similar to: AGI + g729A

Hi, I'm quite new to R (1 month full time use so far). I have to run loop regressions VERY often in my work, so I would appreciate some new methodology that I'm not considering. #--------------------------------------------------------------------------------------------- y<-matrix(rnorm(100),ncol=10,nrow=10) x<-matrix(rnorm(50),ncol=5,nrow=10) #Suppose I want to run the

Hi. I tried the following: F<-function(x,y,b1,b2,b3) b1/(1+b2*(x+b3*y)); t<-data.frame(ExM=c(T$ExM,F(T$S1,T$S2,0.2,0.2,51.28205)),S1=rep(T$S1,2),S2=rep(T$S2,2),gr=c(rep(1,length(T$ExM)),rep(2,length(T$ExM)))); library(lattice); wireframe(ExM ~ S1 * S2, data = t, groups=gr); where T is da dataframe like T<-data.frame(S1=...,S2=...,ExM=...) But the resulting response

Hey! Could you please take a quick look at what I have done? Somehow I get wrong results using the anova(lm()) combination compared to doing a two way ANOVA by hand. Running: Data<-read.table("Data.txt"); g<-lm(ExM~S1*S2,Data); anova(g); Gives: Analysis of Variance Table Response: ExM Df Sum Sq Mean Sq F value Pr(>F) S1 1 4.3679

I seem to be having a problem with transcoding and/or agreeing on a valid codec. I am running a new image pulled from CVS at 1:30 PM CST. The issue occurs when I try to make a call to a toll-free number over sipphone.com. Here's what I see in the console: NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1259545280]: File

Hello! I am wondering how the differences between G729, G729a, and G729b effect call bridging and server interoperability. For example, can one server use the G729 code with another server that uses the G729A codec? Also, which version is Asterisk set up to use? Thanks! Elliot

Hello, When I'm trying to play standard sound files from Asterisk using G729A codec with OH323 channel I get this message: channel.c:1650 ast_set_write_format: Unable to find a path from GSM to G729A It seems that this files must be in G729 format? How can I convert this files to G729? ... or am I wrong? -- wbr, Oleg

Hi all, Will G729A codec exhaust the CPU power? If yes, how many concurrent sessions that P4 server board that can stand? Pls advise. Btw, if G729A has been purchased and installed, what will happen to the Asterisk Server crash say hard-disk when down or faulty, any where to do back up first such as "tar" commands? Any advice will be appreciated tq

Hello, My home system on a DSL line is getting worn out by bad behavior robots. Awhile back, I created a .htaccess file that block countries by IP blocks. Its 2MB in size. I have been running Linux since Slackware 1.0 and moved to Redhat around 2.0. I started after running a BBS using a doorway for newsgroups. Been hooked ever since. So, today, I tried following the directions for

Hello! I have a sip device that is sending in the SDP: rtpmap:98 g729a It does not seem like Asterisk is negotiating the codec properly, because while the call rings, the rtp lines fail. However, on other sip devices that have "rtpmap:18 g729" in their SDP, things work fine with Digium's commercial g729 license. How do I get "98 g729a" recognized by Asterisk? Thanks,

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the

Hi, I tried to play the standard demo-echotest file !. It works when i use an ip-phone (like x-lite or kphone), but as far as i use an PSTN Gateway (from an VOIP Provider) to call my phone - i get the following error: Sep 8 14:58:33 NOTICE[-182461520]: channel.c:1691 ast_set_write_format: Unable to find a path from GSM to G729A Sep 8 14:58:33 WARNING[-182461520]: file.c:779 ast_streamfile:

Hello, I am wondering about the ability of a server that is simply passing G729 through it to have the ability to record the calls. I know for voicemail, meetme, and things like that to work, a G729 license must be installed on the machine since there is transcoding going on. Is this also true for recording of calls? Will I require licensing for each recorded call? Will the server see a

Hi, R newb here. I've coded a function that inputs N dimensional array(s) [or class=numeric if it's dim=1] of coefficients and tstats, where dim(coef_matrix)=dim(tstat_matrix), it will then output a same dimension matrix of coefficients pasted to tstats in brackets pasted to significance stars. If I go straight to the code INSIDE the function, it works 100% as it's supposed to. The

Announcing Moo-Tiff for FreeBSD 2.1. Sender: owner-freebsd-announce@FreeBSD.ORG X-Loop: FreeBSD.org Precedence: bulk Moo-Tiff is a 100% compatible port of OSF/Motif 2.0 for FreeBSD 2.1, built from the official OSF sources and run through the OSF VTS (Validation Test Suite). If it works with OSF/Motif 2.0 on any platform, it can now work on FreeBSD 2.1 with Moo-Tiff ! Moo-Tiff is priced at

I have installed G729 but I cannot make a outgoing call with it. SIP/dennis-2c23 is making progress passing it to SIP/1010-8b60 NOTICE[311316]: File channel.c, Line 1476 (ast_set_read_format): Unable to find a path from G729A to ALAW NOTICE[311316]: File channel.c, Line 1446 (ast_set_write_format): Unable to find a path from GSM to G729A WARNING[311316]: File codec_gsm.c, Line 136

Modem connect speeds on calls through * seem to be lower than calls made through the telephone company lines or our old Rolm PBX. All data calls have 2 wire analog modems on both ends. For my set up I have channels of a Zhone channel bank tied to 2 modems. The Zhone channel bank interfaces my * server with a T400P card. modem --- Zhone Channel bank - * via T400P card - Zhone channel bank -

We were just testing forwarding one of our numbers to a VoIP CLEC and ran into an issue that we've seen before but never figured out the cause of. It seems if you call us and immediately hang up - or if the call is forwarded by Verizon (which stills rings about ? of one time) the Zhone detects the ring and answers the call.. The Zhone however DOESN"T detect that it's no longer

Hey guys, I've had a working Asterisk setup going for a while now, but am having problems with the Zhone Zplex 10b thinking that during ringing, an extension has answered the call when infact it hasn't. This only seems to happen on some of the ports and doesn't appear to be specific to the handset. Does anyone have any suggestions as to how I could go about fixing this (aside from

Hi, We need help in enabling g729a codec for our SIP peer that's using Cisco AS5300. Our codec is purchased from Digium. We are able to dial out the numbers and answer the call, but there's no audio. This is when only g729a is allowed. We noticed when they also allow ulaw codec on their side, the codec used falls back to ulaw and the problem is gone. -------------- next part

Good day We are going to add 6 channels of G729a to our asterisk server running iax between them I have a few question about the hole license thing. In iax.conf do i allow g729 or g729a?What's the difference? This license is for 2 servers,i.o.w 3 per server.How many calls does this give us? For example if server A calls server B does it uses 1 license,server A's license, or does it use