search for: rosebush

...# Read in data test <- matrix(data=c(2.52,9.5,3.07,2.5,1.99,8.95), nrow = 6, byrow=TRUE) # Read in and attach labels (names) to data rownames(test) <- c("Mount Pleasant","Jordan","Oil City","Pleasant Valley","Village of Lake Isabella","Rosebush") # Set plot limits: xmax <- nrow(test) nvec <- ncol(test) ymax <- ceiling(max(test)) yinc <- 1 # Generate Pareto order test <- test[order(test[,1],decreasing=TRUE),] # Set color palette MyCols <- rep(c("lightcyan","cornsilk","lavender"), each...

...an extension that is NOT in the dialplan then it would play a not-in-service gsm file and then play congestion tones. I would rather like this better than just hearing a busy signal on my phones.. I DID search around on the wiki and using google and could not find anything. Thanks. -- Stephen Rosebush, srosebush@desynched.org http://www.desynched.org/ // Hardline // IP Phone USA: 1-248-724-4452 x201 FWD: 63420 x201 Netherlands: +31-(0)20-6598858 x63420 x201 IAXTEL: 1-700-356-6191 x201 United Kingom: +44-(0)870-3403054 x201 SIP: sip:srosebush@desynched.org

Hi everybody, I'm running an asterisk box -cvs version since few monthes, updated it middle of may and a last one on thursday (24 june) Since this one, my H323 calls loose they audio, both sides. Calling directly from Gatekeeper is ok, so problem comes from h323 asterisk channel. I saw few people telling about similar problem begining of month, does they solve their problem? I also grab

That could explain why it wouldn't work on any of my sip extensions I tried it on this morning when I first read about it and thought cool the things you learn. Is there anyway to make it work on Sip extensions? Cheers, Dean -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Aaron J. Angel Sent: Wednesday, 23

Hello I have heard that i can put a file in a certain directory to get * to initiate a call. Is this true ? if so where would i look ? Best Regards Simon Garvey

Does anyone know if the 1-800 iaxtel gateway is down? I've been trying to use it all day today and asterisk says it's ringing: Channel (Context Extension Pri ) State Appl. Data IAX2[iaxtel]/1 ( s 1 ) Ringing AppDial (Outgoing Line) SIP/2201-a253 (home 18888476626 1 ) Ring Dial IAX2/XXX:YYYY@iaxtel.com/18888476626@iaxtel But I

I am trying to set up the Message Waiting Indicator (stutter tone/light) so that my cisco ata-186 will let my phones know there is a message waiting. However this does not seem to be very well documented. I found this on wiki mailboxnumber@context ... where does that go? Do I put it in my SIP.conf definition for my cisco ata, or where. In my SIP cisco definition i already have a