similar to: How to configure PSTN lines permissions todifferent extensions ???

Hello All, i am repeating this question for the sixth time but i think i was not explaining the problem correctly. ............. Now i will try to explain it...... I have 4 telephone lines(PSTN) in my PBX. Now I want to make groups of the extensions to use that lines. e.g. extensions 12,13 & 31 are in groupA extensions 14 - 20 are in groupB extensions 21 - 30

please tell me that what sip based softphone will beused with Asterisk so that i can trasmit and receive video on my LAN . --------------------------------- Yahoo! Mail Use Photomail to share photos without annoying attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I think you need to enclose the speex header files in the extern "C" {} ________________________________ From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On Behalf Of Alex Lupoian Sent: Sunday, March 12, 2006 6:10 AM To: speex-dev@xiph.org Subject: [Speex-dev] help Hi I am building an application on Visual Studio 2005 using Visual C++ Win32 SmartDevice

I want to play a sound file using GotoifTime function. 1) What should be the appropriate format of this type of sound file? 2) Is there any method to copy this file into the destination directory using the browser of a PC other than the asterisk PC (currently i am using cp to copy the file in /var/lib/asterisk/sounds on asterisk PC)??? Waiting for ur kind reply !!

The "error" I was talking about was in your log: "...== Spawn extension (extensions, 4, 3) exited non-zero on 'IAX2/home_server-6364'..." The call terminated here in a error which prevented the dialplan from continuing. Something there is broken, my recommendation is to check you registrations first inside asterisk: > sip show peers Something wasn't

I cant seem to get outgoing calls to be placed properly .. No matter what I try I get an error from the PSTN company saying that the "call can not be completed as dialed" or "you need to dial a one..." The asterisk debugging seems to show the correct number being dialed out of the zap interface... the "9" is being stripped and there is a "1" where it is

I figured I''d post here before submitting a ticket, but I''m seeing some confusing stuff when dealing with YAML now. I was using it to freeze objects in my database, and so i had some data already around to mess with. I upgraded both Ruby (1.8.3 -> 1.8.4) and Rails ( -> 1.1RC1) and this junk started. I''ve outlined the problem in two pastes, which I''ll

We i have settup it like this it giveme caller id agent id and date-time on gsm file but i want them to be in folder on every day basis datewise. exten => _1NXXNXXXXXX,2,Set(CALLFILENAME=${ACCOUNTCODE}-${EXTEN}-${TIMESTAMP}) exten => _1NXXNXXXXXX,3,Monitor(gsm,/rec/asterisk/apr07/${CALLFILENAME},mb) Any Idea ? Faisal > ------------------------------ > > Message: 16 >

Stop advertising. Le 26 sept. 2010 09:46, "Faisal Hanif" <faisal at vopium.com> a ?crit : Hi Abdul-Basit, If you need only different intervals of billing you can easily do it using any AGI as we are doing it in Perl AGIs using post call billing. But if you need realtime billing then the most stable and flexible option is to use FastAGI+ AMI. I have tested it in JAVA and it

I finally got around to writing myself a NUT UPS monitoring tool. NUT's UPS monitoring CGI tools provide a detailed instantaneous view of the UPS's state at any moment, but provide no history. The attached tool, upswatch, uses Kiss Gabor's UPS::Nut module and Tobi Oetiker's rrdtool to provide a graphical history of the major UPS operating parameters via a web browser.

What am I doing wrong? Goif dialstatus: busy CONGESTION not working. exten => _7NXXXXXX,1,Dial(SIP/7780${EXTEN:1}@pstn-5665,60,tr) exten => _7NXXXXXX,n,GotoIf($[$["${DIALSTATUS}" = "BUSY"] | $["${DIALSTATUS}" = "CONGESTION"]]?line2) exten => _7NXXXXXX,n(line2),Dial(SIP/9780${EXTEN:1}@pstn-1270,60,tr) exten => _7NXXXXXX,n,Hangup() When I try to

This appends a single line to a file, with some cleverness involving guessing the right line endings to use. Also adds a test. --- builder/test-virt-builder.sh | 57 +++++++++++++++++++++++++++++++++++- customize/Makefile.am | 2 ++ customize/append_line.ml | 70 ++++++++++++++++++++++++++++++++++++++++++++ customize/append_line.mli | 20 +++++++++++++ customize/customize_run.ml

Hello, Let's say we have the following function: foo <- function(x) { line1 <- x line2 <- 0 line3 <- line1 + line2 return(line3) } And that we want to change the second line to be: line2 <- 2 How would you do that? The two ways I know of are either to use fix(foo) And change the function. Or to just write the function again. Is there

Hello All !!! I have 4 PSTN lines in the PBX server 1,2,3,4. First line will be used by only one extension (i.e. for the boss) for incoming and outgoing. This line is dedicated for him only.( The remaining lines will be shared by the employees 1) Group A have access to lines 2 , 3 & 4. 2) Group B have access to lines 3 & 4 3) Group C have access to line 4

I''m having a problem formatting some text. I have a textarea that I input my text into, which will then be put into an e-mail and sent out. The problem I am having is when there is a single line break. Multiple line breaks work fine (hitting enter twice), but single line breaks don''t seem to work properly. Take the following input: Line1. Line2. Line3. Line4. Using

Dear All, I am writing to ask a clarification on what R, and in particular lme, is doing. I have an experiment where fly wing area was measured in 4 selection lines, measured at 18 and 25 degrees. I am using a lme model because I have three replicated per line (coded 1:12 so I need not use getGroups to creat an orederd factor). The lines are called: "18"; "25";

Here's a bit of a strange one. I hunted through the ocfs-users archives, but didn't see any mention of this, so here goes... Using coreutils 4.5.3-27 on RHAS 3, the utilities perform as expected against non-OCFS volumes: $ rpm -q coreutils coreutils-4.5.3-27 # cat a 3 line test file. Works as expected $ cat /var/tmp/file line1 line2 line3 # copy that file. Works as expected $ cp

I found a error in my script but that is fixed in this version. /Rickard Eriksson -------------- next part -------------- #!/usr/bin/perl # Version: 1.1 # This is a script to use with mrtg to count shorewall entrys in the log. # # Updates can be found at: http://ftp.shorewall.net/pub/shorewall/contrib/ # Made by Rickard Eriksson riceri@home.se # # Here you can see what i have in my mrtg config

HI LIST CAN U HELP ME If there are multiple sip trunks with the same ITSP then an incoming call is arbitarily matched to the last peer with the same host IP address. This is not a serious problem because the DID is still correct but it does have many insidious effects due to the incorrect channel name Example register=myaccount1 at sip.myitsp.com/line1 register=myaccount2 at

Hi, You can exit out of the 'identify()' routine by either: right-click > 'stop' Or click on 'stop > stop locator' in the top-left of the plot window. The script should continue with line1, line2 etc. Cheers Joe Joe Crombie Information and Risk Sciences Bureau of Rural Science Canberra Australia p: +61 2 6272 5906 e: joe.crombie at brs.gov.au