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Hi all, I have a Cisco ATA186 connected to an Asterisk Server (SIP) The dialplan uses 1XX for local extensions and XXXXXXX for external numbers, where the first digit is always different than 1. In this moment, when I dial 123 for example, ATA waits till timeout before dialing that number. The same for the longer one. How can I do to make it dial imediately when 3 digits starting with 1 are

...ar regression for use as a purely descriptive statistic of goodness of fit? Most of the readers of my report will be vaguely familiar and more comfortable with R^2 than with any other regression diagnostics. Paul M. Jacobson Jacobson Consulting Inc. 80 Front Street East, Suite 720 Toronto, ON, M5E 1T4 Voice: +1(416)868-1141 Farm: +1(519)463-6061/6224 Fax: +1(416)868-1131 E-mail: pmj at jciconsult.com Web: http://www.jciconsult.com/ -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send &q...

...table. I am trying to use GLM. By converting the variables to factors and running the following regression: fwr1<-factor(fwr) zz1<-factor(zz1) res1<-glm(fwr1 ~ zz1,weights=cnt) Is that appropriate? Paul M. Jacobson Jacobson Consulting Inc. 80 Front Street East, Suite 720 Toronto, ON, M5E 1T4 Voice: +1(416)868-1141 Farm: +1(519)463-6061/6224 Fax: +1(416)868-1131 E-mail: pmj at jciconsult.com Web: http://www.jciconsult.com/ -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send &q...

...t if another digit is entered (say # 140887133445, the timeout changed to 5 seconds). # if 0 is entered, after 4 seconds, it will be send out. # if 011 is entered, the time out changed to 9 seconds. DialPlan:*St4-|#St4-|911|1>#t8.r9t2-|0>#t811.rat4-|^1t4>#.- On line 2, do you mean that there isn't any loop current, e.g. you don't hear local tones when you press buttons on the analog phone? If so, first be sure that it is not your phone or cord, by trying the working phone, with its cord, in the 'phone 2' jack. Also, make sure t...

My Cisco 7905 can't register with Asterisk (1.0.7-BRIstuffed-0.2.0-RC7k on Debian stable). It could, however, register with another installation of Asterisk and the settings on the phone (apart from the SIP proxy address) haven't changed since then. On the new Asterisk box my sip.conf contains this: [jeremy] type=friend regexten=801 allow=g729 host=dynamic secret=PASSWORD nat=yes

I'm a mathematician and programmer working on experimental surround sound techniques. Some of the ideas I'm working on require dozens of channels. These channels are often highly correlated and are very well suited to compression. I'm new to Vorbis. Does it attempt to address such issues? Thanks, --Richard --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project