Displaying 20 results from an estimated 6000 matches similar to: "VPN on Asterisk"
2007 Jun 17
7
VPN on Asterisk
Hi,
Greetings to All,
Im looking for some help on configuring VPN on the Asterisk PBX that I
have hosted in US. Im currently in Middle East and as everyone knows
some countries here has taboo to VOIP. Im not able to get phy phones
registered to my PBX as they are blocking SIP and IAX2. Hence im
looking for a VPN solution.
For this first i need to setup VPN on my server .. Am i right? Well if
2007 May 04
1
ASA-2007-013: IAX2 users can cause unauthorized data disclosure
> Asterisk Project Security Advisory - ASA-2007-013
>
> +----------------------------------------------------------------------------------+
> | Product | Asterisk |
> |----------------------+-----------------------------------------------------------|
> | Summary | IAX2
2007 May 04
1
ASA-2007-013: IAX2 users can cause unauthorized data disclosure
> Asterisk Project Security Advisory - ASA-2007-013
>
> +----------------------------------------------------------------------------------+
> | Product | Asterisk |
> |----------------------+-----------------------------------------------------------|
> | Summary | IAX2
2010 Oct 08
0
Weird stalling of playback on IAX2 channels on 1.8svn
On 8 Oct 2010, at 15:37, Danny Nicholas wrote:
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tim Panton
> Sent: Friday, October 08, 2010 9:34 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Weird stalling of playback on IAX2 channels
2010 Oct 08
2
Weird stalling of playback on IAX2 channels on 1.8 svn
I've hit an odd issue in a test 1.8 deployment,
playback() stalls mid file. The call stays up, but asterisk stops sending packets.
It doesn't always happen - but on demo-congrats it happens about half the time.
It only happens in IAX calls.
Anyone else experienced it ?
(I filed an issue just in case it isn't just me)
T.
Tim Panton - Web/VoIP consultant and implementor
2010 Aug 23
1
can't build resODBC on SUSE 11.3
What is menuselect actually looking for when it blocks me from selecting res_odbc ?
I've got unixOdbc installed and working. I also have /usr/lib64/libltdl.so.3 - so I'm confused
as it is claiming these are the pre-requisites ?
How can I best track down what it _thinks_ is missing ?
(This is on asterisk 1.8 svn trunk - but I don't think that is important,
I think it is a package
2006 Oct 11
1
1.4 beta2 on intel mac
Has anyone built and run asterisk 1.4 beta2 on an intel mac?
Did it work?
I've got it building ok (once I installed Xcode, wget and bison)
However Asterisk hangs on startup (halfway through loading the modules).
I have not (yet) had time to debug it, but I wondered if anyone else had
done this before me ?
Tim Panton
www.mexuar.com
2007 May 07
6
Could two Asterisk servers connect through VPN
Hi list:
Has anyone done to set up two servers in different remote offices
through VPN
in order to get the VoIP communication?
Thanks for your information.
Tielin Xu
2011 Jan 22
4
Crossover cable for E1 ?
I'm looking to connect a BMC 450 to an asterisk with a Digium Quad E1 card.
Am I right in thinking that I'll need a special 'crossover-E1' RJ45 cable?
If so, any clues where I might buy one in the UK? The Digium card sellers don't seem to
stock such a thing.
Thanks.
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
2006 Nov 13
3
FW: Desktop integration
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=UTF-8" http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000066">
Hi Dean,<br>
<br>
I will check that site - thanks for the hint.<br>
The biggest problem I see with
2009 May 09
5
Rusting Snoms?
This is a bit off topic, because I 'think' it isn't an Asterisk problem.
However I'm not sure and anyhow I'm hoping someone may recognize the
symptom.
We moved offices a month ago. Our trusty SNOM190s (all between 3 and 5
years old)
were packed up for the move, then unpacked a couple of weeks later.
On unpacking them and connecting them to the new network, several of
2011 Mar 08
3
Weird PRI error on an QUAD E1 span: Ring requested on unconfigured channel 255/255
After working fine for a week or so my new Quad E1 asterisk 1.8 system has started rejecting outbound calls from the Nortel
BMC 450 it is connected to.
The cli fills up with these:
sig_pri.c: Ring requested on unconfigured channel 255/255 span 3
Is this likely to be a
1) config error
2) cable issue (I made them)
3) hardware problem with the Digium card
4) software (lib pri)
Any clues?
2007 Jun 20
1
Res: Record CDR in a Oracle database
Hi All,
Thank's for your hint Tim Panton
I could connect my asterisk machine to my oracle machine.
I used unixODBC-2.2.11.tar.gz,
oracle-instantclient-basic-10.2.0.3-1.i386.rpm,
oracle-instantclient-sqlplus-10.2.0.3-1.i386.rpm and the drive from
www.oracle.com (odbc-oracle-3.1.0-linux-x86-glibc.tar) to configure my
asterisk machine.
I can connect to my oracle machine with isql and in
2011 Apr 01
0
Digium launches flying phone-phone
Huntsville 4/1/11
Digium inc today launched a new flying android desk phone.
The new handset includes a small low noise helicopter to assist in
call transfers.
The device has speech and gesture recognition, so when you don't want
to take a call you can either wave it away or say ("Kevin will answer that").
The flying handset has a megaphone and boom mic for priority call mode
2011 Mar 14
0
Anyone (else) need an asynchronous asterisk event->action framework ?
I did a google talk,skype, SIP, asterisk, IRC async event driven voice/IM mashup
for the voip user's conference - (see http://wp.me/pgOOh-4a for a description)
I've ended up with a thing that could (with some work) be turned into an asynchronous
asterisk event->action framework.
The basic premise is that you write filter terms to asynchronously match events and actions to carry out
2007 Sep 20
10
IAX Java Softphone?
Does anyone know of an IAX softphone in Java, whether applet or
application? Even the most minimum featureset, just voice and dialing,
or even embedded in some other app/let. Preferably GPL. Thanks.
--
(C) Matthew Rubenstein
2007 Feb 15
1
error during make while installing Linphone-1.5.1
Hi All,
I am getting this error during make.
please help me./
speexec.c: In function `speex_ec_process':
speexec.c:112: syntax error before "noise"
cc1: warnings being treated as errors
speexec.c:133: warning: implicit declaration of function
`speex_echo_state_reset'
speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes
pointer from integer without a cast
2008 Jan 18
2
Now that MS has to play nice...
Being that you SAMBA developers had to work so hard to reverse engineer
the AD protocols. Will there soon be improvements and more full
featured functionality in SAMBA now that you have access to more
documentation? Is anything on the order of a fully feature AD clone in
the works. Also, how do you dance around patented protocols? Can you
still implement them? Do you have to avoid them?
2007 Feb 05
3
Speex and RTP
> I believe there is new resampling functionality in the speex svn head,
> although I haven't tested it yet. You might also want to check out
> 'Secret Rabbit Code' for your resampling.
Yes, I've just been working on a resampler recently. Its changing a lot,
but it's now usable. I'd actually be quite happy to have some feedback
on it.
Jean-Marc
> Hope this
2006 Dec 12
3
outgoing call on ISDN PRI
HEllo list !
When user A calls user B via Asterisk (Users A and B are registered on
the same Asterisk server ) and an ISDN PRI, user B phone
always shows Asterisk server telephone number. How to hide it and how
to forward user A number ?
We tried usecallerid, callerid, hidecallerid, restrictcid,
usecallingpres in zapata.conf but we always see Asterisk server
telephone number !
Thanks