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...s.list(c(state2, pars2)), { dT=(a*T-a*T*b*T) - (c1*N*T) - (Kt*M*T) dN=a1 - (f*N) + g * (T/(h+T)) * N - (p * N * T) - (Kn * M * N) dC=a2 - (beta * C) - (Kc * M * C) dM= -(gamma) * M + Vm return(list(c(dT,dN,dC,dM))) }) } ### First Half - Tumor growth to 10% pars <- list( Tini=iniT, Nini=iniN, Mini=iniM, Cini=iniC, a=4.31E-3, b=1.02E-14, c1=3.41E-10, Kt=8E-1, f=4.12E-2, g=1.5E-2, h=2.02E1, beta=1.20E-2, gamma=9E-1, Kc=6E-1, Kn=6E-1, p=2E-11, a1=1.2E4, a2=7.5E8, Vm=0 ) tout <- seq(0, num_days, by=interval_size) state <- c(T = pars$Tini, N = pars$Nin...

Hi! I need to perform ANOVA on a couple of data sets. The only information I have are N, Mean and Standard deviation. I am very new to R, so can someone point me to the right direction on where to go? Thank you very much. Sincerely Niny Rao, PhD Philadelphia University

please can someone send to me his files like zaptel & zapta if he si using TE110P thank you

Hi My asterisk output is: chan_sip.so => (Session Initiation Protocol (SIP)) Asterisk Ready. -- Registered SIP '201' at 192.168.0.55 port 33906 -- Saved useragent "X-Lite release 1011s stamp 41150" for peer 201 -- Executing [907768385144 at default:1] Dial("SIP/201-083e75c0", "ZAP/g1/907768385144|60") in new stack [Oct 4 11:54:27]

Recently, I got the following error messages in CLI periodically. Jan 20 17:43:18 ERROR[8641]: chan_sip.c:11002 handle_request_subscribe: Got SUBSCRIBE for extension XXXXXXXXXXX@from-int from 192.168.0.123, but there is no hint for that extension I have no idea what the error message tell me. I am sure I haven't that account XXXXXXXXXXX in my database and there is no hint extensions in dial

How can I access an environmental variable in Asterisk 1.2.5? It should be possible according to: http://www.voip-info.org/wiki/view/Asterisk+variables which says: Environment Variables You may access unix environment variables using the syntax: ${ENV(foo)} ${ENV(ASTERISK_PROMPT)}: the current Asterisk CLI prompt. ${ENV(RECORDED_FILE)}: the filename of the last file saved by the Record

Is there a command in Asterisk that will cause all Aastra phones to reboot and/or recheck for new firmware? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070312/d394c874/attachment.htm

I''m having some trouble figuring out how to add a path prefix to _all_ paths in a site. I see the option to use :path_prefix when using resources, but this isn''t exactly what I want. I would simply like to add a static prefix of /events/foo/ to all paths, including the root url (for now, until I create a proper events controller). I attempted to do this using map.root but it

Hi all, is where a possibility for simply parsing and changing variables for bad characters ? eg. removing a '/' from a number dialed by a manager-connected application changing 123/4567890 to 1234567890 via bash you could simply use 'echo ${exten/\//}' but i couldn't find a working solution for the asterisk-extensions.conf best regards Dirk Rieger Diese E-Mail und

I'm stuck on queues! The way I read what documentation I have found, if I set up a queue like this: [general] persistentmembers = yes [testq] musiconhold=default strategy = ringall timeout = 10 retry = 5 context = testing member => SIP/100 and then add into extensions something like this: [incomingiax] exten => 1234,1,Dial(SIP/100,10) exten => 1234,2,Queue(testq|tTH|||300)

I have a challenge that is ending up quite interesting. I need to identify which SIP phone "touched a call last", that is, which phone did the last transfer or dialed the original call if no transfers were done. It is easy in the case of a regular, non-transfered call. Just put something in callerid= in sip.conf, and that will show up in ${CALLERID}. The same with an attended transfer,

Hi everyone, I am relatively new to rspec and I am running into a wall in testing my views. I have a RESTful resource Contracts with a nested resource of Line_items. I am trying to figure out how to test the "edit" form of the Line_items. What complicates this is the nested routing and how to account for it, and that there is a partial form (_form.haml) that both the edit.haml and

Hey list, I have a problem when I host 2 SIP-accounts on the same Asterisk-server. Asterisk picks out the SIP-account on alphabetic order A --> Z. In my sip.conf : register => user1:passwd1 at server/user1 register => user2:passwd2 at server/user2 [YOCAN-3starsnet] type=peer host=server username=user1 secret=passwd1 fromuser=user1 accountcode=user1_in [ITCENTER-3starsnet] type=peer

Dear all, I am currently trying to configure a PBX make use of a multiple of outgoing lines, currently my extensions.conf looks something like below >> ; extensions.conf ; 20th October 2008 [globals] sip1=201 sip2=202 sip3=203 sip4=204 [general] autofallthrough=yes [default] [incoming_calls] exten => _89859715,1,Dial(SIP/201) exten =>

Hi people, I have the following dialplan, but it doesn't have the behavior that I think it should have. [default] exten => 2001,1,Answer exten => 2001,n,Dial(local/3005) exten => 2001,n,Hangup exten => 3005,1,Set(__RINGTIMER=10) exten => 3005,n,Macro(exten-vm,novm,3005) exten => 3005,n,Hangup When I execute the Originate (AMI) with the argument Channel=local/2001, It rings

Hi, I'm trying to install R for Mac OS X. I printed out the Mac OS X FAQ (version 1.9-1 2004-03-22) and have been following those instructions step-by-step. I'm a _complete_ novice when it comes to non-mac type software installations and am not very unix literate, but I'm hoping I can get this to work. I'm running a dual 1.8 GHz mac with the latest version of panther.