the encoder allows as input - mono only - 8 or 16 bits as floating point numbers (without scaling to 1.0). floating-point wavs (IEEE) will also work, but it's better to scale them to something like 8000. - any sample rate (should be set with speex_encoder_ctl), but prefered are 8/16/32 kHz. but what modes should I use for a given rate? also -

How will this fare? I am planning on putting an asterisk box for my brother in the Philippines but they only have dialup internet. I want them to be able to use a telephone set on a phonejack or linejack card and call me and vice versa via VOIP. My setup in the US is working already with a broadband cable connection. I am thinking that dialup may not work because of the bandwidth required

-----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 am using ohphone (well, I am trying to) to make calls. I will make an H.323 - SIP Gateway but I don't understand the architecture of all this. What is the exact role of asterisk? It can be used as gateway, that I know, but what else can he do? Is it necessary to have ohphone to make calls or asterisk can also do that? So when the gateway it is going to be implemented how is it

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 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 There, i think all VOIP clients for Linux are unusable! i got testet: Linphone + Linphonec all in version 12.2 Kphone gophone and other... the only programm that is usable is gnomemeeting... does anybody knew some other tools? Best Regards, Mark

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

I've a problem with my X100P card. I'm setting up a VoIP to PSTN gateway,with oh323. This works, but when I call an PSTN phone number, some digits are duplicated, so I'm unable to call the right person. Not very clear ? I'll try to do better (sorry, I'm french...) example : I use ohphone (with quicknet hardware), I call asterisk (*192*168*1*204#), asterisk answers, I choose

Hi! When I try to make a call with ohphone, that is the message I get: Call to "06302" aborted, insufficient bandwidth Can anybody tell me a solution or a reason why this messages appears? Thanks a lot! Regards, Mireia

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

Hello, I can use directed call pickup using pickup application (between sip, iax, skinny cals), but unable to pickup call that is ringing on phone behind h323 gateway (using original h323 channel in asterisk), is this even suported? thx PJ exten => _*7.,1,Pickup(${EXTEN:2}) console log, when trying o pickup ringing line 324 (h323), from skinny phone (953) -- Executing

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 -

Hello all, Encouraged by the successful "demo", I'am getting on with Asterisk CVS. I added 2 H.323 extensions in extensions.conf [default] include => demo exten => 701,1,Dial(H323/gm2@192.168.1.20/s) exten => 702,1,Dial(H323/gm2@192.168.1.25/s) With: - [demo] is defined by default in sample.extensions.conf - Asterisk server is running on host 192.168.1.20, on the