similar to: ices2 - memory leak

> 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, 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 =======================================

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

> 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

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

-----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

XXX No documentation. Only handles one disk. Network cards? Do we need to escape YAML format? What firmware types does kubevirt support. --- v2v/Makefile.am | 2 + v2v/cmdline.ml | 21 ++++++++++ v2v/output_kubevirt.ml | 103 ++++++++++++++++++++++++++++++++++++++++++++++++ v2v/output_kubevirt.mli | 24 +++++++++++ 4 files changed, 150 insertions(+) diff --git

Dne 28.1.2016 v 13:37 Brian :: napsal(a): > when you say load - how many concurrent calls? Is there transcoding > happening? sip / PRIs ? what load? > 12 concurrent calls no transcoding SIP under 1.5 with 4x 1Ghz vcpus (its vmware VPS) > On Thu, Jan 28, 2016 at 9:57 AM, Marek ?ervenka <cervajs at fpf.slu.cz > <mailto:cervajs at fpf.slu.cz>> wrote: > >

Dne 13.8.2015 v 17:20 Rusty Newton napsal(a): > On Thu, Aug 13, 2015 at 3:54 AM, Marek ?ervenka <cervajs at fpf.slu.cz > <mailto:cervajs at fpf.slu.cz>> wrote: > > hello, > > is it possible simultaneously use chan_sip and chan_pjsip? > > if yes, can you recommend settings > > i'm thinking about > - chan_sip - for sip

2016-02-19 12:01 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz>: > on my own server > Today, I'm back from holidays trip. First of all, thanks for replying ! I'll try to use jssip as you suggested. Anyway, I'm still failing to understand if wiki's page [1] is still valid with Asterisk 13, and if it's not valid anymore, which is the main change that prevent

2016-02-18 15:42 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz>: > my experience with pjsip for webrtc > http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html > > > Yes I saw this post earlier today. Having to fight 14 days scared me a bit ! Did you set sipml5 on your own server or did you use Live demo (

This was always possible in the past, however does not work in the current release. I believe this is a bug. To: asterisk-users at lists.digium.com From: cervajs at fpf.slu.cz Date: Fri, 9 Oct 2015 10:04:47 +0200 Subject: Re: [asterisk-users] Storing HANGUPCAUSE in CDR search in archives save the records to another table like cdr_extended Dne

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'

Fixes some of the more egregious problems with v1, and also applies properly to the head of git without needing any other patches. Rich.

Previously the kernel_info field 'ki_supports_virtio' really meant that the kernel supports virtio-net. That was used as a proxy to mean the kernel supports virtio in general. This change splits the field so we explicitly test for both virtio-blk and virtio-net drivers, and store the results as separate fields. The patch is straightforward, except for the change to the

These safe wrappers around Pervasives.open_in and Pervasives.open_out ensure that exceptions escaping cannot leave unclosed files. --- common/mlstdutils/std_utils.ml | 39 ++++++++++++++++++++-------------- common/mlstdutils/std_utils.mli | 12 +++++++++++ common/mltools/tools_utils.ml | 39 +++++++++++++++++----------------- dib/dib.ml | 9 ++++----

On Monday, 28 November 2016 10:40:51 CET Richard W.M. Jones wrote: > Create a new directory (builder/template). Integrate all of the > scripts into a single program, so that templates are generated more > consistently. > > This also changes how the index file is generated. The script now > generates the index file fragment and saves it under version control, > and then

Create a new directory (builder/template). Integrate all of the scripts into a single program, so that templates are generated more consistently. This also changes how the index file is generated. The script now generates the index file fragment and saves it under version control, and then generates the final index file by concatenating these. (Previously the index was written by hand which was