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I am just starting to code in R and need some help as I am used to doing this in SAS. I have a dataset that looks like this: Chemical Well1 Well2 Well3 Well4 BOD 13.2 14.2 15.5 14.2 O2 7.8 2.6 3.5 2.4 TURB 10.2 14.6 18.5 17.3 and so on with more chemicals.... I would like to transpose my data so that it looks like this: Chemical WellID Value BOD Well1 13.2 BOD Well2 14.2 BOD Well3 15.5 BOD

...--------------------- The problem surfaced after upgrading to current CVS (10/8) from 9/9. Is anyone else having this problem? Is there something I should be doing differently? Steve ___________________________________________________________ Steve Creel screel@turbs.com

...voicemail system. How could these calls be identified as different from an incoming call to the auto-attendant? How is the appropriate mailbox or extension identified? Thanks, Steve ___________________________________________________________ Steve Creel screel@turbs.com

Dear all, I am trying to fit a generalized linear mixed model to deal with both temporal and spatial pseudorreplication. I have repeated seasonal measurements (3 seasons during 3 years and 2 season during the last year, factor named percod, with 11 levels) of a bird presence/absence (pres.f) on 14 artificial ponds (charca.f). The ponds are integrated in 4 greater spatial units (zepa). I have

Hi, First off, a big thanks to Digium (Mark, John, and Martin) for helping sort out a BellSouth config issue on our PRI. T100P working like a champ! Now it's back to tweaking the configuration on our SIP phones (7960s). The message_uri parameter in the phone's configuration file is working great. Dials comedian mail directly. Is there a way to let voicemail2 know what the incoming

...of the vendors returned thier quotes without including the Cisco phones. So my question: where do you buy your phones? We can't buy direct from Cisco (must have 3 quotes). Thanks... ___________________________________________________________ Steve Creel screel@turbs.com

Hi All, I got a doubt about something I want to do with asterisk. I have this office (site a) with only a Panasonic analog PBX and another office (site b) with an Asterisk Box with an ADIT 600 . I want to interconnect both via IAX. Is it possible to put a new asterisk box in site a without the channel bank and put a card (FXS or FXO???) and connect it to the pbx as a CO line ? What

...of it here and there... does anyone have mgcp working with a 7960? I've gotten the phone to work in "basic phone mode", is that all I'll get, or am I missing something? ___________________________________________________________ Steve Creel screel@turbs.com

...e isn't answered? I would want to unpark and transfer to voicemail, but am not sure how to act on the parked call like that. Am I even close to something that could work? Steve ___________________________________________________________ Steve Creel screel@turbs.com

...s like that. While I can give it a shot (and I will), I'm sure there is someone far more qualified who could probably write it much better and far more quickly. Just my $0.02 Steve ___________________________________________________________ Steve Creel screel@turbs.com

How can I loop through something x number of times in the dialplan? i.e. if I get an invalid extension I want to re-play the menu, but not forever. Maybe 3 tries or something. I'm pretty sure that I've seen it before, where you can increment a variable and do "Gotos" based on it. But I've searched the Asterisk handbook, searched the user archives, and Googled for it,

I bought a couple of 7905G phones with a Callmanager license but i found on the site these phones can have a SIP image (which i downloaded) but before i upload the image i want to know if anybody tested them ? Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031016/4c2821af/attachment.htm

...ther field in the voicemail.conf for 'extension to transfer to' ? 100 => 1234,President's Secretary 101/100 => 1234,Company President Thanks for your thoughts, Steve ___________________________________________________________ Steve Creel screel@turbs.com

I guess the subject says it all.. :) I am running the CVS from right now.. +- 12:25 GMT MySQL CDR logging is installed and working.. Anyone got any ideas?

...respond to any telephony events, it must be stopped and started (though 'stop now' from the CLI does do the job). Any suggestions as to where I might look to find the problem? ___________________________________________________________ Steve Creel screel@turbs.com

I am wondering if it's possible to use a bunch of cards in a PCI backplane instead of going out to the extensions with T1 and then and adapter. How are people connecting to large amounts of extensions?

It is possible to connect ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank to Asterisk ? Somebody offered me that hardware, but I do not know if thats good hardware for Asterisk. rgs, Bartosz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030820/4a9e4608/attachment.htm

...us what > is local to you, > right? Care to finish the details so those on the > list can help. > -- > Steven Critchfield <critch@basesys.com> > > > --__--__-- > > Message: 3 > Date: Mon, 24 May 2004 13:34:02 -0400 (EDT) > From: Steve Creel <screel@turbs.com> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Channelized T1, SIP > phones, HW Echo Canceller > Reply-To: asterisk-users@lists.digium.com > > I have a channelized T1 coming in from our telco, > terminated onto a TE405. > There are three channelba...

I have two Adtran 750's connecting our analog phones to asterisk. On occasion, I get a channel that gets "stuck" off hook. 'show channels' shows: Zap/27-1 (longdistance s 1 ) Rsrvd (None) (None) And will just stay like that until the phone is manually picked up and hung up again (or asterisk is stopped/started). I guess this is a function of an unclean hangup (being

hello, i found in list archives some notes about grandstream sip voip phones. Does anybody succesfuly tested those phones with asterisk ? Mark ? What about the prices ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ ------------------------------------------------------------ A mind