similar to: Monitor a particular SIP call for training purposes

Hello, There are 3 PRI's connected to the card each from different operators. Especially echo occured on span 3 is really annoying. Configuration files are as follows. Is there something wrong in conf ? Zapata.conf -------------------------- [channels] context=default switchtype=euroisdn usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes

Do you have any extension in default context of your extensions.conf file to accept incoming calls ? It must be something like; exten => 12345678,1,Answer() exten => 12345678,2,Playback(Welcome) ... 12345678 = The DID number you are calling to reach E1 Idris -----Original Message----- From: Erick Perez [mailto:eaperezh at gmail.com] Sent: Thursday, July 26, 2007 7:03 AM To:

Hi, I'm planning on migrating someone who uses a very mature system. They would be logging in either as AgentLogin() or AQM. The main requirement however, is: The supervisor will have a control panel, where he will see how many of his agents are on call. If they are, he can "right-click" on the agent and get the options Call Monitor (where the super just listens in on the call,

I have a dataframe df with two columns x and y. I want to count the number of times a unique x, y combination occurs. For example x<- c(1,2,3,4,5,1,2,3,4) y<- c(1,2,3,4,5,1,2,4,1) df<-as.data.frame(cbind(x, y)) #what is the correct way to use ddply for this example? ddply(df, c('x','y', summarize, ??) #desired output -- format and order doesn't matter # (x, y)

Phil, It sounds like your carrier is just using a channel bank to split off six of your E1 channels into analog FXO ports. You will want this for fax lines, security systems, and dialup connections since doing this over VoIP/Asterisk can be problematic. It is best to keep these lines out of the PBX. I have done several implementations on IBM x305 and x306 servers and they work great. I

Sample code from *R CookBook* (awesome book btw) *11.12: Finding the Best Power Transformation (Box-Cox) Procedure* require(MASS) x <- 10:100 eps <- rnorm(length(x), sd = 5) y <- (x + eps)^(-1 / 1.5) m <- lm(y ~ x) # ***************** What does the *which* in this line do??? ******************************************** plot(m, which = 1) [[alternative HTML version deleted]]

Hi List, Not sure if this is the place for this so here goes ... We have a number of Polycom 501's connected to our * box and they work great. Some of our users have added a few entries into the directory on the phone. The problem is on those particular phones they now sometimes get "resource full" on the phone when accessing the directory. No central directory was configured.

I have a Linksys PAP2 connected to Asterisk. Have one of the FXS ports working fine. I am unable to get the other to work. Does anybody have an example configuration to make both work. Both are registering fine but there's just no dialtone on the non working port. TIA

Is it possible to give a caller three goes at an extension number then hangup? exten => s,1,Zapateller(answer|nocallerid) exten => s,2,PrivacyManager exten => s,3,Ringing(1) exten => s,4,NoOp(${CALLERID}) exten => s,5,SetMusicOnHold(random) exten => s,6,Background(silence/1) exten => s,7,Background(thank-you-for-calling) exten => s,8,Background(silence/1) exten =>

Maybe not the place for this but thought I'd post the info for others. I purchased a cisco 7960 off ebay and needed to convert to SIP for *. I know * supports SCCP but I wont go into that here. I'd read on voip-info.org that a contract could be purchased for approx $8 to allow me to download the firmware. I though, being in the UK, i'd get one through a reseller in the UK.

# I have a dataframe in the following form: track <- c(rep('A', 3), rep('B', 4), rep('C', 4)) value <- c(0.15, 0.25, 0.35, 0.05, 0.99, 0.32, 0.13, 0.80, 0.75, 0.60, 0.44) df <- data.frame(track=factor(track), value=value) #> print(df) #track value #1 A 0.15 #2 A 0.25 #3 A 0.35 #4 B 0.05 #5 B 0.99 #6 B 0.32 #7 B 0.13

I'm curious what asterisk solutions there are out there for inbound call centers with multiple DIDs. I'm looking for solutions for a setup where single system may have 1000 DIDs going to it, one for each account. Each account may not get that many calls. Solutions that will all reporting on calls coming into different accounts, call routing for queues based on distribution groups. Like

I am trying to get the Vim-R-Plugin<http://www.vim.org/scripts/script.php?script_id=2628> to work with gvim and R on Windows 7. When I open a .R file in VIM, it complains and says ""Python interface must be enabled to run Vim-R-Plugin." I have installed pywin32 for python 2.7, and added the following 4 lines to my _vimrc per the instructions

Hi, I'm currently looking to run Asterisk in the office to replace an old PBX and would appreciate a little help. We are moving offices and will have 8 digital lines. My questions are: As there are 8 digital lines is this known as PRI? Which Digium card would be the best fit? Would you recommend looking at the echo cancellation cards? We are UK based: is caller id supported by Asterisk

I have a dataframe df with columns x, y, and height. I want to create a heatmap-like plot that creates a grid of x by y, and then color codes the grid depending on the value of height. Is there a ggplot2 object to do this? I'm able to easily do this in Excel with pivot tables and conditional formatting so I'm including an image that is close to the output I want. I want to be able to

Trying to find my feet here. If I wanted to connect Asterisk to a PRI and throw in a T1 Adtran channel bank into the mix for fax machines would the following work? Connect PRI line from telco to Port 1 on the Digium Wildcard TE205P. Connect Adtran TA-624-T1 to Post 2 on the Digium Wildcard TE205P also, would I need a crossover to the channelbank or is it a patch lead like the connection to the

Hello, My primary computer is a small netbook. At home I have lot of mostly very old computers. I'm thinking about creating some cluster (OpenMosix based) or to use the fastest machine only for video + audio encoding (upload from netbook to cluster or server and do encoding there) Could you share with me, if you are using some of these approaches already and how? - encoding on more

Does anyone know if version 3.1 is Call Manager or SIP? Thanks.

I'm working on an LLVM backend for Idris, a garbage-collected pure functional programming language, and have experienced some frustration that LLVM's GC support, specifically with regard to mapping roots, operates only on allocas. This entails a lot of otherwise unnecessary stack allocation (especially in a pure language, where in-place mutation is rare) and imposes limitations on what

Phil I have very good experience with the vegasteam ATA's devices.(you might also want to look @ sipura ATA's, since vegastream is doing an oem on there boxes) They support modem until v.90 speeds and faxes for g3. They are expensive, and again, work great and configure very easy joash ________________________________ From: asterisk-users-bounces@lists.digium.com