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...: num 68.8 NA 143.7 NA NA ... $ D2005 : num NA NA 163 NA NA ... $ D2006 : num NA NA 189 NA NA ... $ D2007 : num NA NA NA 14 1.91 ... How do I transform this into something that I can make a time-series of? Sincerely, Gabriel Bergin gabriel@bergin.se [[alternative HTML version deleted]]

Hello all, I'm struck with a very strange problem today. I've an AGI with some code subroutine snippet as follows: sub enable_sbc($) { my $carrier = shift; my $tmp = substr($carrier,1); my $jkh = $tmp; $server_port = $ast_agi->get_variable("SIPPEER($jkh,port)"); $ser_ip = $ast_agi->get_variable("SIPPEER($tmp,ip)");

...ric" with a factor response will be ignored I understand that there is something wrong due to the fact that some of the variables are factors. But as far as I've grasped, it should be possible to include factor variables when using lm(). Am I in error in thinking this? Sincerely, Gabriel Bergin Undergraduate economics student [[alternative HTML version deleted]]

...or there are any considerations I've missed. If anyone is interested in the project please don't hesitate to contact me. I would be especially grateful to anyone with experience releasing open source software that wouldn't mind receiving an email here and there. Thanks. - Mike Bergin mikeb at bergin dot me -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.llvm.org/pipermail/llvm-dev/attachments/20100323/f0c16e58/attachment.html>

...ch queue, it should be 5th in the queue. e.g., helpdesk 3 calls 1 Zap/23-1 2 Zap/12-1 4 Zap/20-1 isp 2 calls 3 Zap/22-1 5 Zap/18-1 Is this type of setup possible and if so, what needs to be done in the config files to accomplish this? Thanks Ron Bergin

Hi, I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone. - If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup. Regards Upendra. -------------- next part

Hi all, I'm trying to run an AGI in PERL which uses the module DBD-Oracle. Currently my AGI is working fine in my two servers but not in my other four servers. When I tried execute an AGI (as a user asterisk) in command line it works fine (even I also declare environmental variables in user profile and in my AGI), but when I tried to call my AGI (perl) in dial plan, it don't get

I'm wondering if any other Asterisk users have a recommendation for a reliable SIP Trunk provider that supports Asterisk and offers decent support. I've worked with Coredial, Broadvox, and Broadvoice and have had some bad experiences with each of these providers. Broadvoice offers low cost service, however I have constant issues with Broadvoice blocking my customers due to Asterisk