I'm having a problem with IAX running through an intermediate asterisk
box. Perhaps a small diagram will explain the situation better:
*A --- [cloud (public internet)] --- *B [cloud
(private network)]--- *C
Asterisk server's A, B, and C, are all connected together with
So no one else has a problem routing IAX traffic through an
intermediate Asterisk server? Does anyone else use Asterisk in such a
configuration?
On Thu, Mar 26, 2009 at 2:45 AM, Andrew Hakman andrew.hak...@gmail.com wrote:
I'm having a problem with IAX running through an intermediate asterisk
:
On Fri, Mar 27, 2009 at 12:50 AM, Andrew Hakman andrew.hak...@gmail.com
wrote:
So no one else has a problem routing IAX traffic through an
intermediate Asterisk server? Does anyone else use Asterisk in such a
configuration?
On Thu, Mar 26, 2009 at 2:45 AM, Andrew Hakman andrew.hak...@gmail.com
if the problem goes away.
On Thu, Mar 26, 2009 at 10:50 PM, Andrew Hakman andrew.hak...@gmail.com
wrote:
So no one else has a problem routing IAX traffic through an
intermediate Asterisk server? Does anyone else use Asterisk in such a
configuration?
On Thu, Mar 26, 2009 at 2:45 AM, Andrew Hakman
Hey now, I'm a newschool programmer and I use vim (and vi, when necessary).
Andrew
On Wed, Oct 21, 2009 at 8:02 PM, Jeff LaCoursiere j...@jeff.net wrote:
Steve Edwards asterisk@sedwards.com wrote:
Since I'm an old-school C programmer, I use emacs as my editor. I fire
up gdb (the GNU C
On Wed, Nov 4, 2009 at 1:44 PM, Joseph syscon...@gmail.com wrote:
On 11/04/09 15:20, Adam Tauno Williams wrote:
I have two of these and experience a lot of echo on PSTN line (FXS line works
OK).
The echo is almost impossible to get rid of, so test it first before you buy
this unit; Google
on two different PSTN lines and I have no echo
problems with them.
Andrew
On Wed, Nov 4, 2009 at 5:07 PM, Joseph syscon...@gmail.com wrote:
On 11/04/09 16:01, Andrew Hakman wrote:
On Wed, Nov 4, 2009 at 1:44 PM, Joseph syscon...@gmail.com wrote:
On 11/04/09 15:20, Adam Tauno Williams wrote:
I
Hi
Is there anyway to add logic to dialplan pattern matching? I would
like to match all toll free numbers with one pattern, so 1800, 1877,
1866, 1855, etc. I can't figure out how to do this in dialplan syntax.
As a programmer, I want to say 18[00 or 77 or 66 or 55 etc]. Can't
figure out if this
The SPA-3000 is notorious for falsely detecting DTMF tones in regular
voice, and when it thinks it hears DTMF, it will produce a short
real DTMF tone that's only audible to the SIP side of the device, not
the PSTN side, or out of band SIP DTMF message (dependent on how you
have the device setup).
I'm having a strange problem with a sip client and 2 asterisk servers
connected together with a sip trunk. Here's a rough layout
sip_client -- Asterisk A -[sip trunk] -- Asterisk B
when the sip client tries to dial an extension on Asterisk B, Asterisk
A sends the invite to B using
Are you using openvpn? If so, there's an option in the server config
file that allows vpn clients to talk to other vpn clients, otherwise
they can only talk to the server. Using canreinvite=no is just forcing
the traffic to go through the server, which is why that makes it work.
I must say VPN +
Windows, yes, but used to be through 3rd party software. Doubt this
has changed as Windows has no focus on any useful network anything.
Linux, yes, and it's definitely not complicated. Probably take 2
minutes to setup if you already had bridge utils installed, maybe 5 if
you had to install the
You probably are not advertising the routes across the vpn properly.
Does your setup look like this
asterisk[network a]openVPN server[network b -
vpn]-openVPN client[network c]-sip client
where network a, b, and c are all separate subnets?
Is your vpn setup for routing
On Thu, Feb 11, 2010 at 6:37 AM, mosbah.abdelkader
mosbah.abdelka...@gmail.com wrote:
Hello,
I have the following situation: A firewall is blocking all SIP and RTP
traffic in the side of some of my clients. My clients cannot change settings
of the firewall.
I need to solve this problem and
On Tue, Mar 23, 2010 at 1:41 PM, Alejandro Cabrera Obed
aco1...@gmail.com wrote:
Dear all, I have an Asterisk SIP server in a LAN environment and I want your
opinion in order to decide the use of an audio codec:
What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip
I use openvpn for VOIP traffic all the time. It's not a commercial
application, and only one simultaneous call usually on each vpn link,
but I even have a VPN client on a Linksys WRT-54g wireless router with
1 phone behind it - it works flawlessly, so it does not take a lot of
CPU to run a vpn
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