similar to: Re: [asterisk-dev] SRTP implementation

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''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'',

> 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 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) --

hi, do you someone test this http://www.grandstream.com/y-gxv3000.htm? video works? (it's have H264 video codec) i want this topology gxv-3000 - nat -{Internet}- Asterisk -{Internet}- nat - gxv-3000 --------------------------------------- Marek Cervenka LCNA - http://lcna.slu.cz =======================================

Hi! I am having trouble getting the following configuration to work: PHONE1 <-- rtp --> Asterisk <--IAX--> Asterisk_SRTP_1 <--- srtp ---> Asterisk_SRTP_2 <-- rtp--> PHONE2 This means, I am using regular voip clients without srtp support on both sides, but the communication between the 2 Asterisk_SRTP boxes must be secure. The Asterisk_SRTP_2 box is registered in the

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

Asterisk Version: 1.8.1.1 Problem: Multiple Parking Lots Issue: Not redirecting to the right parking lot. Always uses the first parking lot from "parkedcalls show" or "features show" Asterisk Working Version: 1.6.1 Steps Taken: In features.conf added: [parkinglot_test] context => parkedcalls-test parkext => 700 parkpos => 701-710 parkingtime => 120 findslot

I'm in the process of testing a TLS/SRTP install. My experience is improving with each new challenge, but this one is a great test of my 2 month experience with Asterisk. When I dial 6003 from 6001, it takes 35 seconds until I get the error message that 6003 is circuit-busy. Any help would greatly be appreciated. Below is the error message and the extensions and sip.conf files. *CLI>

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 ++++----

Firstly I apologise for the length of this query but I am hoping that if I document everything I did someone might respond / be able to help. My Configuration is Samba 3.0.2a as a PDC on Redhat 8. I cannot for the life of me get the "Domain Admins" functionality to work I am hoping that another set of eyes can shed some light on this problem as I have now spent 41 hrs googling /

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

Hi all, I upgraded one of our servers running asterisk 1.6.X to 1.8.2.3. I compiled it with SRTP support. Everything seems to work OK but I am having a weird issue. I cannot disable SRTP. I tried the /encryption=no/ in /sip.conf /and the /_SIPSRTP_CRYPTO=disable/ on my dailplan and it keeps trying to use the SRTP. Well, right now I have to have/ noload=res_srtp.so/ on my /modules.conf /otherwise

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

Sorry all, forgot to hit reply-all. ---------- Forwarded Message ---------- Subject: Re: [Ocfs2-users] Determining which version of ocfs2 tools a filesystem was created with. Date: Thursday 17 February 2011, 12:33:36 From: Mikey Austin <mikey at mikeyaustin.com> To: Sunil Mushran <sunil.mushran at oracle.com> On Wednesday 09 February 2011 11:40:01 you wrote: > On 02/07/2011

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm running Asterisk 11.3.0 on wheezy. I'm trying to do TLS +SRTP with blink SIP clients as shown here https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial. TLS is fine and I can call between clients. SRTP is a different matter, my SIP clients return: SIP 488 "Not acceptable Here" I'm really stumped on this

-----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