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hi, i''m trying move from file: based disk to tap:aio but things don''t work i have centos4 dom0 with centos4 domU xen 3.0.4-testing changeset: 13138:d401cb96d8a0 self compiled [root@xen linux-2.6.16.38-xen]# grep XEN_BLKDEV_TAP .config CONFIG_XEN_BLKDEV_TAP=m config disk = [ ''file:/var/lib/xen/test.img,hda1,w'',

please look at http://www.voip-info.org/wiki/view/Asterisk+SRTP and try compile&run clients with srtp (linksys,gxp-2000, minisip, twikle, ...) --------------------------------------- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA - http://lcna.slu.cz =======================================

hi, i have rh72 systems + updates libvorbis, libogg, vorbis-tools (xslt,xml2) recompiled rpm from rh8.0 ices2 klient celeron 1.Ghz 512RAM icecast2 server duron 700Mhz 256RAM 100Mbps network 4 streams 128 kbs ogg from playlist(random) i have noticed memory leaks in ices2 (randomly) what type of info do you need to correct this? (im newbie to debugging) --

hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? thanks -- --------------------------------------- Marek

Hi, I have this simple program: #include <regex> int main() { std::regex re; re.assign(std::regex("foo")); // SIGILL return 0; } It runs smoothly if compiled with g++ but raises "illegal instruction" when compiled with clang++: g++ -std=c++11 -O0 -g -o test-g++ test.cpp clang++ -std=c++11 -O0 -g -o test-clang++ test.cpp ptomulik at barakus:$ ./test-g++ ptomulik

> hi, > > i have rh72 systems + updates > libvorbis, libogg, vorbis-tools (xslt,xml2) recompiled rpm from rh8.0 > ices2 klient celeron 1.Ghz 512RAM > icecast2 server duron 700Mhz 256RAM > 100Mbps network > > 4 streams 128 kbs ogg from playlist(random) > > i have noticed memory leaks in ices2 (randomly) > > what type of info do you need to correct this?

Hi all What headset do people use with the GXP-2000? Any recommondations for or against particular models? Thanks Peter -- Peter Bowyer Email: peter@bowyer.org Tel: +44 1296 768003 VoIP: sip:peter@bowyer.org

> Olle E Johansson wrote: >> >> 23 apr 2007 kl. 19.55 skrev Russell Bryant: >> >>> John Todd wrote: >>>> To morph this into a -dev thread: if this patch were to become (again) >>>> useful and error-free, is there any objection or usefulness in adding it >>>> to TRUNK? Personally, I think there is, if there is a method by which

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Greetings Everyone! Version 1.0.6 of Asterisk, zaptel, libpri, and Asterisk-addons has been released. There is also a new tarball for Asterisk-sounds. They are available for download on the digium FTP site: ftp://ftp.asterisk.org/pub/asterisk/ ftp://ftp.asterisk.org/pub/zaptel/ ftp://ftp.asterisk.org/pub/libpri/ ChangeLogs are available with the

For those of you who may be interest.... IAX2 loads are now available for the standard builds... http://www.aredfox.com/edownloadsiax2.htm Just a word of caution... Remember to change the ports over to 4569 from whatever. And don't forget to grab the palmtool from http://www.aredfox.com/download/tools/PalmTool.zip My own testing of IAX2 with both a phone and an ATA is that IAX2 is

Hello folks, I'm having trouble configuring Asterisk to play an "invalid extension" message to anyone dialing an undefined extension. First I tried using the 'i' pseudo-extension, but it didn't work at all; searching the wiki I found that page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension where it basically says that the 'i'