similar to: sementation fault with mpg123

{Note the DATE ! this message bounced ... because it was sent to R-announce which was definitely the wrong list anyway. MM - your list maintainer } Dear R-Team I have installed R version 0.990 and the foreign package under Windows NT.. Whilst lookup.xport() works as it should, read.xport() crashes reporting an "access error" as well for larger datasets as also for small

Hi, Doing more search i've discovered package RGDAL that can write a geotiff file with projection. I saved a geotiff file in UTM projection and if i read the file back in R and check the projection seems that everything is OK. But if i load the file in ArcGIS (ESRI product) i get the warning that the file is missing spatial reference so it cannot be projected, but it is displayed correctly.

Hi everyone, This is my first attempt at a PV guest install from scratch and I''m a tad confused. Have I answered one of the questions virsh-install asks wrongly? Does anyone know of a general guide for installing PV guests with virsh-install? (Most Centos docs seem to use the GUI based virt-manager and I only have command line access). Am I meant to make use of the files here:

Hi, I recently installed asterisk, i used sterisk-1.4.20.1, i i set directrtpsetup to yes, no whow would i know if the rtp/media is not passing to asterisk. any tool> or can u just sniff? regards, ron

Hi, I'm getting a situation where 2 of my trunks (loopstart pots) are occationally bridged together (3624 Newbridge channel bank - asterisk signalling=fxs_ks for trunks) and are staying off hook until I do a 'soft hangup' on one of them. When I listen on a butt set each of the lines are silent (or at least by the time I find them). I am guessing one of the end users isn't properly

Does anyone know what this means "Warning [65542]: chan_sip:c:501 retrans_plct: Maximum retries exceeded on call 7438737dc873850@172.16.0.52 for seqno102 (Non-critical Request. 172.16.0.52 is the Asterisk Server I'm guessing that I have something miss configured just not sure what it is. If you need more info just ask.

Do some reading about contexts in *. Basically, you want all "public" sip requests to land in a dialplan context that has no access to PSTN, and requests from your own SER box(es) to land in another context (that DOES have access to PSTN). You can achieve this by adding an entry to your sip.conf for your SER box with it's IP address (and context) specified. ----- Original

Hi, I'm trying to add mysql CDR onto a vanilla Asterisk 1.2 install. The add-ons pack has been installed for a while, so now I'm trying to add the Mysql config. I've created a mysql database, added the grants for a user acces, and can run a mysql -u asteriskcdruser -p and can connect to the database. I've been using this as a guide:

, as it seems to be running Asterisk-11. =A0I&#39;ve previously installed A= sterisk-11+FreePBX in a VM, and this appears to be very similar. =A0Is ther= e any upside to using AsteriskNOW vs. Asterisk+FreePBX? Other than the obvi= ous fact that everything is nicely placed on an iso for ease of installatio= n?<br> <br> As for the actual upgrade, is it possible to step through each

hii guys: i get the message from the asterisk: Started music on hold, class 'default', on Local/s at skype-web-callback-dial-263to263-1775,1 [2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: Unexpected freqency 11025 [2008-11-11 14:32:41] WARNING[1781]: file.c:322 fn_wrapper: Unable to open format wav [2008-11-11 14:32:41] WARNING[1781]:

right, so far, here is what I've done: I succeed in take in a new variable the UUS1 field sent with the connection request for incoming calls. It was quite simple afterall... (I just had to find where the data CMSG->Useruserdata is coming in chan_capi.c) Now I would like to know where this field is instanciated for outgoing calls in order to control this step? I am looking for that but I

<snip> Replaces: pickup-9582-c0a80101-d-4 at 192.168.101.102 <snip> This Replaces header refers to RFC3891 which is not yet supported in Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA) This INVITE fails with : <snip> chan_sip.c: Trying to pick up 7792 at subs <snip> app_directed_pickup.c: No target channel found for 7792. If I'm dialing *87792 instead

Hi i got lots of this from the asterisk console what does this mean? format_wav.c:247 update_header: Unable to find our position asterisk console: Oct 24 16:39:19 WARNING[4432]: format_wav.c:247 update_header: Unable to find our position Oct 24 16:39:19 WARNING[2812]: format_wav.c:247 update_header: Unable to find our position Oct 24 16:39:19 WARNING[4430]: format_wav.c:247 update_header:

Hi, I've spotted weird crash of Asterisk cvs Stable. I have defined queue in queues.conf : [prodaja] music = default announce = queue-markq strategy = ringall context = from-pstn timeout = 15 retry = 5 maxlen = 0 announce-holdtime = no announce-frequency = 30 announce-holdtime = yes monitor-format = gsm|wav|wav49 monitor-join = yes eventwhencalled = yes member => Agent/1000

I get the following error when trying to play wav files for my IVR menu. Does anyone know what this means or how to fix it? [Oct 17 01:04:23] WARNING[9799]: format_wav.c:124 check_header: Does not say fmt Thanks! David

or we need to implement a SIP server to integrate with Asterisk in order to= provide full picture of VOIP system? Thanks. > Date: Wed, 5 Sep 2007 13:30:21 +1000 > From: devraj at gmail.com > To: asterisk-users at lists.digium.com > Subject: Re: [asterisk-users] How to make call from asterisk? >=20 > Helps us help you further, what do you intend to do? >=20 > - Dial

i call to my tollfree number buy my CLI send the next error: Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected freqency 22050 Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/procall3.wav Aug 14 08:15:22 WARNING[25931]: file.c:828 ast_streamfile: Unable to open procall3 (format ulaw): No such file or directory Aug

Hi, I've got a brand new Asterisk 11 setup for which I would like to keep the number of loaded modules to a minimum. My goal is to this setup in a pure SIP environment, for switching incoming calls to outgoing tSIP trunks. When I leave autoload=yes in /etc/asterisk/modules.conf, I can handle an incoming SIP call with a Playback app. When I leave autoload=no in /etc/asterisk/modules.conf, it

I have a persistent problem with a PBX I commissioned recently. After a few days it goes into a spasm, creating thousand of log files and giving the message below on the CLI. Dell PE 1600 with Sangoma A200. pbtpbx*CLI> show version Asterisk 1.2.14 built by root @ pbtpbx.local on a i686 running Linux on 2007-01-13 18:31:56 UTC Asterisk Queue Logger restarted Rotated Logs Per SIGXFSZ

Hi All, I am trying to get Asterisk up and running on my new Mandrake 9.1 install. I've installed Linux in the "standard" mandrake security mode, and "su" to do my attempts at install. I managed to obtain the source from CVS, and have been able to compile Zaptel. I then ran insmod zaptel, and also make config. I think I have compiled and loaded Zaptel successfully as