similar to: Signaling tones in Speex

Hi Jean, Thats great news for me to start off with as I was planning to go with 16 Kbps ADPCM keeping in mind the issues and options I had. Now, whether the additional computation cost is worth the significant bandwidth savings, I have to see. Just wondering if it is possible to extend this logic to G3 and G4 fax as well, i.e. using a higher bit-rate and complexity mode for modem or fax instead of

In case a system is incapable of fax relay or if it is disabled, one of the easiest and safest options is to go for 40 kbps ADPCM compression (for fax upto 14.4 kbps)..even am new to this problem and the fair bit of seraching which i've done seems to suggest that the standard sloutions are to simply 'bypass' it else compress using ADPCM (40 k for fax upto 14.4 k, 32 k for fax upto 9.6

Mainak Chakraborty wrote: > Hi Jean, > Thats great news for me to start off with as I was planning to go with > 16 Kbps ADPCM keeping in mind the issues and options I had. Now, whether > the additional computation cost is worth the significant bandwidth > savings, I have to see. > Just wondering if it is possible to extend this logic to G3 and G4 fax > as well, i.e. using a

Hi Mainak, Speex will definitely be able to handle DTMF. The only question is what the minimum rate for that will be. I remember testing on a few tones at 8 kbps and I didn't hear too much distortion (though there have have been). The only thing I can recommend is to use a higher complexity setting that one would normally use. For voice, I can't really tell the difference between quality

I am using Audacity to record some voice prompts. The .wav files I'm producing are of stellar quality. However, once I turn them into .gsm, they sound buzzy and muffled. I know that some of this comes with the territory, but I wonder if there is anyone out there who does this routinely, and who can advise me as to the MO I could use that results in the highest quality in the resulting

Is there any way to bring the bitrate in ices2 down below 32 kbps? I'm setting up a demo for someone of how to use linux to do net radio broadcasting. The setup I'm thinking of is to use ardour plus jack to mix two (maybe more) input sources (live audio and recorded tracks/programmes), then send the mixed audio to ices2 for streaming to icecast2, using the jackified version of ices2. This

Is there any way to bring the bitrate in ices2 down below 32 kbps? I'm setting up a demo for someone of how to use linux to do net radio broadcasting. The setup I'm thinking of is to use ardour plus jack to mix two (maybe more) input sources (live audio and recorded tracks/programmes), then send the mixed audio to ices2 for streaming to icecast2, using the jackified version of ices2. This

This wav produces a bit of audible static when encoded at the highest bitrate vorbis will allow me to encode at (30.7kbps avg.): http://staff.xmms.org/zinx/misc/5551234.wav.gz An encoded version, with the bitrate set to approximate 128kbps (I think it output to 29 some odd kbps): http://staff.xmms.org/zinx/misc/5551234.ogg This is with the latest CVS tree as of Sat May 12 10:06:17 UTC

On Sun, 29 Aug 2004 17:53:29 -0700, Ralph Giles wrote: > On Mon, Aug 30, 2004 at 03:03:28AM +0100, Andy Baxter wrote: > >> Is there any way to bring the bitrate in ices2 down below 32 kbps? > > Generally the trick for this is to downsample the audio before encoding. > You can ask ices to do this with a resample stanza in the config file: > > <resample>

> Tanya M. Lattner wrote: > >> How large of a change have you made? With 3 days before the branch >> creation, I strongly advise people not to be checking in major changes. > > Depends how you look at it. Structurally, it separates two files into > four and moves some functionality from one class to a new class, so in a > sense that's a big change.

Hi, We are experiencing this issue of redial dtmf tones generated randomly in the Voip calls, we have asterisk 1.4.21.2, dahdi 2.0.2.2 and we have dtmf as rfc 2833, we have a cisco router at the location Cisco 2431, 8FXS (only one FXS is used for Fax and rest are empy) connected to the netgear switch and all the phones are connected to this switch and there are no non sip devices in the

Hi, I am using SJPhone here for testing ivr with Asterisk. I haven't seen any problem here yet. I have tried different things for that and finally got it working. I am not an expert to explain more about that, but here is the general section form my sip.conf. dont know whether it will help... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ;

[image: Post]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD9wPTI4NjU%3D&b=2#28652>Posted: Thu Apr 09, 2009 8:34 pm Post subject: DTMF and IVR ... Sorry but URGENT[image: Reply with quote]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vcG9zdGluZy5waHA%2FbW9kZT1xdW90ZSZwPTI4NjUy&b=2>

Greetings list, Does anyone have any information (providers' support) about G.729E? Voip-info.org came up empty, the implementers guide from the ITU wants my credit card and the rest of the pages I found simply made a few comparisons between it and iLBC. >From what I understand, the codec is supposed to play nicely on lower power hardware but I can't find much more info than that.

We have 3 Grandstream Budge Tone 100 phones which are being very fluid on incoming calls. They are set up as extensions 2501, 2518, and 2536. When calling out to another phone, they always identify themselves correctly. But sometimes they will respond to the wrong incoming calls. (By respond, I mean that the phone rings and if someone picks up the receiver, the call then goes thru.) For

Maybe its just me but I find this question a little confusing, the tone duration should have no impact on tone recognition and typically in my experience the duration of the tone is defined by how long the user holds down the button !? > -----Original Message----- > From: James Sizemore [mailto:james@deny.org] > Sent: 22 August 2003 17:33 > To: asterisk-users@lists.digium.com >

Can someone who really knows the Ogg command-line encoder, help recommend the best setting for 33.6k modem stereo music streaming? (56k doesn't count cuz many people's 56k modems don't work at a full 56k, and I want them to be able to surf CD Baby at the same time as listening. 2 minutes / 120 seconds of audio should be about 400k.) I'm at my wit's end: tried everything I

Hi, How do I access the loop variables in a loop. for(i = 0; i < N; i++) for(j = 0; j < M; j++) A[i][j+k] = i + j; Is there anyway for me to know that in A[i][j+k], i & j are loop variables whereas k is not! Regards, Prasenjit Chakraborty Performance Modeling and Analysis IBM Systems & Technology Lab

Hello, I am having problems getting one xlite clients to communicate through asterisk. I am getting an error message: chan_sip.c:15593 handle_request_register: Registration from '"chinmay chakraborty"<sip:1234 at 10.44.32.193 <sip%3A1234 at 10.44.32.193>>' failed for '10.44.32.193' - No matching peer found sip show peers Name/username Host

Hi, I sent an email to Mark Fisheh of Oracle Corp. & posted this issue at OTN under Linux thread this morning. I hope that someone among you might have experienced this and can help. On that basis, I am sending this to you too. I am stuck & will really appreciate if you can shed some light on this. Thanks. Anjan