similar to: Calculating required bandwidth

Hello Jean-Marc: I've been watching the speex development from its inception because I and several Tech Startup Connection members have a very important application for this voice encoding/decoding. Further, we are quite familiar with ACELP as implemented in G.729. As far as I know Speex is also ACELP ... yes/no? Question ... what do you see as the advantages of ACELP compared to MP3? I

Hi, Is there any tested good HTB script for traffic shaping available like as that of CBQ available at. http://freshmeat.net/projects/cbq.init I am n new bie and need to work on htb. -- Regards, M Arman _______________________________________________ LARTC mailing list LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/cgi-bin/mailman/listinfo/lartc

Hello everyone, I''m a newbie experimenting with CBQ shaping and am facing a few problems. Can any of you please help? TEST SETUP: +---------------+ +----------------+ | 10.0.0.103 |----------->| 10.0.0.102 | +---------------+ +----------------+ 10.0.0.103: Linux, 100Mbit/s NIC 10.0.0.102: Windows, 100Mbit/s NIC, iperf tcp server (ports 2000 and 2001) WHAT I

I need to set up an office full of Cisco 7960 phones behind NAT with the server out in Colo. The first test phone registers fine, but the second one does not register. The first phone's registration looks like so: /SIP/Registry/3115552368 :64.169.xx.yyy:38836:3600:3115552368:sip:3115552368@64.169.xx.yyy:5060 When the second phone tries to register, it gets back a 404 not found. Not a

2003-06-28 Bandwidth Study - John Todd (jtodd @loligo.com) Purpose: ------------- To obtain a better chart of actual bandwidth usage per codec as seen "on-the-wire" when using IAX2 trunking between two Asterisk telephony servers. Discussion: ------------- Past threads on the asterisk-dev and asterisk-users lists have indicated that the optimal way to save bandwidth on

---------- Forwarded message ---------- From: Marco Mouta <marco.mouta@gmail.com> Date: Feb 15, 2006 1:58 PM Subject: Which ATA device do you recommend? To: asterisk-users-request@lists.digium.com Hello, I'm developing a Voip Solution for a client, which ATA SIP do you recommend? there are some ATA devices fully tested with Asterisk? I hope that Asterisk experient users could give me

Anybody using a Sipura 3000 for FXO with Asterisk? Mine is working except for one small nit... When a call comes in from the PSTN, the Sipura answers it and then passes it on to Asterisk, which plays extension ring tone. I'd prefer for the POTS line to stay on-hook while the extension rings, and to only be answered by the Sipura when the extension answers. Has anybody made this work?

I have several DIDs (working well) with LiveVoip and I just signed up for some outbound minutes. Unfortunately they did not send connection instructions. I tried: exten => _1NXXNXXXXXX,2,Dial(IAX2/userid:password@217.160.244.186/${EXTEN}|60|s) but I get the error Feb 19 15:14:09 WARNING[21453]: chan_iax2.c:5546 socket_read: Call rejected by 217.160.244.186: No authority found --

And the main point of interest is the following: how much difference in SPEEX terms would it be if it were developed with ACELP in mind? (not that i would do an aSpeex codec implementing ACELP, just to know and point my efforts) -- John Villar Gerente de Proyectos Computadores Flor Hard Soft 2058 C.A. www.florhard.com -------------- next part -------------- A non-text attachment was

Our puppetmaster runs 0.24.8 on Ubuntu 9.10. Our clients are either Ubuntu 9.10 or Ubuntu 8.04. We''ve just brought up our first Ubuntu 10.04 machine. This machine installs puppetd 0.25.4. The Ubuntu 10.04 machine can''t seem to present it''s certificate request properly. In my masterhttp.log, I see [2010-10-20 13:09:06] 174-143-141-55.static.cloud-ips.com - -

Does anyone know of a speech recognition module (like say yes or no, or numbers) I guess due to the complexity of speech recognition it might just be found in commercial applications or am I wrong like always? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050128/119168cb/attachment.htm

Hi List, I am a newbie, just came to know about asterisk a few days back while installing suse 9.2. I have a question for which I am sorry to say,but I havntread through all the archives, but AFAIK i didnt get the answer in the archives. My question is like this. I am in Nebraska, US. I have broadband cable connection at my home. And I have friend and family in other country. Using asterisk

Hi all, We have been fighting with telco for a entire week. Today they came here with a LITE3000 to analyze what is going on. When I configure zaptel with no external clock, E1 gets aligned/synchronized with bit rate in 2048000 bps, both me and telco. span=4,0,0,ccs,hdb3,crc4 But when I configure span4 to get clock source from telco they become unsynchronized. TElco bit rate stays in

Can anybody tell me the difference between a TE410P and a TE405P? Is it JUST the 5v vs 3.3v pcis slot spec, or is there some thing else? </edg>

I need to send calls to a bank of servers in a round robin fashion. Has anybody implemented this in dialplan language? If not, in some other fashion. </edg>

Hi All, First of all, I'm sorry if my question is offtopic on this list. In such case please ignore this post and/or contact me directly. I'm asking my questions there because I feel you had similar problem before starting developing Speex. My story: my friend developed 3gpp content creator and he would distribute it in binary form. But there is problem with AMR licensing (the terms

I am looking for a good provider for IAX2/800 termination. I am currently using FreeWorldTel and wanted to use NuFone but it seems that both of them don't provide customer service. FreeWorld has terrible voice quality and NuFone never answers their phone or responds to messages. Thanks, Linn

Thank you Jean-Marc. My understanding is that G.729 is a telephone codec, so there must have been some reason why its developers went to 10ms/frame. Do you know why that might be? From a recent post on this list I saw somebody talking about your decoded sample rate being 8KHZ/sec. and then he mentioned that being 160 bytes at 20ms/frame. That said, I take it that your decoded samples are

Hi, I'm taking a straw-poll to see who out there is planning on going to AstriCon. I would like to hear from both new members of the community and gurus. What kinds of things would you like to see at an Asterisk Conference? What topics are good BOF (Birds Of a Feather - informal discussion group) fodder? What parts of Asterisk require the most attention? FYI - AstriCon is October 12 - 14

Hello Jean-Marc: How many bits do you have in a frame please and what is your frame rate? For example, G.729 (ACELP) has 12 bits/frame including VAD and the frame rate is 100/second ... I'm looking for the comparable figures for SPEEX. Thank you. --- B. Mitchell Loebel, CEO, VP Engineering 408 425-9920 InstaFlash International Corporation (formerly