similar to: Is logic right?

I am new to asterisk. I need help getting started, if it's even worth getting started. I say if it's worth getting started because I'm not sure if my hardware will even work with asterisk. I have a US ROBOTICS 56K V.90 PCI SOFT MODEM. I have standard twisted pair telephone wire. I can't afford to alter my hardware. I know I won't be able to do any sophistocated VoIP stuff.

Hi everybody, I have a data set with a value and a status (positive or negative case) and I want make a ROC Analysis. So, with ROCR Package, I have got the ROC curve (True Positive Fraction [tpf] according 1-True Negative Fraction [1-tnf]). http://r.789695.n4.nabble.com/file/n4469203/01.png But, now I want a new graphic which show the sum of true positive fraction and true negative fraction

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

Hello All, I was wondering where does Asterisk stores the blacklist numbers? I looked into the dialplan and it shows that it *"Set(DB(blacklist/${blacknr})=1)"* the number... Does it save in MySQL DB? hyperion*CLI> show dialplan app-blacklist-add [ Context 'app-blacklist-add' created by 'pbx_config' ] '1' => 1.

Hello everyone... I'm trying to get up a testing pbx installation. Following instructions of what've read from the handbook and from asterisk's wiki, I wrote the dial plan as follows: [general] ; ; static = yes ;[globals] ; [default] ; exten => 0,1,Answer() exten => 0,2,Playback(fcopba1) exten => 0,3,Hangup() exten => *0,1,Answer() exten => *0,2,Record(fcopba1:gsm)

When I first to the "add extension" the "show dialplan" has the lines that say "SIP/" but after I do a "save dialplan" and a "stop gracfully" and restart the lines with "SIP/" are gone. ************************ "Show dialplan" before: ************************ asterisk01*CLI> [ Context 'default' created by

I have an "outbound-ld" context as follows: [ Context 'outbound-ld' created by 'pbx_config' ] '_1NXXNXXXXXX' => 1. Macro(enumdial|${EXTEN}) [pbx_config] 102. Wait(1) [pbx_config] 103. Set(LINE=${IF($[${LINE}=pots]?link2voip:${LINE})}) [pbx_config]

I have a [start] context that all my inbound and '0' calls are routed into. Because of the way I want to set my system up, I want to prompt the user to enter a 1 if they know the extension, or a 2 for a directory and nothing else. It works, however there is a 5 to 10 second delay after enter the 1 or 2 before the system responds. I have read over the wiki on how asterisk handles digit

Greetings, Running CVS HEAD about 3 weeks old, I have been beating my head trying to get this to work properly.. Or at least figure out what's going on. Maybe I have done things wrong... I have created a 'catch all' extension at the end of our last context where all phones & voicemail extension exist. This catch all is included in all and works quite nicely except when voicemail

Asterisk isn't playing my voicemail greetings even though they are defined. Below are the relevant configs(from show dialplan) as well as the level 3 verbose messages asterisk is giving. Also a listing of the directory. Asterisk just plays the "The person at extension..." message, not the greetings I have recorded. Thanks -- asterisk*CLI> show dialplan macro-stdexten [

Hello Asterisk users, Can someone thwack me with a clue stick please? I am following the Asterisk TFOT book Dial() example trying to get the limit and announcements to work as per below. These settings seem to have no effect. There are no warning messages after 4 minutes or every 30 secs thereafter and the call lasts longer than 5 minutes. gunner*CLI> show dialplan [ Context

Hi all, I am running a Samba 3.0.0-14.3E server on a RedHat Enterprise WS3. Here's the problem: When I am connected from my PC (Windows 2000) and run the MacAffee v4.5.1 virus scanner on the connected share, I see the following in the Samba log file (debug level=2) for what appears to be each file in the share: [2004/07/20 13:02:51, 2] smbd/open.c:open_file(250) sullivan opened file

Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' => 1. Wait(1) [pbx_config] 2.

My wife and I each have our own workstation dual-booting WinXP and Gentoo Linux. We also have a third which runs Gentoo all the time. I wanted to set up samba on the third box and provide some extra storage space for both our Windows installs. It worked for a few days, then all of a sudden it stopped letting my account (michael) in while still letting my wife's account (amy) use the share.

I have a dataframe that looks like this: > str(chr) 'data.frame': 84 obs. of 7 variables: $ county: Factor w/ 3 levels "Broome","Nassau",..: 3 3 3 3 3 3 3 3 3 3 ... $ item : Factor w/ 28 levels "Access to healthy foods",..: 21 19 20 18 16 3 2 6 17 8 ... $ value : num 8644 15 3.5 3.9 7.7 ... $ low : num 7897 9 2.5 2.6 7 ... $ high : num 9390

Yes, I have tried that too (i.e, exten => +17775551212,1,Log(WARNING, TWILIO)). It does not work and NO error message in CLI. I have also tried http://orourketech.com/elastix-plus-sign-caller-id-messing-things/ since I first emailed this group, but that does not seem to work either. Here is my log: [Dec 2 15:09:28] NOTICE[26464]: res_pjsip_session.c:1920 new_invite: Call from

I was wondering why *foo at default should match '_*[0-9a-zA-Z].*0.' in 1.6.13. Who is making the parse error, * or me? CLI> dialplan show *foo at default '_*[0-9a-zA-Z].*0.' => 1. NoOp(${EXTEN}) [pbx_config] 2. Set(accountcode=${CUT(EXTEN,*,2)}) [pbx_config] 3. Set(extension=${CUT(EXTEN,*,3)}) [pbx_config]

i have a box running debian sarge with asterisk installed from distribution packages: CLI> show version Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by kk@nyx on a x86_64 running Linux I have managed to configure a simple dialplan (the PBX task is quite simple as this is a small office with just a few phones) I have one Zap (PSTN) line connected to it and one SIP to a local provider. After

At approximately 3:15pm I shut down the office MySQL server to change out some hardware. Shortly after I received a call from one of two customers whose asterisk servers output CDR data to that server. They could not place or receive calls. Shortly after that I received a call from the other customer. I'm below providing output from the message log (At debug level). I don't see much

Hello, I am new on Ubuntu. I would like to use R, but I tried Kate and Scite. The first one keeps trying to use KDE applications,while the other does not understand the language. I have searched for another editor for hours, in vain. Which editor should work with Ubuntu? I am looking forward to your answer, thanks, Camille