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...in the data. I need to obtain similar results in R, but I do not know. Please, Could any one suggest how to make this with R? Thanks in advance, Xavier ======================= Xavier G. Ordo?ez Doctoral Student Facultad de Educacion Departamento MIDE Universidad Complutense de Madrid C/ Rector Royo Villanova s/n 28040 - Madrid Spain Tel: +34-913946140

Dear R people, I wonder if you could give me a hand with some of my data. I have a very typical analysis in biology, however it is difficult for me to find the right way to analyse. I had a group of animals, I gave them a treatment, and I measure a variable along time -one??s per day- along 5 days,for example(fake data): Animals Time1 Time2 Time3 Time4 1 1 5 3

Hi, I've noticed that when I set a phone on DND (phone-side DND, meaning it rejects calls with a busy status, SIP 486 response code I believe) the queue keeps on trying the phone over and over again. This creates issues in terms of CDR entries - in a scenario where there is only one phone on DND, and a delay between attempts of 1 second, the queue will attempt to ring the single phone

...ameter on inbound traffic violates RFC7118 (Reported by Marek Cervenka) * ASTERISK-26546 - mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2' (Reported by Tzafrir Cohen) * ASTERISK-26566 - res_rtp_asterisk: RTT miscalculation in RTCP (Reported by Hector Royo Concepcion) * ASTERISK-26604 - chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc. (Reported by Michael Kuron) * ASTERISK-26603 - [patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no (Reported by Alexei Gradi...

...ameter on inbound traffic violates RFC7118 (Reported by Marek Cervenka) * ASTERISK-26546 - mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2' (Reported by Tzafrir Cohen) * ASTERISK-26566 - res_rtp_asterisk: RTT miscalculation in RTCP (Reported by Hector Royo Concepcion) * ASTERISK-26604 - chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc. (Reported by Michael Kuron) * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported by snuffy) Improvements made in this release: ---------------------------------...

...und traffic violates RFC7118 (Reported by Marek Cervenka) * ASTERISK-26546 - mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2' (Reported by Tzafrir Cohen) * ASTERISK-26566 - res_rtp_asterisk: RTT miscalculation in RTCP (Reported by Hector Royo Concepcion) * ASTERISK-26604 - chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc. (Reported by Michael Kuron) * ASTERISK-26603 - [patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no (Report...

...Reported by Alexei Gradinari) * ASTERISK-24330 - Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118 (Reported by Marek Cervenka) * ASTERISK-26566 - res_rtp_asterisk: RTT miscalculation in RTCP (Reported by Hector Royo Concepcion) * ASTERISK-26604 - chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc. (Reported by Michael Kuron) * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported by snuffy) * ASTERISK-26520 - codec_opus: Generated fmtp line...

...Reported by Alexei Gradinari) * ASTERISK-24330 - Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118 (Reported by Marek Cervenka) * ASTERISK-26566 - res_rtp_asterisk: RTT miscalculation in RTCP (Reported by Hector Royo Concepcion) * ASTERISK-26604 - chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc. (Reported by Michael Kuron) * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported by snuffy) * ASTERISK-26520 - codec_opus: Generated fmtp line...

...- Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118 (Reported by Marek Cervenka) - [ASTERISK-26566 <https://issues.asterisk.org/jira/browse/ASTERISK-26566>] - res_rtp_asterisk: RTT miscalculation in RTCP (Reported by Hector Royo Concepcion) - [ASTERISK-26604 <https://issues.asterisk.org/jira/browse/ASTERISK-26604>] - chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc. (Reported by Michael Kuron) - [ASTERISK-26608 <https://issues.asterisk.org/jira/browse/ASTERISK-26608>...

Hello, We are using AstTAPI to make call?s from Outlook and it woks perfect. Our issue is that it doesn?t work so well with our CRM. Although the TAPI line initialization is successful, AstTAPI doesn?t report the Phone Number and TAPI doesn?t know about our extension. Anybody knows how to solve this? Ours log: 2005/08/02 15:42:32:476 Information 0x50002 in Process 2040 Thread 1784 Location: