search for: psv

Dear R-list, I try to transform a mathematica script to R. #######relevant part of the Mathematica script (* p_sv *) dd = NN (DsD - DD^2); lownum = NN (L-DD)^2; upnum = NN (H-DD)^2; low = lownum/(2s^2); up = upnum/(2s^2); psv = NIntegrate[1/(s^NN) Exp[-dd/(2s^2)] (Gamma[1/2,0,up] + Gamma[1/2,0,low]),{s,sL,sH}, MinRecursion->3]; PSV = psv/Sqrt[2NN]; Print["------------- Results ------------------------------------"]; Print[" "]; Print["p(sv|D_1D_2I) = const. ",N[PSV,6]]; #######...

...le script is at the end of the email (with initial values for easy reproduction) and can be pasted directly into R. The problematic part (which is included below of course) is <--- Original Mathematica ---> (* p_svbar *) UiA = Ni (Dsi - 2Di A + A^2)/2; UiiA = Nii (Dsii - 2Dii A + A^2)/2; psvbar = NIntegrate[1/(UiA^(Ni/2)) 1/(UiiA^(Nii/2)) Gamma[Ni/2,UiA/(sH^2),UiA/(sL^2)] Gamma[Nii/2,UiiA/(sH^2),UiiA/(sL^2)],{A,L,H}, MinRecursion->3]; PSVbar = psvbar/(4 Log[sH/sL]); Print["p(s?v|D_1D_2I) = const. ",N[PSVbar,6]]; </---> <--- translation to R ---> i...

Hi guys, How's things going ? Got a bit of a weird one here that I've been unable to solve. I have a Panasonic PBX linked to a Sirrix Quad BRI card that is running in TE (ptp) mode in a Asterisk box - this then links through Internet to another Asterisk box via IAX2. When a user on the Panasonic PBX system dials the extension of my Sirrix Asterisk box, Asterisk answers and says

...5.34*10^-7*(CO2-450)) >> ?TT <- *Temp*+273.15 >> >> #Saturation vapor pressure over ice >> # A1 <- -13.928169 >> # A2 <- 34.7078238 >> # Theta <- (*Temp*+273.15)/273.16 >> # Y ?<- ?A1 * (1 - Theta^(-1.5)) + A2 * (1 - Theta^(-1.25)) >> # Psv <- 611.657*exp(Y) >> >> ?#Saturation vapor pressure over water >> ?Tc ?<- ?647.096 ? ? ? # K ? : Temperature at the critical point >> ?Pc ?<- ?22.064 * 10^4 # hPa : Vapor pressure at the critical point >> ?nu ?<- ?(1-TT/Tc) >> ?a1 ?<- ?-7.85951783...

Hi, today i wrote a function in R of the type: index.refraction <- function(Temp,Press, RH, CO2) When i try to plug a negative number in Temp, i got this type of error: " n <- index.refraction(Temp= -40,100,80,CO2) Messages d'avis : 1: In Ops.ordered(left, right) : '-' is not meaningful for ordered factors 2: In Ops.factor(left, right) : - not meaningful for factors 3:

Hello, more and more operators in Europe offer music instead of ring tunes. E.g. instead of the 400 Hz or whatever tunes, the caller will hear J-Lo, or Mozart.... Currently I will have to answer the line to do that. Is there a way to do this with asterisk? Regards, Marc -- CTO Marc Storck MS Networks SA mstorck@luxadmin.org Internet Service

I am using a HFC-S card in nt mode with zaphfc driver to connect an internal isdn bus. I would like to signal an incoming call on, let's say, 4 phones. Right now I use: Dial(Zap/g1/21&Zap/g1/22&Zap/g1/24&Zap/g1/23&Zap/g1/29,,t) where g1 are my two isdn channels provided by HFC-S card an the 21,22,etc my internal numbers. When the command is executed however, only the

Hello All, I saw some coverage of this in the list archive but no one seems to have posted a resolution. I am using Asterisk@Home 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR. >From there the call is routed to groups based upon input. However, there is no ringback indicated to the IAX caller. Does anyone know how to resolve this problem? Thanks, Wiley

Here is the setup: POTS -> PRI -> Asterisk -> ATA (Fax) The ATA is set to only 711. Asterisk's sip.conf sets this device to only 711. Yet, faxing works less than 50% of the time. I cannot possibly be the only person with this problem. I can't even get app_rxfax to work. It answers but rxfax drops the call 5 seconds into it. If anyone out there has near 100% success with

Folks, A nice new data resource has come up -- http://data.un.org/ I thought it would be wonderful to setup an R function like tseries::get.hist.quote() which would be able to pull in some or all of this data. I walked around a bit of it and I'm not able to map the resources to predictable URLs which can then be wget. There's some javascript going on that I'm not understanding.

...ntense debug" on first the internal (pbx) span and then the externa (pstn) doing the same call both times. The call is initiated from span 2 (the pbx), goes out span 1, comes back in span 1 on another channel and is finally forwarded back to span 2. The log is available from http://www.psv.nu/jbl/problemlog.txt Asterisk receives a CONNECT (line 1427 in the log) for the outgoing leg on the pstn span. That connect was originally sent when the incoming leg to the pbx answered. It does not get delivered to the originating span (the pbx). It sees neither the call proceeding, alerting...

hi all, i am trying to get my TE410P (see previous posts) working in Austria (telekom Austria - i am still waiting for an answer for my questions). my /etc/zaptel.conf looks like -------------------------------- span=1,1,0,ccs,hdb3,crc4,yellow span=2,2,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 loadzone=at defaultzone=at -------------------------------- after

Is there a soft phone for PocketPC or iPaq? If not, is someone working on it? I will be more than willing to contribute my mite if needed. Thanks, -- sudhir

...Sam Lex Lethol <lethol@gmail.com> wrote on 23/09/2004 15:31:39: > I tried the xten one and didn;t like at all.. > > Havent tried to SJPhone, but my guess is that it has better support. > > Lethol > > On Thu, 23 Sep 2004 08:13:10 +0200 (CEST), Peter Svensson > <psvasterisk@psv.nu> wrote: > > On 22 Sep 2004, Sudhir Kumar wrote: > > > > > Is there a soft phone for PocketPC or iPaq? If not, is someone working > > > on it? I will be more than willing to contribute my mite if needed. > > > > Xten has a product, possi...

-----Original Message----- From: Peter Svensson [mailto:psvasterisk@psv.nu] Sent: 25 November 2004 10:54 >> They work just fine if your pstn provider is at all serious. If not, >> switch. They don't belong in the pstn business anyway. BT so you probably have a point. To be fair to BT I'm not saying that the ISDN does drop during pow...

I am have same issue with PRI and overlap dialling is not enabled. Stuart -----Original Message----- From: "Peter Svensson"<psvasterisk@psv.nu> Sent: 01/02/05 16:55:52 To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] Zap channel occasionally misses dialing thefirst digit On Tue, 1 Feb 2005, Andrew Kohlsmith wrote...

Hi, I wonder which PRI interface card is most stable and supported for EuroISDN and Asterisk ? Are they stable enough ? Any tips ? Thanks in advance, regards, Rob.

Hi! I have a runing asterisk box and i want to dial to a analog. pbx using a 4FXS Welltech. Let's say that my pbx have no. 700. If i want to dial to a person in that direction i have to dial pbx prefix (ex. 700), wait for pbx to ansear with hello message and after that to dial internal number(ex. 101). It is possible to dial directly 700101 and asterisk to dial PBX prefix, wait for PBX to

...w do I debug the Zap channels ? You need to set up debugging in the corresponding conf file for Asterisk. Debugging of the PRI signalling is then set up with pri debug span ??? or pri intense debug span ??? Peter -- Peter Svensson ! Pgp key available by finger, fingerprint: <petersv@psv.nu> ! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF ------------------------------------------------------------------------ Remember, Luke, your source will be with you... always...

Hi All, I can get Grandstream 100 SIP phones for EUR 75. I'm not sure about the pricing for these in Europe, so I'd like to hear from people here whether that is a reasonable price for them? TIA & BRgds -- Francesco Peeters ---- GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit