Displaying 8 results from an estimated 8 matches for "naturalnet".
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2013 Oct 30
4
[Bug 71035] New: EVO engine failure, probably (not?) related to EDID corruption
...nouveau at lists.freedesktop.org
Summary: EVO engine failure, probably (not?) related to EDID
corruption
QA Contact: xorg-team at lists.x.org
Severity: normal
Classification: Unclassified
OS: Linux (All)
Reporter: nik at naturalnet.de
Hardware: x86-64 (AMD64)
Status: NEW
Version: unspecified
Component: Driver/nouveau
Product: xorg
Created attachment 88342
--> https://bugs.freedesktop.org/attachment.cgi?id=88342&action=edit
dmesg output
** Slightly modified versio...
2013 Nov 16
3
Make phone ring through webserver using Asterisk
What is the easiest way? And how can it be implemented?
I thought to something like:
1. I request a page to the webserver
2. Perl sends to asterisk a number to dial (Perl and asterisk are
running in the same machine)
3. Asterisk calls the phone
or
1. A Perl sip client registers to remote asterisk server
2. Perl sip client sends to asterisk the number to dial
3. Phone rings
2013 Oct 17
4
MusicOnHold starts magically for no reason
Dear list,
on Asterisk 1.4.21 which is being used in a callthrough scenario -
callers call via PSTN to a DID coming in via SIP and then dialing
outbound via DTMF and the outbound calls get routed via some SIP
termination provider - lately I see that every now and then MusicOnHold
gets triggered like this on outbound calls:
Started music on hold, class 'default', on
2013 Oct 01
2
is g729 codec free? or under license???
hello all,
i have problem in using g729 codec. my asterisk version is 1.8.22. when i
run "core show codecs" in asterisk, there is a g729 codec in the list so i
assume that i can use it for my channels. but connection can not be set
when i use it for my h323 channel.
i read somewhere that codec g729 is a commercial codec and i should buy its
license in order to use it. is it true? if
2013 May 14
0
Call Diversion Override
Hi,
for a call routing setup with my mobile phone, I'd like to set the CDO feature on an outgoing SIP call. I know my SIP proxy provider will pass the call to a Teles SS7 gateway, keeping most of it intact.
The goal is to forward all calls from the mobile phone number to Asterisk, which will pass it back should I be unavailable there. Upon passing it back to the mobile number, I of course
2013 Aug 04
1
UDP hole punching and invitations
Hi,
I read about the invitation protocol that will be introduced in 1.1pre8.
Is there any mechanism of UDP hole punching to establish a VPN behind NAT?
Cheers,
Nik
--
Diese Nachricht wurde von meinem Android-Mobiltelefon mit K-9 Mail gesendet.
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2013 Nov 10
2
Not seeing any more LocalDiscovery broadcasts
Hi,
I am playing with LocalDiscovery again and have noticed that I do not
see any LocalDiscovery broadcasts anymore.
I am using tinc 1.1-pre9 in switch mode and have set LocalDiscovery =
yes in tinc.conf. I do not see any broadcasts on any network and I also
do not see anything in the debug output.
What to do?
-nik
--
# apt-assassinate --help
Usage: apt-assassinate [upstream|maintainer]
2013 Apr 04
2
LocalDiscovery detecting nodes through tunnel
Hi,
I have tried the LocalDiscovery feature of tinc.
The problem is that it also sends broadcast probes out the CPN interface
*and* detects nodes on the VPN. A connection is then established through
the tunnel, which effectively breaks connectivity between the two nodes.
I do not think that discovering hosts on the VPN makes sense in any way.
How can it be disabled?
I could easily netfilter