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...hat would happened if someone make a huge file transfer: will it make trouble on the Sip connections ? I think so, that?s why I?m asking you if there is a good (better) way? Maybe connect SIP phones on a separate network and setup a linux box as router w/ traffic shaping ? Thanks. Yves Raber yves@swissinfo.org _______________________________________________________________________ Dreaming of a Swiss Account? Get it here: http://freemail.swissinfo.org

Hi, When I use NoCDR application I obtain this warning in console log: Mar 1 11:16:08 WARNING[3513]: cdr.c:114 ast_cdr_free: CDR on channel 'SIP/492-7371' not posted Mar 1 11:16:08 WARNING[3513]: cdr.c:116 ast_cdr_free: CDR on channel 'SIP/492-7371' lacks end Can someone explain to me what is due? Thanks.

Hello, hopefully someone will remember my previous problem to load the Rcmdr library from within GNU R resulting in an error message: "libnvidia-tls.so.1: cannot handle TLS data". Some suggestions have been raised by Christian Schulz and others that unfortunately didn't work around this error. Nevertheless I felt very grateful for your suggestions! This morning I investigated

Hi All: A couple of weeks back, I asked a question on the list that I was invited to provide an introductory lecture on R to a group of academicians in Kolkata. I thank all of you who had generously guided me in providing me web links and words to the wise. Time to give back. I did the presentation on introduction to R and uploaded the presentation files at the following site:

Hello, first trials to run R from inside of Emacs repeatedly gives me: ---------------------------------------------------------------- > ?sink WARNING: terminal is not fully functional - (press RETURN) ----------------------------------------------------------------- Seems like this is going to happen to all help calls using a questionmark! Any idea how to change this, maybe this

Hello, I just encountered this error from within R: -------------------------------------------------------------- > library() Warning message: library '/usr/local/lib/R/site-library' contains no package in: library() --------------------------------------------------------------- which is basically correct since all the looked up packages are in /usr/lib/R/site-library, not in

Hello, I am tinkering a bit with options(), namely, how to query and set them up to suit my needs. 1) The basic problem seems that the editor entry in options looks like this: > options() --------------------------------------------------------- $editor [1] "emacsclient" --------------------------------------------------------- After setting this option to /usr/bin/emacs by

Hi, I currently following some introductory material "Data mining with R", which was highly reccommended to me (its really great!). During my studies I tried to run a given function to identify and substitute unknown values. Seemingly, running this function (page 57) did not work and gives me an error message. It would be very easy just to copy and paste this function to ask for

Hi, after starting Emacs/ESS/R environment I tried to launch "edit" or "fix". This normally should fire up the $editor, isn't it. Instead of this I regularily I run into an error that there something wrong with $editor. ----------------------------------------------- > op <- options(); str(op) Amongst many entries you'll find this: $ editor

Hello, pointed to the article "Debugging Without (Too Many) Tears" of Mark Bravington in Rnews Vol 3/3, December 2003 there is a package mvbutils mentioned. So I started to install this package from within R > install.package(mvbutils) I receive an error essage that this package is not found! Is due to a merge,removal or rename of this package? When I try to follow this

Hi all, I have installed a test machine with asterisk in order to try it. I have a problem with capi channel (chan_capi 0.3.4a). When an external call directed to an internal Ip phone is not answered I obtain this warning repeated many times: .... .... Jul 12 16:13:43 WARNING[1209214400]: app_dial.c:302 wait_for_answer: Unable to forward frame Jul 12 16:13:43 WARNING[1209214400]: app_dial.c:302

Hello, I'm writing a AGI Perl script and use the following lines to initiate a call : $res = $AGI->exec("DIAL $dialstr"); And then : $answeredtime = $AGI->get_variable("ANSWEREDTIME"); The problem is that I need to know immediately when the call has begun. I can have the call duration & establishment time at the end of the call but not in real time when it

Hello, I was wondering if there is a simple way to limit the number of simultaneous calls in an Asterisk PBX ? I've seen that we can make this easily per channel (like in SIP.CONF) : incominglimit=X, but I'm looking to limit the maximum calls all channels together. Another thing. Working with asterisk-perl, I need to get the call duration, currently I use

I've setup a SIP channel between two Asterisk box, and use Manager API to generate some calls. It's working quite fine, except this message (on the caller-side) : Jan 10 18:18:09 WARNING[25046]: chan_sip.c:6805 handle_response: Forbidden - wrong password on authentication for INVITE to '"sip1" <sip:asterisk@192.168.1.200>;tag=as77e9ebbb' But the call is going

I'm trying to disable musiconhold, simply because I don't need it. But then chan_iax2 is complaining : undefined symbol: ast_moh_stop. Is there a way to completely disable moh (maybe a compilation option) ? Thank you. Yves

Hello, I'm trying to find a good solution for load-balancing of several Asterisk box, for each VoIP protocol (IAX,SIP,H323). My idea is to have a front VoIP router (or several) who dispatches the calls of the different boxes. This routing can be done with SER for SIP (redirect server) & GnuGK for H323 (gkrouted 2). Now for IAX, is it possible to configure an Asterisk box to act as

Hello, When dialing somewhere and the other side is down, Asterisk waits until dial timeout before sending "CHANUNAVAIL". I think that if after several seconds there are not any reply (I mean at the IP level) we could consider that the link is just down and handle the situation. Is it possible to configure Asterisk to have this behaviour? Many thanks. Yves.

Hello, right now I'm thinking about running R 2.0.0 on box A (Debian SID) but at the same time having access to the ressources of box B (Ubuntu Linux) regarding disk capacity, RAM, idle CPU cycles . Is there anyone of you that has already installed and administred such a tiny (home based) cluster? Well, the term cluster seems quite a bit of an exaggeration related to what I am looking

Hello, I'm using Asterisk stable (1.0.3) with Asterisk-oh323 (0.6.5). Everything is working fine, well, except that : when a call is made from an h323 device (gnomemeeting for example), the caller does not hear any ringing at all, he suddenly hears the person who answers the phone. That can be quite disturbing for the users. Any help would be very welcome. thank you. Yves

Hi, I am fairly new to GNU R ! At the moment I am doing an intensive learning on the basics of GNU R-1.91, especially graphics like plots and alike, by reading the introductory docs! Well, except some occasional glitches (X11 output errors) everything seems to be fine, thanks to developers for this fine program! But there is a slight problem with the size of fonts in graphics, i.e. its very