search for: taff

...uot;, which I downloaded and extracted to "C:\Program Files\R\R-2.6.2\library" on my pc. I tried > nora <- read.dta("nora.dta") and got Error: could not find function "read.dta" I guess "foreign" needs to be installed. How do you do this? John Taffe Biostatistician Centre for Developmental Psychiatry and Psychology Faculty of Medicine Nursing and Health Science Monash University, Australia

...('/Event: Newchannel.*/') or die "Unable to determine call status", $tn->lastline; # wait for asterisk to process $tn->print("Action: Logoff\n\n"); I'm not a programmer (as u can probably tell) so any pointers would be much appreciated. Cheers, Taff. ___________________________________________________________ ALL-NEW Yahoo! Messenger - all new features - even more fun! http://uk.messenger.yahoo.com

Howdy, Was there ever a fix for this? I have Trix 2.6 running asterisk 1.4 and have to set an extension with call-limit=1. However that user can no longer do attended transfers from Linkys 962 ip phone. Is there anyway around this? Cheers, Taff..

Hi All, Does anyone know of a way to dial two different outbound numbers and bridge them together using the Asterisk API? Cheers, Taff. --------------------------------- ALL-NEW Yahoo! Messenger - all new features - even more fun! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050120/4c52ad3f/attachment.htm

...way of increasing the delay before asterisk picks up the incoming PSTN call? I'm using a tdm400p with fxo card. It seems to pick up the inbound call immediately. I want to delay detecting the call by about 10 secs if poss. Done some searching but couldn't find anything relevant. Cheers, Taff!!! ___________________________________________________________ Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail http://uk.messenger.yahoo.com

...vr then is sent out to an extension using the dial command. We have to carry the rtp streams in this case as asterisk cant send the reinvite after the ivr has stopped playing the message as we already connected the call. Question: Any way around this or is there a better way we can do it? Cheers, Taff....

Hi, I'm thinking of purchasing a TDM400P card and was wondering if anyone has experienced any echo issues with phones off these cards connecting to the PSTN via the X100P cards? I have had my fingers burnt with a Voip phone > X100P. Cheers, Taff. --------------------------------- ALL-NEW Yahoo! Messenger - sooooo many all-new ways to express yourself -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040622/2cc00c26/attachment.htm

...on channel 1 Sep 9 12:31:48 WARNING[4830]: Timeout, but no rule 't' in context 'incoming' Show version Asterisk CVS-Nv1-0-9-08/17/05-15:34:37 Uname -a (running centos 4.1) Linux npc-asterisk-01 2.6.9-5.0.3.EL #1 Sat Feb 19 18:26:49 CST 2005 i686 i686 i386 GNU/Linux Any ideas? Taff! ___________________________________________________________ Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail http://uk.messenger.yahoo.com

...aps, Is there a way to detect/highlight poor quality voice calls going through an asterisk server? Was thinking of picking up a cdr record or some other variable showing poor quality on calls when the internet is having issues. Is there any qos or poor audio quality variables available? Cheers, Taff. ___________________________________________________________ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. http://uk.answers.yahoo.com/

My wife's got an appropriate Southern England (Wimbledon) accent and I'm sure she would try her hand. Does anyone have a comprehensive list of the words that need to be said? Matt, do you have them if your wife's done a set for French users? Mark, if you have the kit maybe you could chop up the file? I write a utility to chop up and compress the wave file based on some of the C