On Fri, Jan 26, 2007 at 12:34:30PM +, Tim Panton wrote:
Unless you are monitoring calls, want full CDR etc,
then that's what you want anyway.
CDR are not affected by how the audio flows.
While technically true, I believe (it may have changed in 1.4) that if you
allow reinvites, the
On Sat, Jan 27, 2007 at 01:55:31PM -0500, Jerry wrote:
On Fri, Jan 26, 2007 at 12:34:30PM +, Tim Panton wrote:
Unless you are monitoring calls, want full CDR etc,
then that's what you want anyway.
CDR are not affected by how the audio flows.
While technically true, I believe
On 25 Jan 2007, at 06:57, Brad Templeton wrote:
On Mon, Jan 22, 2007 at 09:59:06AM +, Tim Panton wrote:
In the meanwhile, use IAX, which understands about NAT pretty well.
If you have multiple SIP phones on a LAN behind a NATing router, just
put a small asterisk box on the LAN. It can
, January 25, 2007 11:19 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] NAT solutions
From: Brad Templeton [EMAIL PROTECTED]
I have a really dumb question. It appears that Yahoo, MSN, AIM, you
name
them, they don't have a NAT problem, and some use SIP. I don't
think
On Thu, 25 Jan 2007, Yuan LIU wrote:
Thanks for this information. Does this mean two IAX boxes can talk behind
their respective NAT's (without any server sitting in voice path)? I'm
imagining this:
Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2
Using IAX, yes. It's quite
Gordon Henderson wrote:
On Thu, 25 Jan 2007, Yuan LIU wrote:
Thanks for this information. Does this mean two IAX boxes can talk
behind their respective NAT's (without any server sitting in voice
path)? I'm imagining this:
Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2
Using
On 26 Jan 2007, at 06:19, Yuan LIU wrote:
From: Brad Templeton [EMAIL PROTECTED]
I have a really dumb question. It appears that Yahoo, MSN, AIM,
you name
them, they don't have a NAT problem, and some use SIP. I don't
think they
all stay in voice path, either. What takes?
When you
From:"Ken Williams" [EMAIL PROTECTED]Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2 is one ofthe easiest configs to put together.Works extremely well and requiresopening a single port on each NAT.
Now I realize that I took the wrong assumption that all NAT traversal is blind traversal.
On Thu, Jan 25, 2007 at 10:19:06PM -0800, Yuan LIU wrote:
Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2
If Asterisk1 can talk to Asterisk2 at trunk level, I'll be happy.
While I'm not sure of what tricks * plays at all levels, you
can certainly make this work if you have control of
On Fri, Jan 26, 2007 at 12:34:30PM +, Tim Panton wrote:
For a remote phone, not on the same network as the Asterisk
box (in which event the NAT worries are different) you definitely
want to use the same protocol for the phone as for your
term/orig provider. Otherwise you will be forced
On Wed, 24 Jan 2007, Yuan LIU wrote:
I have a really dumb question. It appears that Yahoo, MSN, AIM, you name
them, they don't have a NAT problem, and some use SIP. I don't think they
all stay in voice path, either. What takes?
Their SIP servers aren't behind NAT firewalls, so the problem
-Original Message-
Gordon Henderson
Sent: 25 January 2007 08:17
On Wed, 24 Jan 2007, Yuan LIU wrote:
I have a really dumb question. It appears that Yahoo, MSN, AIM, you
name them, they don't have a NAT problem, and some use SIP. I don't
think they all stay in voice
On Wed, Jan 24, 2007 at 11:09:21PM -0800, Yuan LIU wrote:
From: Brad Templeton [EMAIL PROTECTED]
On Mon, Jan 22, 2007 at 09:59:06AM +, Tim Panton wrote:
In the meanwhile, use IAX, which understands about NAT pretty well.
If you have multiple SIP phones on a LAN behind a NATing router,
From: Brad Templeton [EMAIL PROTECTED]
I have a really dumb question. It appears that Yahoo, MSN, AIM, you
name
them, they don't have a NAT problem, and some use SIP. I don't think
they
all stay in voice path, either. What takes?
When you control both ends of the path, you can eliminate
On Mon, Jan 22, 2007 at 09:59:06AM +, Tim Panton wrote:
In the meanwhile, use IAX, which understands about NAT pretty well.
If you have multiple SIP phones on a LAN behind a NATing router, just
put a small asterisk box on the LAN. It can manage your hairpin
calls internally, save you
From: Brad Templeton [EMAIL PROTECTED]
On Mon, Jan 22, 2007 at 09:59:06AM +, Tim Panton wrote:
In the meanwhile, use IAX, which understands about NAT pretty well.
If you have multiple SIP phones on a LAN behind a NATing router, just
put a small asterisk box on the LAN. It can manage your
On 21 Jan 2007, at 07:55, Brad Templeton wrote:
Some NAT problems you can solve, some you never will.
Many modern phones have NAT support in them, via STUN, or a static
external IP
address. Most NATs also offer port forwarding, so you can open a
hole for the
SIP port in the NAT so all
Some NAT problems you can solve, some you never will.
Many modern phones have NAT support in them, via STUN, or a static external IP
address. Most NATs also offer port forwarding, so you can open a hole for the
SIP port in the NAT so all outside can reach it.
(With port forwarding, you need a
On Thu, 18 Jan 2007, Voip Asterisk wrote:
I know that NAT is something no one really likes to talk about, but does
anyone know how work with it elegantly? There are many providers which deal
with it on a daily basis in fact they cater to it, is this possible to do
with asterisk or does it
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Gordon Henderson wrote:
If you only have one * box behind the NAT gateway then I don't really
see a big issue with it to be honest. Port-forward on the
firewall/router device (5060 and 1 through 2) to the * device,
and use STUN on the
Bernardo Vieira wrote:
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Gordon Henderson wrote:
If you only have one * box behind the NAT gateway then I don't really
see a big issue with it to be honest. Port-forward on the
firewall/router device (5060 and 1 through 2) to the *
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Bernardo,
Just a thought: Try using a different SIP port for one of the
extensions, if possible.
Bob...
Bob,
Tanks for the tip. I had actually done that before, as a matte of fact
that's the solution I have in place now. The thing is, even
Bernardo Vieira wrote:
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Hash: SHA1
Bernardo,
Just a thought: Try using a different SIP port for one of the
extensions, if possible.
Bob...
Bob,
Tanks for the tip. I had actually done that before, as a matte of fact
that's the solution I have in
I know that NAT is something no one really likes to talk about, but does
anyone know how work with it elegantly? There are many providers which deal
with it on a daily basis in fact they cater to it, is this possible to do
with asterisk or does it require other exotic setups? I even know of a
Voip Asterisk wrote:
I know that NAT is something no one really likes to talk about, but
does anyone know how work with it elegantly? There are many providers
which deal with it on a daily basis in fact they cater to it, is this
possible to do with asterisk or does it require other exotic
What about open sip stack:
http://www.opensipstack.org/
?
Use a far end nat traversal appliance. Acmepacket , kagoor and Jasomi
are some examples.
Leo
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