similar to: Fw: " PROBLEM SOLVED" - Error with function

Dear R forum, Please refer to my query regarding "Error with function". I forgot to mention that I am using "plyr" library. Sorry for inconvenience. Regards Katherine --- On Tue, 23/4/13, Katherine Gobin <katherine_gobin@yahoo.com> wrote: From: Katherine Gobin <katherine_gobin@yahoo.com> Subject: [R] Error with function To: r-help@r-project.org Date:

Dear R forum, I have a data.frame as given below: df = data.frame(tran = c("tran1", "tran2", "tran3", "tran4"), tenor = c("2w", "1m", "7m", "3m"))  Also, I define libor_tenor_labels = as.character(c("o_n", "1w", "2w", "1m", "2m", "3m", "4m",

Hello, Maybe some of you tried the SIP support recently introduced by Quintum in their AS devices. I have one Asterisk machine connected to PSTN via E1. It works properly. On the other side I got an ADSL line, with NAT and few devices behind it, like computer with X-Lite client installed or mentioned Quintum device. It works great - calls initiated from there are OK, as well as PSTN originated

Hello *'s, I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone integrate it with asterisk if anyone what is the scenerio i have scenerio which is quite simple but i am confused about it whether it is possible or not : I integrate it with asterisk for interanet no PSTN at all just only IPphones and analog phones connected on FXS port.Is it's neccassary to cannect with

Hello *'s, I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone integrate it with asterisk if anyone what is the scenerio? i have a scenerio which is quite simple but i am confused about it whether it is possible or not : I integrate it with asterisk for intranet no PSTN at all just only IPphones connected through ehternet port and analog phones connected on FXS port.Is

Dear R forum I have data.frame as df = data.frame(rate_name = c("USD_1w", "USD_1w", "USD_1w", "USD_1w", "USD_1m", "USD_1m", "USD_1m", "USD_1m", "USD_2m", "USD_2m", "USD_2m", "USD_2m",  "GBP_1w", "GBP_1w", "GBP_1w", "GBP_1w",

Hi, I'm having problems to register Quintum Tenor DX 3020 on a Asterisk box with SIP. Asterisk always returns "Username/Password mismatch". I've tried all configurations that was on the Quintum's manual, but no success. I've tested the same username and password with a Linksys (PAP2-NA) with the same asterisk box, and it worked fine. Where is the problem ?

Hello all, I post this issue thinking too that could help other people on an asterisk deployment over distributed offices considering both quality, prices, devices and so. Well, i am working on a deployment of a telephony system based in asterisk. My company have a central office with seven remote offices connected all through a VPN. To reduce and evaluate costs i consider solutions like:

Dear Katherine, For your first question: If you are creating these files in a specific folder, then list.files() #[1] "Individual_Present_Value_BONDS.csv"? "Individual_Present_Value_Equity.csv" #[3] "Individual_Present_Value_FOREX.csv" gives you which files are present.? But, suppose you have other files too in the folder and you want to check only the above

Hi Guys, Hope fully somebody out there will have experienced this and can shed some light on how it was overcome. Current setup includes asterisk 1.6.2.11, GNU GK and a Quintum Tenor CMS on the same lan. Earlier I was unable to make a sip call from the CMS back to a sip client registered on my asterisk box. So I moved onto passing the call from the Quintum CMS to a Quintum Tenore DX which is also

Dear R forum, (Pl note this is not a finance problem) I have two data.frames as currency_df = data.frame(current_date = c("3/4/2013", "3/4/2013", "3/4/2013", "3/4/2013"), issue_date = c("27/11/2012", "9/12/2012", "14/01/2013", "28/02/2013"), maturity_date = c("27/04/2013", "3/5/2013",

I am introducing the scan() function to my class. Consider the following file (Scanexamp.Rnw ) \documentclass[12pt]{article} \begin{document} <<>>= height = scan() 64 62 66 65 62 69 72 72 70 part = scan(what = character(0)) "Soprano" "Soprano" "Soprano" "Alto" "Alto" "Tenor" "Tenor" "Bass"

Hallo zusammen, dann will ich mal wieder ein bi?chen Schwung auf die RoR-Liste bringen: Schildert doch einfach mal eure Erfahrungen mit Tools rund um Rails, also zum Beispiel Capistrano (formerly known as Switchtower), RadRails etc. Welche Plugins setzt ihr ein, von welchen w?rdet ihr abraten? Oder folgt ihr dem Tenor "Reuse is overrated" und schreibt euch die Sachen lieber neu? --

Hi, just wondering if there is anyone that can help me configure my quintum box to operate with asterisk. i have tried and made numerous attemtps configuring the tenor to work with asterisk@home but have been unlucky. anyone out there has a cheat sheet to configure this device. thanks.. for some reason i cannot get it to work. your help is appreciated.

This looks like the same case as the thread: Problem ATA-711-723-Oh323-Asterisk that was posted back in August. Although, given the nature of trampled memory problems it could be almost anything. The Oh323 library dies in a call to realloc. This is almost certainly indicative of someone stomping on memory somewhere previously, or else a memory bug in the pwlib library itself. (Which I find

Hi all, I am planning to use VoIP gateways to connect remote offices to Asterisk. Not having much experience with these and Asterisk I would appreciate any info/experience that you might share with me as to their interoperability with Asterisk. I am interested with in rather bigger gateways (order of E1's) from: AudioCodes - Mediant Mediatrix - 1531 Quintum tenor Multupath D3000 Has anyone

I just upgrade quintum A800 with new SIP firmware ---------- Product Name: Tenor Analog A800 Multipath Switch - 8 ports (Rev. B) Gatekeeper Status: Mini GK Calls Allowed: 8 Feature Bit Status: -PS/+RB/-ER Languages allowed: 1 Serial Number: A002-00308F Ethernet Address: 00-30-E1-00-30-8F IP Address: 10.101.0.10 Subnet Mask: 255.255.255.0 Default Gateway: 10.101.0.1 System Software Version:

Hi all, Im trying to build an small home system. I have 2 pots lines, and i need to make 8 extensions and be able to use my old analog phones. What would you recommend to use as the 8 FXS switch? I saw some equipment from quintum, they have a Tenor AS that offer 4 FXS ports. But i don't know if it is the best solution. Does anyone have a better solution to build this system? If an analog

Hi, I am looking at a project which requires VoIP QoS monitoring and failover re-routing to PSTN, without dropping the call ideally. While I have hardware solutions available such as the Quintum Tenor series, I see no reason why Asterisk can't have this feature with some effort obviously. Plus I note it is on the wish list at http://www.voip-info.org/wiki/view/Asterisk+Wishlist (last item