similar to: VoIP over virtualized VPN

Displaying 20 results from an estimated 9000 matches similar to: "VoIP over virtualized VPN"

2010 May 05
1
Getting calee audio in Asterisk (real time)
Hello, I need to capture calee's audio in real-time in order to capture operator messages (I've written sound recognition software that works with Jack: http://github.com/Motiejus/SoundPatty/). Jack does the following: Incoming call audio -> audio in to jack, audio out from jack -> current Asterisk application Outgoing call audio <- current Asterisk application However, I need
2010 May 26
1
Jack in /usr/local/ means failure for asterisk
Hi, I have been headbanging with asterisk and Jack for a while, decited to ask other linuxists for an advice. The problem is that Jack is compiled from source (0.118) in /usr/local/, but menuselect says "XXX" for it (cannot enable it). I need jack... Otherwise I will inotify Monitor WAVs, what is bad :`( After installing jack from sources: Add system-wide
2010 May 21
2
Connecting 1-2 GSM ports to asterisk?
Hi, List, I am looking for a cheapest (and therefore most funny) way to attach GSM card to my asterisk home box. Needed features: Calls+SMS in/out one or two SIM cards (ports) Should I try looking for a GSM PCI card that is compatible with linux/asterisk, or GSM USB card, or modern full-blown SIP GSM gateway (with ethernet)? Maybe an ordinary cell phone with USB interface and mangling with
2013 Feb 03
4
CentOs 6 DHCP Server and virtual interface
Hello All I have looking for any specific answer for one thing. I have a virtualized Server with only one physical interface eth0 (WAN). To run OpenVPN i need to use DHCP server. And here is the question: is there a chance to run DHCP server on eth0:0 interface? Or it is impossible ? Thanks in advance. -- /Best Regards *Greg*/ UML Professional (Cert #251574932)
2010 May 11
2
Creating a HTTP Request on missed call?
Hello there, I have successfully installed and configured asterisk for use as an office PBX using SIP trucks and Voip handsets (using g.729 codec) which works great. Now I wish to try and configure asterisk to do a HTTP request and submit callerID to an external website when a call is missed. eg Someone calls PBX and rings extension 100 -> Call is not answered -> HTTP request is initiated
2010 Aug 08
3
How to track a call result originated from originate AMI command
Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were
2010 Aug 04
2
Identify remote prompts: Partial audio matching?
Ok, here's the challenge: I would like to be able to find, match - and then react - upon prompts that are presented by the outbound/remote side of a call. Think mobile phone and "This user is temporarily unavailable". Collecting a limited number of known prompt snippets should not be a problem, but how would you then detect their presence in a longer recording (or live audio
2020 Mar 14
4
Q: Samba AD, Pfsense, Windows 10, vpn
Your pfSense firewall has OpenVPN built into it already, and you can point pfSense authentication back to your samba AD. We support over 400 users in this model. The configuration file for OpenVPN is common to all users, and they authenticate with their AD credentials. > On Mar 14, 2020, at 7:21 AM, Michael Howard via samba <samba at lists.samba.org> wrote: > > On 14/03/2020
2015 May 05
2
OpenVPN Clients Intermittently Cannot Call In
----- Original Message ----- > From: "Administrator TOOTAI" <admin at tootai.net> > To: asterisk-users at lists.digium.com > Sent: Friday, May 1, 2015 6:42:38 AM > Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In > > Le 01/05/2015 00:05, Andrew Martin a ?crit : > > ----- Original Message ----- > >> From:
2015 Oct 18
2
Feasibility of Tinc vpn with my current setup.
Hi all, I am new to Tinc VPN and really would like to make full benefit of this implementation if possible. I would like to know whether I will be able to use Tinc to its full potential. My current setup is as follows, IPfire router/firewall(openvpn client) --->ISP(Internet)--->Amazon VPS(openvpn server). The ipfire router is behind a CARRIER-GRADE NAT, I am able to reach the network
2016 Apr 04
1
VPN suggestions centos 6, 7
On 04/04/2016 12:11 PM, Jussi Hirvi wrote: > This made me google around a little, and I found some good info here. > They, too, kind of recommend openvpn. > http://www.howtogeek.com/211329/which-is-the-best-vpn-protocol-pptp-vs.-openvpn-vs.-l2tpipsec-vs.-sstp/ > This is not good information. In brief: "There are some concerns that the NSA could have weakened the standard,
2006 Nov 12
2
ipsec-tools with cisco vpn client
Hi, anybody successfully running win32 client with Cisco vpn client against ipsec-tools? I'm looking for elegantly running VPN road warrior solution. Scenarios are: - ipsec-tools with Cisco vpn client - pptpd with Windows XP native client - OpenVPN with OpenVPN Windows client - ??? Any hints? Thanks for reply. David Hrb??
2020 Mar 15
2
Q: Samba AD, Pfsense, Windows 10, vpn
> Am 15.03.2020 um 08:21 schrieb S?rgio Basto via samba <samba at lists.samba.org>: > > ?On Sat, 2020-03-14 at 07:43 -0700, gabben via samba wrote: >> Your pfSense firewall has OpenVPN built into it already, and you can >> point pfSense authentication back to your samba AD. We support over >> 400 users in this model. The configuration file for OpenVPN is common
2004 Dec 30
12
Multi-Hop VPN Issue looking for Solutions
I''ve just discovered that I do not have access to the remote gateways for a set of IPsec tunnels to remote networks. This prevents me from changing the routing table on those gateways. I need "roadwarrior" systems connecting to me local network using OpenVPN (tun) to be able to access those systems. Since the remote gateways don''t know about 10.100.1.0/24, where my
2010 Aug 05
2
AMI Command
Hi, Is there a way to check on AMI if a user is currently engage on the phone? i would like to display on my portal whether a user is calling or not. thank you regards Ron
2016 Apr 04
2
VPN suggestions centos 6, 7
And openvpn. Avoid ipsec as it's too complex and pptp is unsecure. Eero 4.4.2016 9.55 ip. "Richard Zimmerman" <rzimmerman at riverbendhose.com> kirjoitti: > SoftEther VPN > > Once setup, it just works.... > > Regards, > > Richard > > > --- > Richard Zimmerman > Systems / Network Administrator > River Bend Hose Specialty, Inc. > 1111 S
2016 Apr 05
7
VPN suggestions centos 6, 7
IPSec is not recommended solution nowdays. OpenVPN runs top of single udp or tcp port, so it usually works on strictly firewalled places like in hotels and so on. -- Eero 2016-04-04 23:18 GMT+03:00 Gordon Messmer <gordon.messmer at gmail.com>: > On 04/04/2016 10:57 AM, david wrote: > >> I have seen discussions of OpenVPN, OpenSwan, LibreVPN, StrongSwan (and >> probably
2008 Nov 24
1
PPTP VPN server
Hi I've been using linux to give VPN access to my corporate LAN using the following software: Centos 5.2 x86 kernel 2.6.18-92.1.18.el5xen pptpd (poptop) 1.3.4 ppp 2.4.4 The Centos server has directly connected the Internet Router, on one interface (eth1) and the LAN on another (eth0) and it works as the firewall/VPN server of my LAN. It mostly works, however, if I try to connect using
2007 May 02
2
VPN between Asterisk server and phone client
> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Biju > Sent: Wednesday, May 02, 2007 5:38 AM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [asterisk-users] VPN between Asterisk server and phone client > > Hi, > > I wish to make a secure
2010 Aug 13
2
realtime sip peers : musiconhold class
Hello list, I'm using asterisk 1.4.30 and realtime sip. I notice that the field "musiconhold" is not working as when putting someone on hold, the default musiconhold class is always used. musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; [106002] mode=files directory=/var/lib/asterisk/moh/106002 random=yes my realtime sip peers have the