search for: naturalnet

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. -------------- n?chster Teil -------------- Ein Dateianhang mit HTML-Daten wurde abgetrennt... URL:
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