similar to: Mysql Support int Asterik-11

Hi all, I am trying to interparete the result of the following output from lm; fit1 =lm(Feed _Intake ~ weight + season + weight*season) Season has three classes(x,y,z) Reults are Estimate (Intercept) 21.51559 weight 2.13051 factor(season)y 10.59739 factor(season)z 1.30421 weight:factor(season)y

Dear R-experts, I am running a script that has the following structure: windows(height=5.5,width=8) dat<-read.csv("myfile.csv") names(dat)<-c('a','b','c','d') dat<-dat[,1:4] xyplot(dat$a~dat$b) Then I usually save the plot as a PDF (from the menu in the R console). I can save the PDF twice in a row; but at the third time, right when I

Dear all anyone have idea about connect asterisk with mediant 2000 audiocode configuration ... anybody have configuration about it --------------------------------- Get your own web address. Have a HUGE year through Yahoo! Small Business. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List; All we know that in voice, there are a type of communications between endpoints, for example: in some communications we do a proxy for media and signaling while other communications we do a proxy for only signaling. Where I can determine these things in Asterisk if I am using SIP and if I am using H.323? Regards -------------- IP Telephony and Contact Center Engineer Eng. Bilal Ghayad

Hello All. I am having some trouble with call transfers when asterisk is the 2nd party called and I hope to benefit from your experience. I want to use asterisk for call park/pickup and have configured openser to relay calls made to ruri 700-720 to asterisk running on localhost:5069 Call flow: phone A calls phone B (both phones are polycom) Phone B answers then phone b

Dear Alex; Thanks for your kindly reply. Please explain for me what do u mean exactly in "a la" in the following sentence u wrote it below? " in SIP, this can be done via "re-INVITEs" a la the canreinvite= option for SIP peers in sip.conf" Another thing, do u mean that it is easier (better) if we need H.323 endpoint to talk with SIP endpoint then we use full

Dear R helpers, I am working with R 2.4.1 GUI 1.18 (4038) for MacOSX. I have a matrix of 10 000 genes and try to run the following commands: > model.mix<-makeModel (data=data, formula=~Dye+Array+Sample+Time, random=~Array+Sample) > anova.mix<-fitmaanova (data, model.mix) > test.mix<-matest (data, model=model.mix, term="Time", n.perm=100, test.method=c(1,0,1,1))

Hello, What is the recommended way to offer high availability to Guests? For example, let host1, host2 and host3 be libvirt and KVM/Qemu enabled. If I start a guest in host1, how can I guarantee that it will stay online if host1 goes down? GlusterFS, for example, would take care of storage; but what about CPU and RAM? How can this be accomplished? Thank you for any feedback or comment in

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040112/e9dceeb4/attachment.htm -------------- next part -------------- Hello One of the option in VoicemailMain() is "Adavance Options". Could anyone explain what are these ?. Because whenever I select Advance Options, it repeats the same process of asking "Change Folders,Advance

Hello everybody, I would like to know whether it is possible to run Dial and MeetMe commands simultaneoously on the same channel. I am using a C AGI as below but it seems to me that only the first command that is called in the agi is executed. ........... // Pr?paration de la commande pour l'appel du client fprintf(stderr,"%s%s",numtocall," is the number to

Hi, We have got a new PRI card at one of our Office locations and now I need to install the the device on a remote server. Is there any way to know if the device is loaded already. When I give " cat /proc/zaptel/* " it returns the following. # cat /proc/zaptel/* Span 1: WCT1/0 "Wildcard TE122 Card 0" (MASTER) B8ZS/ESF RED IRQ misses: 2 1 WCT1/0/1

Hi all, I recently put a new version of my french introduction to R online. It is more specifically targeted at social sciences students and researchers, but could be interesting for beginners who are not really familiar with statistics and coding. The document is available (in french) in PDF, as well as the Sweave source code, from the following page : http://alea.fr.eu.org/j/intro_R.html

Hi folks, I have a problem, shame on me I feel as newbie, I cannot open Firefox though ssh. ssh server is 7.2 and ssh client (running Xorg) is Fedora 23. Firefox does not open. I've tried a lot of methods, even adding xauth cookie, and for example: firefox --no-remote --no-xshm debug3: receive packet: type 90 debug1: client_input_channel_open: ctype x11 rchan 2 win 65536 max 16384 debug1:

On Jul 24, 2013 2:52 AM, "Jacob Carlborg" <doob at me.com> wrote: > > Do your LLVM development on Mac OS X :) Should work well. Apple is one of the bigger supporters of LLVM, so I'd hope OS X would be a suitable dev platform. > It depends on what your needs are. Using VirtualBox will probably be the easiest. It also allows you to run both Mac OS X and Ubuntu

Dear R - help, I am working on the Credit scorecard model. I am using the Logistic regression to arrive at the regression coefficients model. I want to use the Hosmer - Lemeshow test . In order to understand the use of R - language, I had referred the following URL       http://www.stat.sc.edu/~hitchcock/diseaseoutbreakRexample704.txt The related data 'diseaseoutbreak' is available

Hi, I am using ubuntu-12.04 and installed asterisk from repository (apt-get install asterisk). I have configured it to work with odbc, *CLI> odbc show ODBC DSN Settings ----------------- Name: asterisk DSN: asterisk-connector Last connection attempt: 1970-01-01 05:30:00 Pooled: No Connected: Yes But it still show me the following error [Jul 31 12:36:18]

Hi, is it possible to use Asteriks for translating SIP to H323 and vice versa? I am looking to implement the following Setup SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC Basicly i want SIP fones to talk to H323 fones and and SIP Fones to access PSTN Gateway(s) in a H323 network. Anyone got something similiar running? Any ideas? best regards,

I am newest to this group and would appreciate your help! Is it possible to use quicknet phone jack with asteriks@home ver 0.6? Little has been mentioned about use of quicknet products' adaptability with asteriks@home I do have a couple of old jacks to startup right away. Your guide is most welcome. Thanks, Mike __________________________________ Celebrate Yahoo!'s 10th

We are running Asterisk 1.4 and need some help to determine how (if) * supports 3 party warm transfers. I've searched quite a bit and all I can find is information on "attended transfers". What we are looking for is: (1) external inbound call A comes to * extension B, caller A is placed on hold and extension B calls external third party C. After explaining caller A issue to

Hi all, Long story short, I'd like to configure a folder containing: INBOX (some other folders) -> unseen emails from these folders + all other emails (also seen) from the same thread -> seen emails from these folders, but only the recent ones + all other emails from the same thread I tried: # inthread refs (or (or recent unseen) recent ) inthread refs unseen # inthread