similar to: pre-newbie - some basic questions...

Evening, I've got * asterisk up and running with nufone account for inbound and outbound calls from the world... everything works quite nicely except that I noticed last night that when I have a call active from * to nufone, web surfing on the network slows to a crawl... additionally, I have a VPN tunnel to another office (using CIPE) and while inter-office (i.e. - SIP to SIP) calls

Hello All, Wondering what sort of real world mileage people are getting out of different internet connecions - i.e. different DSL connection speeds, cable modems, etc... Is it reasonable to hope to carry 10 - 15 concurrent calls on a 768K DSL? I'm not talking about theoretical BW or looking for any difinitive absolute guarantee... With DSL and Cable - there is no guarantee, so I'm

I'm attempting to install * for the first time and encountering some problems... I'm running an old 166MMX processor with 50megs RAM... SuSE 8.1 distro... when I make clean ; make install in the /usr/src/asterisk dir, I get a "error: termcap support not found". the termcap libraries are installed. Anyone run into this before? Thanks, dave redmore

Has anybody experienced problems with a DSL modem stopping all other traffic when an IAX channel opens up? I've really traced this thing out as best I can - it isn't a lack of bandwidth on the DSL connection (I've had three concurrent Ulaw calls coming in at once with no problems), nor a problem within the network (other intranetwork traffic isn't affected by the call) - the

want to get from SIP to ISDN or from SIP to SIP. I only have a ADSL connection that means 786kb/s downstream and 128kb/s upstream so i can max handle 2 sip calls at once. I want to have 25 Accounts because of the different numbers for the different phones. Good, i wanted to buy a ready built pc from ebay but now i think i will built a 19" rack case so i will built in a 1 - 1,5ghz intel

I am using Audacity to record some voice prompts. The .wav files I'm producing are of stellar quality. However, once I turn them into .gsm, they sound buzzy and muffled. I know that some of this comes with the territory, but I wonder if there is anyone out there who does this routinely, and who can advise me as to the MO I could use that results in the highest quality in the resulting

I've been trying to draw quantile linear regression lines across a scatterplot of my data using attach(forrq) plot(PREGNANT,DAY8,xlab="pregnant EPDS",ylab="postnatal EPDS",cex=.5) taus <- c(.05,.1,.25,.75,.9,.95) xx <- seq(min(PREGNANT),max(PREGNANT),100) for(tau in taus){ f <- coef(rq(DAY8~PREGNANT,tau=tau)) yy <-

Hi there, >library(faraway) >pima pregnant glucose diastolic triceps insulin bmi diabetes age test 1 6 148 72 35 0 33.6 0.627 50 1 2 1 85 66 29 0 26.6 0.351 31 0 >data(pima) >pima pregnant glucose diastolic triceps insulin bmi diabetes age test 1 6 148 72 35 0 33.6

Hello, I am having trouble with creating a simple bar chart using R. My file just consists of 1 column called reason. Within this column, there are 2 responses(Room for Improvement and Info entered is all relevant). My code for creating the charts is as follows: d <-read.table("C://project/graphs/reason.csv", sep=",", header=TRUE) pie(d$reason, labels=d$reason,

Hi all, This is my first post here - I started with Asterisk a few days ago and have "fallen in love" - fantastic product. I've only got softphones connected at the moment - I'll probably order the FXO/FXS cards in about a month (and then think about getting some hardware SIP phones). Our current phone system is quite a few years old and isn't growing with us (when a single

Hi, I just got a SPA-2000 in and was finally able to complete my asterisk setup. I'm making my outgoing calls through iconnecthere from the asterisk server however I'm running into a problem when placing calls. I can connect fine but when the person (or answering machine) picks up I hear them talk for a about half a second then there is a half a second pause or muted period and then the

All, in 1.3.26 F:NV won't go into the game after you select new game or continue. The circle "loading" screen appears but then then stops and that's it. There are no errors in the console as well beyond what you normally get in 1.3.24. So before I run some regression tests if I get the time I was hoping to find out the debug channels I should load to see what's going on?

all, i believe. GLSL is disabled by default on Nvidia GPUs.

Is it possible to send the CallerId to IconnectHere with Asterisk when making outbound calls? I read somewhere that it doesn't work. I have set up everything to send the correct CallerId info to IconnectHere but I get a "442-887-926267" caller id. In [globals] ICONNECT1=1713...(my number) MYNAME=My Name I set up the Caller Id in the dialing macro: [macro-iconnecthere] exten =>

Hi, during the past weeks i tried to set up a network through a cipe tunnel, building a vpn to some road warriers. With few words, the tunnel is etablished, and at the tcp/ip level everything works fine. NET A : 192.168.0.0/26 NET B : 192.168.102.0/26 trans-net : 192.168.101.0/26 in ascii-art, it looks like this: eth0:192.168.0.5 +-----------+ cipcb0:192.168.101.1

Hello everyone, Last week the CVS version of tinc gained the ability to run under Windows, in a Cygwin environment (http://www.cygwin.com), making use of the virtual network driver from the CIPE project (http://cipe-win32.sourceforge.net/). Preparations were also being made to support Windows without Cygwin. Today James Yonan of OpenVPN (http://openvpn.sourceforge.net/) anounced a port of

For the past several days I can no longer use iconnecthere with asterisk. It is broken in BOTH directions; I can neither make nor receive calls. On outbound calls I get an immediate error: -- Got SIP response 400 "Bad or Missing To" back from 213.137.73.140 On incoming calls, the call switches through OK, and for a few seconds I get audio in both directions, although much

I am going to have to find a fix for this problem or I'm going to have to quit using iconnect. About one call in 10 or so, iconnect's gateway gives me an error (console output appended below). So upon receiving the error, which as a 4XX error means, "Fatal," asterisk gives up and drops the call. But not iconnect!! The phone at the other end starts ringing, and rings

Hi all, I'm working on a setup for a small office. I'd like to use SIP/iconnecthere most of the time, because they're cheap. But they only allow a single call. When the single iconnecthere line is in use, I'd like to use IAX2/voicepulse instead: exten => _1NXXNXXXXXX,1,Dial,SIP/BYEXTENSION@iconnect exten => _1NXXNXXXXXX,2,Dial,IAX2/npI42VoD38@voicepulse/${EXTEN} Well,

Hi, I have a PC with two graphic cards one Intel onboard and one Nvidia dedicated card. lspci shows me both and both graphics drivers are installed. Code: > lspci | grep -i vga 00:02.0 VGA compatible controller: Intel Corporation Mobile 4 Series Chipset Integrated Graphics Controller (rev 07) 01:00.0 VGA compatible controller: nVidia Corporation G98 [GeForce 9300M GS] (rev a1) wine.log