search for: comfortable

Hi: When I call str() on an mlogit object, I seem to get warnings. This code is from an example provided in the mlogit documentation: library(mlogit) data("Train", package="mlogit") tr<-mlogit.data(Train, shape="wide", choice="choice", varying=4:11, sep="", alt.levels=c(1,2), id="id")

Hello, we have a small Asterisk Network where Siemens PBX's are connected via PRI (Zap) to an Asterisk and the Asterisk's are connected through IAX, so this looks like this: Phone1 --- Siemens PBX --- Asterisk --- (IAX) --- Asterisk --- Siemens PBX --- Phone2 Now, when Phone1 calls Phone2 all wents well until there is silence - then the line seems to be death. My users wanted to have

Hi All, I have a sip trunk up and running with a CUCM Express, passing calls fine except for a comfort noise error I'm getting on Asterisk: NOTICE[7520]: rtp.c:788 in process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: x.x.x.x I know Asterisk does not support comfort noise. I have "no comfort noise" on all

hi, when VAD is enabled, can i make the decoder simply produce comfort noise in the event that no voice was detected? i'm working on a p2p voice app. when no voice is detected, i'm thinking that i can make the transmiting endpoint send a signal to notify the remote endpoint that VAD is in effect, instead of having to send the whole packet that doesn't have voice anyway. on the

Hello, I am trying to use the preprocessor VAD to encode at lower bitrate during silence periods. I am able to run the preprocessor and get the VAD flag for each frame, and I am quite happy with it's performance. I would like to know how to pass the preprocessor VAD flag to speex encoder -- basically, i want to force the encoder to generate comfort noise when preprocessor detects silence.

Hi, How can I disable comfort noise on Asterisk? Szabolcs Szasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100129/3b27a653/attachment.htm

I have an Asterisk 1.4 box which is sometimes getting the message below. Here is the weird part, the CNG is coming from ANOTHER ASTERISK SERVER. 209.220.119.19 is an Asterisk 11 box. [Oct 8 11:59:27] NOTICE[20798]: rtp.c:849 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 209.220.119.19

Hi guys, I'm using Zyxel Prestige 2602R, as router/SIP-ua with my architecture SER+Asterisk. Normally, everything is fine. In these days I'm experiencing some problems: some guests said me that, if he call everything is right, but if is called, he cannot hear the caller. Immediately, I though into an RTP-Proxy problem, but is not. Then I saw that message appear on the Asterisk CLI, during

Hi everyone, I've got my * system up and running and I'm really pleased. I've gone with G.711 (alaw) and I've stumbled across a problem; when people place calls internally some people think they have been cut off if the line is quiet for a few seconds. Is there a way of getting comfort noise on the call? I'm using the STABLE release and cisco 7960 phones under FC-1 Cheers

...nd the Sweave FAQ suggest that Emacs would be a great development environment for working with Sweave. So far, it doesn't seem to fit that description... Maybe I'm doing something wrong. I am going to describe the way I do things, and hope that people on this list can suggest something more comfortable. I edit the file in one buffer and keep one open for the R prompt. Every time I need to recompile the whole document (Sweave to, say, DVI) I go through the following: 1. call sweave manually via the R prompt 2. open the resulting LaTeX file (I have not found a way to refresh this, maybe there'...

...f the tests failed because they cannot include alloca.h. This may be a linux dependency. Anyway, again I saw the same results with or without my change. I cannot run qmtest at all. I don't seem to have it or any way of building it. There's no rush to commit this. If you don't feel comfortable doing so, it can wait until Chris gets back. And as it's my first submission, you shouldn't feel comfortable :) Jeff On Sat, 28 Aug 2004 22:20:57 -0700 Reid Spencer <reid at x10sys.com> wrote: > Jeff, > > Chris isn't likely to respond to this for a while as he's...

Hi, the function TukeyHSD gives incorrect results for balanced incomplete block designs, as the example below shows, but I can only half fix it. There are two problems, 1. It uses model.tables to estimate treatment means, 2. It uses the wrong standard error The first problem can be fixed using dummy.coef, if the lines > TukeyHSD.aov function (x, which = seq(along = tabs), ordered = FALSE,

...of the ones I''ve checked out are anywhere close to being as streamlined and clean as I want. I''m thinking about writing my own forum, not only because writing your own code is the best way to get what you want, but also because im sure that''d be a good way to get more comfortable with ruby on rails... but before I jumped into that I wanted to see if there are any forum/message board projects that stand out as something everyone thinks is a good solid ruby forum. --------------------------------------------- Michael J. Anderson mja_online@hotmail.com

I had MCGP working to a ADIT 600 fine with debain sarge stable / asterisk stable - wanted to try red hat and got the below message - then I re-installed debian and am still getting the same message below - any comments are greatly appreciated - I did play with the config files with no prevail - the Adit seems to be doing its job per tech support at CAC. I listed my conigs below I go off hook

I am having issues with GTA:SA using the keyboard/mouse combination listed in the subject line. Character control is impossible. Does anyone have any ideas a a fix? Thanks, mp

Hi all, As an experiment in playing nice with others, we''ve added the ability in rspec''s trunk to do this: class ThingExamples < Spec::ExampleGroup def should_do_stuff ... end end This is how rspec 0.1 worked, and for people already comfortable with the classes/methods approach of Test::Unit, it is a more comfortable entry point to rspec. For others, however, it has created a problem: you can''t write helper methods that start with should_ because rspec treats them as examples. Quick show of hands, please: +1 (with comments ple...

...AMD ATHLON 64 DUAL CORE 4000+ (socket AM2) Mother board ASR SK-AM2 2 Ram Corsair Value 1 GB DDR2 800 Mhz Hard Disk WESTERN DIGITAL 160 GB SATA2 8MB I'm a newcomer to the Linux world. I started using it (Ubuntu 7.10 at work and FC4 on laptop) on a regular basis on May. I must say I'm quite comfortable with it, even if I have to re-learn a lot of things. But this is not a problem, I will improve my knowledge with time. My main problem now, is that I installed Ubuntu 7.10 Gutsy Gibbon on the new one amd64. To install R on it i followed the directions found here http://help.nceas.ucsb.edu/ind...

...AMD ATHLON 64 DUAL CORE 4000+ (socket AM2) Mother board ASR SK-AM2 2 Ram Corsair Value 1 GB DDR2 800 Mhz Hard Disk WESTERN DIGITAL 160 GB SATA2 8MB I'm a newcomer to the Linux world. I started using it (Ubuntu 7.10 at work and FC4 on laptop) on a regular basis on May. I must say I'm quite comfortable with it, even if I have to re-learn a lot of things. But this is not a problem, I will improve my knowledge with time. My main problem now, is that I installed Ubuntu 7.10 Gutsy Gibbon on the new one amd64. To install R on it i followed the directions found here http://help.nceas.ucsb.edu/ind...

Mojo with Horan & Company, LLC wrote: > And it makes *clear* calls assuming you're within allowable range. > Speakerphone seems to work well too. I meant to mention that the DTMF tones and dialtone sound like they're played at such a high volume that they clip through the handset's speaker. DTMF is rfc2833, so what I'm hearing through the handset isn't affecting

When I make a call using sip, I get the line NOTICE[327696]: File rtp.c, Line 292 (ast_rtp_read): Unknown RTP codec 19 received Repeated many times on the console ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to ;bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls allow=gsm allow=ulaw