similar to: Asterisk 'losing' upstream provider registration state during small network outages.

I'm going to try and ask this again and keep it short and as too the point as I can while still providing enough info to be of use. PLEASE advise if I am going about this wrong or asking too much. I'm seriously doing my BEST to throughly read the docs and try a bunch of things BEFORE coming here to ask and possibly annoy. If is documentation that explains thsi process in terms that

Steve, 1) go to /etc/asterisk 2) open modules.conf for editing using vi 3) add this line: noload=pbx_wilcalu.so 4) Save the file 5) Restart asterisk Lightup the candles, open the Cabernet Savignon ( or whatever your prefernce) and call your girlfriend. ;) Seshu -----Original Message----- From: asterisk-users-bounces@lists.digium.com

I have a problem with R. I try to compute the confidence interval for my df. When I want to create the plot I have this problem: Error in xy.coords(x, y, xlabel, ylabel, log) : 'x' and 'y' lengths differ. I try this code: library(dplR) df.rwi <- detrend(rwl = df, method = "Spline",nyrs=NULL) write.table(df.rwi,file="rwi.txt",quote=FALSE,row.names=TRUE)

aa <- (structure(list(X.0.85 = c(-1.02, -1.17, -1.29, -1.39, -1.46, -1.5, -1.52, -1.5, -1.46, -1.39, -1.3, -1.19, -1.07, -0.93, -0.79, -0.65, -0.5, -0.36, -0.22, -0.08, 0.05, 0.18, 0.3, 0.41, 0.52, 0.62, 0.72, 0.81, 0.89, 0.98, 1.05, 1.13, 1.19, 1.25, 1.29, 1.31, 1.31, 1.29, 1.24, 1.16, 1.06, 0.93, 0.77, 0.58, 0.38, 0.16, -0.07, -0.31, -0.89, -1.05, -1.19, -1.31, -1.41, -1.47, -1.51, -1.51,

I am trying to get Asterisks to connect to our StanaPhone so that I can use it to route my outgoing PSTN calls to. We have a free account and if I can get this working are willing to pay for an actual minutes with them. Here is what I have in my sip.conf: [stanaphone] type=friend secret=pAsSwOrD ; skewed for this message. username=3475341914 host=sip.stanaphone.com

Hi r-users,   I would like to extract the data that match.  Attached is my data: I'm interested in matchind the value in column 'intg' with value in column 'rand_no' > cbind(z=z,intg=dd,rand_no = rr)             z  intg rand_no    [1,]  0.00 0.000   0.001    [2,]  0.01 0.000   0.002    [3,]  0.02 0.000   0.002    [4,]  0.03 0.000   0.003    [5,]  0.04 0.000   0.003    [6,] 

For some reason I now seem unable to control which codec is chosen. The problem happens with outgoing calls to Stanaphone. Even if I chose disallow=all and allow=ulaw as the only codecs it connects with GSM. Has anyone else got problems with these settings? Any suggestions? As I recalled it, such a setup would not establish a call if the ulaw-codec was not offered by the provider. Stanaphone has

Sorry if this comes thru twice, I had the wrong account selected to send the first time... Callers to the number get ringing, I get stuff in my asterisk console, and it calls my softphone and ata, but answering either gets silence, and the caller gets the ringing stop, if they wait ages they get the stanaphone voicemail. I have had the account for ages, and it never has worked, other sip

I have been running Asterisk for years on a machine with a public IP. Most recently, I have been running 1.2.17, from the day it came out, with no (noticeable) problems. Yesterday, I switched over to a new server that is on the same public subnet, one higher than the original server. I built 1.2.17 from source on that machine (as I did on the old server). My firewall on the new machine is

I see that stanaphone is not accepting new customers. Does anyone know if they are doing ok? I have a number with them and would like to start redirection people before it gets canceled on me if they are having trouble.... thanks Todd

Hello, After much searching of voip-info.org and google, I'm finally giving in and asking the list. The setup I have is this:- Single X100P card in a Debian system Inbound/Outbound POTS line connects to the X100P Sipura 2000 and Budgetone 100 on the LAN 1 Cordless and one conventional phone connected to the sipura Account on Stanaphone.com for eitherbound SIP calls. (I have other SIP

Hi, I am pretty new to all of this but was able to set up an asterisk server and have been able to succesfully connect to asterisk with x-lite as sip client. I have also connected asterisk to FWD (using iax2) and to voipjet (also using iax2). Now I am trying to connect asterisk to Stanaphone. It has to register as a SIP client but I am not being succesful at all. My asterisk server sits behind a

I've a matrix like this: 1985 1.38 1.27 1.84 2.10 0.59 3.47 1986 1.05 1.13 1.21 1.54 0.21 2.14 1987 1.33 1.21 1.77 1.44 0.27 2.85 1988 1.86 1.06 2.33 2.14 0.55 1.40 1989 2.10 0.65 2.74 2.43 1.19 1.45 1990 1.55 0.00 1.59 1.94 0.99 2.14 1991 0.92

I have written a wrapper for heatmap.2 called heatmap.w.row.and.col.clust which auto-generates breaks using breaks<-round((c(seq(from=(-20 * stddev), to=(20 * stddev))))/20, digits = 2) #(stddev in this case = 2.5) This has always worked well in the past but now I am getting an error that non-finite breaks are being generated. Drilling down, it seems that my wrapper is generating finite

I WROTE THIS FUNCTION BELOW test <- function(x, ...) UseMethod('test', x) test.data.frame = function(x, model, which, error, ...) { av <- aov(formula(model), data = x) res <- test.aovlist(av, which = which, error = error) return(res) } test.aovlist <- function(x, which, error, ...) { mm <- model.tables(x, "means") tabs <- mm$tables[-1]

Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061207/92c7f946/attachment.htm

Anyone having problems with Stanaphone registration today? I'm getting the following.. Nov 25 11:35:58 NOTICE[229390]: chan_sip.c:4053 sip_reg_timeout: Registration for 'xxxxxxxxxx@216.128.82.18' timed out, trying again -- Got SIP response 500 "Internal Server Error" back from 216.128.82.18

I have data as below.Please let me know how the ACF and Pacf used to determine the order od arima model. Is there any rules need to be followed to determine order.Please advise > turkey.price.ts Jan Feb Mar Apr May Jun Jul Aug Sep Oct Nov Dec 2001 1.58 1.75 1.63 1.45 1.56 2.07 1.81 1.74 1.54 1.45 0.57 1.15 2002 1.50 1.66 1.34 1.67 1.81 1.60 1.70 1.87 1.47 1.59 0.74 0.82

Hello, is there already a function in any R package which does source code formatting of deparsed lists? Let's create the following list: x <- list(a = round(rnorm(3), 2), b = round(rnorm(3), 2)) xx <-c(aa = round(rnorm(30)), f = function(a) a + b, list(x, x)) Now, I want deparse it in a way that yields something like: list( aa = c(0.25, 0.18, 0.84, -1.25, 0.09,

Hi All, I have been reading through the list quite a bit, and I am going to post this more as a poll than anything else. I am working on setting up a very small business with something that resembles a professional voice system. My idea is to use Asterisk with a SIP provider and SIP clients. I currently have a Vonage account already. So adding the 9.99 a month Soft Phone would be easy.