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Is it possible to enable automatic forwarding of Xprint server displays? The -X/-Y switches just forward the video Xserver display but it seems there is no way to get the Xprint server displays forwarded, too. Thanks... Felix Schulte -- _ Felix Schulte _|_|_ mailto:felix.schulte at gmail.com (0 0) ooO--(_)--Ooo

All, I get this whenever trying to dial to a peer when the peer registered to another server. I'm basically trying to use realtime to check for the peer and dial it. Mar 24 09:16:47 VERBOSE[4527]: -- Executing Dial("SIP/brak-f69f", "IAX2/brak-test/107") in new stack Mar 24 09:16:47 DEBUG[4527]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name =

All, I'm having a heck of a time getting hdlc to work on kernel 2.6.11.9 .. I compiled hdlc, hdlc_gen, hdlc_cisco, hdlc_raw, into the kernel (note into, and not 'modules'). System comes up, I configured zaptel.conf span=1,0,0,esf,b8zs nethdlc=1-24 modprobe wct4xxp ztcfg sethdlc hdlc0 cisco ifconfig hdlc0 up All of this works fine, believe it or not. I have a T1 cross plugged

Dear R-people, I want to use an idea from LaTeX in the work flow with R. It is possible in LaTeX to have a main file from which other files are called (ie included). So for example if you have book, the main index file would call each chapter separately. Is there something comparable in R that follows the above 'include' idea from LaTeX? A main file that calls parts of an analysis or

Anyone have experience with Sipura's? Anyone know if they offer a warranty? Would like opinions on these, good or flame. We bought *one* to test with and it died, can't even get a response from Sipura "support". Could anyone recommend another device to replace these? Prefer 1 or 2 port design. Ty :-)

Bueller? Is this a lib of some kind? Google and lists bring up nada, this is from ast cvs head latest on Fedora Core 3. /usr/bin/ld: cannot find -lidn collect2: ld returned 1 exit status make[1]: *** [app_curl.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 [root@zoot asterisk]# uname -a Linux zoot 2.6.9-1.667smp #1 SMP Tue Nov 2 14:59:52 EST 2004

Hi All, I''ve downloaded a Puppet module and I''m trying to add a parameter to it by editing lib/puppet/type/<resource>.rb. I simply added: newproperty(:pcfree) do desc "My description here" end In the corresponding file in lib/puppet/provider, I do something with :pcfree. Whenever I call the resource from Puppet with my new parameter, it keeps

...at org.xapian.examples.SimpleIndex.main(SimpleIndex.java:44) It did not make any difference when I set LD_LIBRARY_PATH to the absolute "built" path either. I got the same results. I am using the Apple Java 5 J2SE. Has anyone gotten the bindings to work under OS X? -- john david schult usa : georgia : buford m: 404.510.5163

http://bugzilla.mindrot.org/show_bug.cgi?id=827 felix.schulte at gmail.com changed: What |Removed |Added ---------------------------------------------------------------------------- CC| |felix.schulte at gmail.com ------- You are receiving this mail because: ------- You are...

Hi there, I''m trying to setup Glassfish config management using puppet. I''ve found larstobi''s module here <https://github.com/larstobi/puppet-glassfish/>which I''ve cloned and am starting to tweak, as it would appear that the mentioned module is based on Solaris, therefore I''m starting to add support for EL. My code tree is

All, I'm not sure how to word this question but we're noticing a lot of our asterisk boxes no longer have multiple asterisk child processes. i.e. doing a 'ps ax' reveals only 1 asterisk PID when normally I'm used to seeing 8+ .. There is no rhyme or reason to it, and we're using the safe_asterisk script which has always worked in the past. Ast 1.2.4, zap 1.2.4, naturally..

Followed the "instructions" on voip-info.org regardinging fedora FC2, making Zaptel seems to work fine, however when I modprobe I get this. It looks like a version mismatch somehow. Ideas? If this ooc, sorry first post here :-) modprobe tor2 WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.358smp/misc/zaptel.ko): Invalid module format WARNING: Error inserting zaptel

Hi, I am trying to index a number of Spanish language text files, but a large fraction of the files are generating errors like the following... Error: exception 2 not handled: Error decoding input string. Check that you have the locale set correctly however it looks to me like my locale matches the file type. Running the file command on the files returns $ file

We are a voip terminating company, we're using Channelbank with FXS modules, Rhino, CAC, etc.. What we're wondering is, is how to would you echo cancel a channelbank. Of course we're realizing that cancel'ing on the T1 (on Ast) does no good (we think?) because the analog conversion is at the channelbank. Suggestions? Lowering the gain helps but we're looking for a real solution

Ok, we're trying to use Asterisk as a PBX in our office. Our original plan was to use a Cisco 7960 with a 7914 attached. Short story is, no one updated chan_sccp in a long time and the 7914 is questionable at best anyway from what I've heard. We couldn't ever get chan_sccp to compile, I went to an older version of Asterisk and that broke some of our SIP devices. We tried using a couple

All, Having problems terminating to a Nuvox PRI, the tech at Nuvox is saying Asterisk is transmitting in CCITT (aka ITU?) when they're expecting (and will only accept) ANSI. The question is, is there a simple way to change this or am I stuck with rewriting code? I googled and checked the mailing list and found nothing, I could be barking up the wrong tree I guess. PRI is not my forte.

I asked this question once before with no answer. Hopefully someone can help me as I cannot see a way to do this. I am wanting to differentiate inbound calls voice from FAX. The purpose of course voice gets g729 and FAX gets 711 (ulaw). The problem I'm having is everytime it matches the SIP peer (like it should) but it's always goes to the prefered codec. Anyone have suggestions on how to

I found a few mentions of the 7914 being used with Asterisk, these all covered SCCP/skinny though. Does anyone know if the 7914 can even be used with SIP? If so, any pointers? Is it a services thing? Anyone get the operator (line/extension status) to work with it. Thanks for the help, Cisco doesn't even mention ANYTHING about SIP + the 7914. Matt

I'm assuming asterisk does not have a SIP jitter buffer in place? Any ideas on how to help with this going over a data T1 where VoIP is shared with regular traffic? Problem is when people are downloading the voice is jittery, even lossy. Matt

I get this when adding a user in ser (using serctl) root@brak sbin]# ./serctl add +18165551212 blahblah blah@blah MySql password: error: 400; check if you use aliases in SER Um error 400?? I'm lost. no docs, frustrated. venting. Matt