similar to: Discovery on packages

I am making an ajax call from js to call a method (assocboxchange) in my controller (AssociatesController), using XMLHttpRequest. I know the XMLHttpRequest works fine because I use it in other places with success. My problem is my URL I am using for this request doesn;t access the method in my controller which I (think) I am specifying. I am having it post to /channels/assocboxchange/" with

I have a form with contingent select boxes (the state is contingent on the country selected, so when the country selected changes, the state changes -- I am using the Carmen plugin for getting my state names and country names together, but not the functionality I am interested in achieving). Everything works fine except I cannot figure out how to amend my code such that: A. When the form is first

I have an int, ''floorfrozen,'' in my table, ''channels'' that can take a value of 0 or 1. When I render a page, an image in a link_to_remote is called differently depending what the value of channel.floorfrozen is, in a partial, as follows: <% if @channel.floorfrozen %> <%= link_to_remote image_tag("/images/icecube.gif"), {:url =>

Hi All, Rails code must be readable, formatted and well indented. I wanted to do indent code in my previous projects. Is there any built in rails plugin/gem that can help me to indent my existing code. Any ideas will be really appreciated ! NOTE: Please don''t suggest editor plugins. Thanks & Regards, Sandip R~ -- Ruby on Rails Developer http://sandip.sosblog.com

Dear All, I'm using Asterisk 1.4.16.2 with Zaptel 1.4.7 on Debian with kernel 2.6.18. I have two analog lines Zap/1 and Zap/2 as group 1 in zapata.conf. I'm using below context for dialing out. [outbound-local] exten => _9XXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten => _9XXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten => _9ZXXXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten =>

Hi, Urgently response would be wonderful, system is a Fedora Core 2. I have a Ericsson BP250 connected to 1 port on the TE405P and another connected to a local telco ISDN30. I have been running CVS-HEAD from about a 2 months ago and upgraded it again just in cause it was a version issue (didn't fix it) but this is what I am getting. When a person calls out from an extension on the BP250 to

Todd How do you have the context in the phones sip configs set? Bryant From: "Todd Reese" treese65 at gmail.com Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk

When i made internal call into my LAN using x-lite sip phone client I retrive in askterisk CLI : ----------- ERROR ---------- Verbosity is at least 6 -- Remote UNIX connection -- Executing Dial("SIP/201-979d", "SIP/201|60|t") in new stack -- Called 201 May 29 18:09:28 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call

Friends I am looking at Rcpp and I am a bit stuck on a simple matter. (I am calling R from c++, if there is a better way...) Given this simple example using the TTR package and the SMA function which returns a simple moving average.... Rcpp::NumericVector rv; for(int i = 0; i < 100; i++){ rv.push_back(rand()); } Rcpp::Environment TTR("package:TTR");

Hi:please i am working on fedora 31, then i installed R (language), and i want to install a R package ("DMwR"), but i get these warning messages ...: The downloaded source packages are in ?? ??/tmp/RtmpCBUyte/downloaded_packages? Updating HTML index of packages in '.Library' Making 'packages.html' ... done Warning messages: 1: In install.packages("DMwR") :

Hi, I want to fit moving average trend in R. In google, I see that it is in the package 'TTR'. But, I can't install this package. I have used the following code: >install.packages("TTR") But, it says there is no package called 'TTR'. Can you help me? Regards, Suman Dhara [[alternative HTML version deleted]]

Hi list, For months everything worked super here in our setup. This week I implemented some new idea in our webbased calendar system. I thought it would be nice to have an option that tells asterisk you are not available for calls during an appointment. For this to work I could no longer use the ringgroup setup: Dial(SIP/10&SIP/11&SIP/12,40,tr) So I thought, why not use the Local channel

Hi all. I was reading a sample config someone had posted relating to call forwarding, and in it, they use a Dial command with components that I cannot find any reference to. Can someone point me to a reference which could explain the difference between Dial(SIP/100|20|Ttr,,wW) and Dial(SIP/100,,wW) Specifically, what is the |20|Ttr ? I cannot seem to find any reference which would indicate

Dear all, I'm totally new to R. Recently I've been trying to use getYahooData in TTR package in order to download stock index daily open/high/low/close. The downloaded data is in the format of Open High Low Close Volume 2000-01-04 18937.45 19187.61 18937.45 19002.86 0 2000-01-05 19003.51 19003.51 18221.82 18542.55 0 2000-01-06

I installed the lattice package, and got an error that R was not able to remove the previous version of lattice. Now my installation seems to be currupt, even affecting other packages. I am getting this error when loading TTR: > library(TTR) Loading required package: xts Loading required package: zoo Error in loadNamespace(i, c(lib.loc, .libPaths())) : there is no package called

Greetings all, I am curious to know if either of these two sets of code is more efficient? Example1: ## t-test ## colA <- temp [ , j ] colB <- temp [ , k ] ttr <- t.test ( colA, colB, var.equal=TRUE) tt_pvalue [ i ] <- ttr$p.value or Example2: tt_pvalue [ i ] <- t.test ( temp[ , j ], temp[ , k ], var.equal=TRUE) ------------- I have three loops, i, j, k. One to test the all of

Hi, First a confession - The box in question is a 1.2.35 box, so this may be solved in a newer version as I know the JB code is all hugely changed, but... It may be worth checking into. Scenario: - IAX outbound call from Asterisk, which rings okay. - Remote end sends ANSWER, which we immediately ACK. - The ANSWER control packet gets put into the JB (that's how I read the code) - The remote

Hi all, I have this weird problem. I'm running asterisk 1.0.3 on Debian Sid (official debian package). We have 2 fritz ISDN cards. All is working great. Till I called the bank. It rings one time and then gives me the congestion tone. Here is what I see on the CLI (phone nr obfuscated for privacy reasons): -- Executing Dial("SCCP/michiel-00000004",

Rowland Penny a écrit : > On Thu, 4 Aug 2016 12:12:42 +0200 > JB <jb at eikeo.com> wrote: > >> Hello, >> >> I'm trying to migrate an old PDC controller running samba >> 3.0.4 to a more decent server. Now, I use samba 4.2.10 (from >> debian/jessie). >> >> My smb.conf is : >> >> # Global parameters >> [global] >>

> > 1. Do I need to activate jbenable in sip.conf? Or is it enough to call > > the JITTERBUFFER function? > > You only need to use the JITTERBUFFER function. > > The jbenable option will enable a jitter buffer on every channel > created for that peer (or, if global, for every peer in the system). > Depending on the version of Asterisk, it will also place the