Displaying 20 results from an estimated 200 matches similar to: "SRTP testers needed"
2007 Mar 20
4
blktap howto
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'',
2007 May 01
0
Re: [asterisk-dev] SRTP implementation
> 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
2004 Aug 06
2
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?
(im newbie to debugging)
--
2004 Aug 06
0
Re: 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?
2005 May 23
1
Grandstream GXP-2000 headset
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
2005 Feb 27
2
[Asterisk-Dev] Asterisk 1.0.6
-----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
2004 Dec 20
3
PA1688 Chipset IP Phones & ATA's
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
2006 May 09
1
grandstream GXV-3000
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
=======================================
2010 Jul 22
3
Soft phones.
Hey, all. I'm looking -- if possible -- for a decent, multi-platform
soft-phone. Specifically, Linux and Windows; that way, I'll go through
the same issues my end users do. I've noticed a couple (e.g., minisip,
which seems abandoned, and sip-communicator, which, honestly, is probably
a great IM client, but has a confusing interface for actual phone calls).
So I'm wondering if
2011 Oct 05
1
call pickup
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
2007 Dec 25
1
Softphone to be installed on the Mobile
Thanks a lot for the help.
But if Asterisk has private IP address and the only
way to access it from remote sites is to have vpn
connection to the site that asterisk existed (the site
has vpn), then how that will happen from the Mobile to
be able to run the softphone from the mobile?
Any help?
Regards
Bilal
-----------------
I installed out of curiosity today, and guess what?
You can do SIP
over
2009 Apr 14
0
SRTP testers needed
please look at
http://www.voip-info.org/wiki/view/Asterisk+SRTP
and try compile&run clients with srtp (linksys,grandstream,aastra,
qutecom, eyebeam, ...)
digium need feedback for srtp inclusion to 1.6.3.0
http://bugs.digium.com/view.php?id=5413
if you need additional info, i'm on jabber - cervajs at njs.netlab.cz
thanks!
---------------------------------------
Marek Cervenka
2006 Jun 14
7
open source sip softphone (Window OS version )
are there any open source sip softphone (Window OS version )?
2004 Dec 20
7
'I'nvalid extension handling problems, even with workaround
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'
2009 Feb 12
1
NUT and Buildbot (was Re: New Mustek UPS model working)
On Wed, Feb 11, 2009 at 11:30 AM, Fr3ddie <fr3ddie at fr3ddie.it> wrote:
> Charles Lepple wrote:
>> When something gets updated, you would then run 'svn update' to pull
>> the latest changes in.
>
> Ok.
>
>> The dependencies are set up such that if only one driver is changed
>> during the SVN update, running 'make' on that directory should
2006 Oct 11
10
GPL Softphones
Hi,
I'm searching for GPLed softphones. I found WengoPhone but actually not
available for Asterisk PBX, only for Wengo network. I found Kiax but only
for IAX protocol.
Did you know a good GPLed softphones which works on Windows ?
Thanks
Greg
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2007 Nov 21
5
Softphone to be installed on the Mobile
Hi All;
Is there a softphone that can be installed on a mobile
(new mobile models), so it can work with Asterisk as
following:
1) As SIP or H323 client, with the ability to add
button functionalities (call pickup, call transfer,
...) so if there is a wireless network, then it can
use it to connect to Asterisk and work as client, but
from the Mobile.
2) If there is no wireless network, then it
2018 Mar 27
1
[PATCH FOR DISCUSSION ONLY] v2v: Add -o kubevirt output mode.
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
2005 Jun 27
8
OT: Good soft-phone on Linux
Hi Folks,
I am wanting advise on a good soft-phone on Linux. I have looked at
Gnophone but cannot seem to get it to compile under debian sarge. I am
now looing at sipXphone seem to be picking up that it is not that
stable, but perhaps someone here can advise on what softphone I can use
on Linux.
Thanks in advance,
Hamish
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2017 Apr 06
0
[PATCH v4 3/9] v2v: linux: Replace 'ki_supports_virtio' field.
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