search for: sveave

Dear list, Latex/Sweave has trouble processing Sveave-output coming from the summary-command of a linear Model. >summary(lmRub) The output line causing the trouble looks in R like this Signif. codes: 0 ?***? 0.001 ?**? 0.01 ?*? 0.05 ?.? 0.1 ? ? 1 In my Sweaved Tex-file that line looks like this Signif. codes: 0 ?***? 0.001 ?**? 0.01 ?*? 0.05 ?.?...

Hi I want to be able to use one provider if I dial 0 before the number and another if I dial 1 before, how can I do that in asterisk 1.6? /ralf Ralf Tr?skman, IT AdLibris AB, Box 3667, 103 59 Stockholm. Bes?ksadress: Sveav?gen 56C, 111 34, Stockholm - Obs ny address! Dir: +46-(0)8-5460 60 91, mob: +46-(0)70-7548074, vxl: +46-(0)8-5460 60 00, fax: +46-(0)8-5460 60 99 ralf at

Hello List! We're having trouble making call deflection on ISDN PRI. We would like to transfer a call to an external extension but keeping the callerid of the caller so it can be presented to the receiver of the transferred call. At the time we're using Zaptel 1.4.5.1, Asterisk 1.4.11 and Digium hardware TE420B. We've ordered the service (CD) from the phone company. The

Hi List! I got this error while upgrading zaptel: make -C firmware hotplug-install DESTDIR= make[1]: Entering directory `/usr/src/zaptel-1.4.7.1/firmware' Attempting to download zaptel-fw-oct6114-064-1.05.01.tar.gz --10:53:09-- http://downloads.digium.com/pub/telephony/firmware/releases/zaptel-fw-oct6114-064-1.05.01.tar.gz Resolving downloads.digium.com... failed: Temporary

Hi [Feb 17 07:59:45] WARNING[21539]: channel.c:3477 ast_request: No channel type registered for 'USTM' [Feb 17 07:59:45] WARNING[21539]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'USTM' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) I get this after I restart my asterisk 1.6, it all worked yesterday. I have the

Hi Im using a sip phone SPA921, and the one that calls me can hear me but I cant hear them, when I make the call I can hear them. Im running asterisk 1.6 behind a firewall, I have port 10000-20000 for rtp and 5060 for sip forward to my asterisk. Any tips? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Sveav?gen 56C, 111 34 Stockholm, Sweden Dir:

We have 40 nortel ip2002 phones connected to asterisk 1.6, the problem I have is that the phone looses the connections with the server and then drops calls, we can reconnect but the customers don't like it. Anyone has the same problem? /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Sveav?gen 56C, 111 34 Stockholm, Sweden Dir: +46-(0)707548074, vxl: