Gordon Henderson wrote:
So I thought I had SIP and NAT cracked a long time ago, but
something's just happened that's sort of upset the cart )-:
I have an * box behind a NAT firewall. Nothing unusual there, this is
something I've done many times - sip.conf has the correct
nat=
On Fri, 1 Jun 2007, Anthony Francis wrote:
do sip debug and then look again if still nothing then from linux do tcpdump
-Avvv host ip-address of problem device and see if its getting blocked by
iptables or not even reaching you. You should prolly show us what your
sip.conf looks like and the
On Jun 1, 2007, at 9:45 AM, Gordon Henderson wrote:
[snip]
Both these SIP - external PSTN provider connections register OK on
the * box, and outgoing calls placed over either connection works
perfectly. Outgoing callerId (set by the external provider) works
as expected. ) I have dialling
On Fri, 1 Jun 2007, Tom Rymes wrote:
On Jun 1, 2007, at 9:45 AM, Gordon Henderson wrote:
[snip]
Both these SIP - external PSTN provider connections register OK on the *
box, and outgoing calls placed over either connection works perfectly.
Outgoing callerId (set by the external provider)
If I have several local networks, can I specify that?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: Thursday, March 29, 2007 1:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
] On Behalf Of Eric
ManxPower Wieling
Sent: Thursday, March 29, 2007 1:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP NAT
Mike Hammett wrote:
I hate SIP. The only reason I'm doing this is that its cheaper than
deploying the server to a colo
Discussion
Subject: Re: [asterisk-users] SIP NAT
According to sip.conf.sample the answer is...well, I guess you can look
in /path/to/src/asterisk/configs/sip.conf.sample and see for yourself.
Mike Hammett wrote:
If I have several local networks, can I specify that?
-Original Message
What do you mean by 'non-standard' IP block?
Is the Asterisk machine behind a NAT, or are only your clients?
Did you look at the nat setting sin sip.conf?
Do you have a static public address that can be routed to the Asterisk
box?
From: [EMAIL
]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Lopez
Sent: Thursday, March 29, 2007 11:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] SIP NAT
What do you mean by 'non-standard' IP block?
Is the Asterisk machine behind a NAT, or are only your
Mike Hammett wrote:
I hate SIP. The only reason I'm doing this is that its cheaper than
deploying the server to a colo facility. My provider has given me a
non-standard IP block, so I can't do typical routing.
I have Asterisk server - MT w\NAT - PPPoE - MT - Provider.
I setup a
16 aug 2006 kl. 23.54 skrev Technical Support:
We have a client running Asterisk using a dynamic IP. When the IP
lease is renewed to a different address, their SIP connections to
external clients fail (one way audio).
A simple asterisk restart fixes the problem, but they can't count
on
marek cervenka wrote:
can you someone explain this bug? (or point me to number from
bugs.digium.com)
2006-03-28 19:07 + [r15699] Olle Johansson [EMAIL PROTECTED]
* channels/chan_sip.c: Fix breakage of NAT support for peers with
qualify=yes. Thanks Damin for access to your system,
As a general rule if the phone is behind NAT there
should be no issues. Server behind NAT = Lots of
issues (which can all be worked out). You will have to
specify NAT=YES in the dial plan.
Regards,
Dovid
--- Moises Silva [EMAIL PROTECTED] wrote:
you can redirect the ports of the router as
you can redirect the ports of the router as well. Or you can configure
your SIP phone to use a STUN server. Please read in voip-info.org
about SIP NAT, there are good suggestions.
regards
On 1/20/06, Michaël Gaudette [EMAIL PROTECTED] wrote:
Hello,
I'm a bit new to SIP, and I've set up a SIP
I have solved one part of the problem. I'm able to register. I'm able to call
SIP phones and I can hear them. The only problem is that they can't hear me.
So, this is the situation.
Softphone_1 (on public IP) = Internet = Router = * (private IP) =
Softphone_2 (private IP)
SP_1 can call and
mohammad wrote:
I have problem with incoming Sip call to users behind Nat.I set the following
for my users behind Nat:
nat=yes
canreinvite=no
qualify=yes
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
___
In my instance I'm using m0n0wall, but this is a hardware-neutral
question.
Sometimes, yes and no. The trick in Monowall I founds is to use the auto
add in Monowall to create the rules. If you manually create the rule, she
don' work. Why? Dunno. Using 1.11 built on Thu Nov 11 23:02:41 CET
Pizco,
SER is definitely better suited to deal with NAT issues then ASTERISK is. I
suggest looking at SER and NAT helpers like media proxy application (part of
SER). I also recommend looking at NAT devices at SER wiki page to make sure
that your router/nat device is compatible. In general, this
Hi, Alex.
Thanks for your answer, it is encouraging enough.
I'm reading through SER Admin's guide and I'm beginning to like it.
I've also installed serweb and looks very useful. It seems a lot of things
can be done with both asterisk and SER.
My router/nat device is a debian box with
On Tue, 2005-05-03 at 10:02 +0200, list wrote:
Hi,
have a setup which should not be unknown to others;
Asterisk behind wall doing NAT, and out in the wild world behind linksys
router a Polycom phone. The Polycom phone is on DMZ. It should register
with my server.
sip conf:
[4031]
Hei,
thanks that was it... Could bet I did try that earlier, but now its
working. Actually I did manage to join the 2 phones in conference but
not directly. Interesting.
thanks again
On Tue, 2005-05-03 at 17:34, Tor Setane wrote:
On Tue, 2005-05-03 at 10:02 +0200, list wrote:
Hi,
have a
.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kim Lux
Sent: Friday, January 28, 2005 1:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP + NAT = horrible mess
Comments below.
On Fri, 2005-01-28 at 08:18 +0800
I don't think you can use NAT = yes unless there is a STUN
server involved. See my post yesterday for my Grandstream settings.
No, I had nat=yes working with my Cisco 7960 which did not provide it's
public IP. However, you need to tell the IP Phone to start using the IP
and port that *
NAT=yes Rules
STUN=SUCKS
rtp streams =Rules
I have lots of devices connected behind NAT without trouble but in
fact with STUN was a real MESS
regards
Humberto
On Fri, 28 Jan 2005 10:40:25 -0500, Nabeel Jafferali
[EMAIL PROTECTED] wrote:
I don't think you can use NAT = yes unless there is
Discussion
Subject: Re: [Asterisk-Users] SIP + NAT = horrible mess
Comments below.
On Fri, 2005-01-28 at 08:18 +0800, Leo Ann Boon wrote:
Kim Lux wrote:
I was expecting to have to port forward too and yet our setup doesn't
require it, not on the laptop nor on the wireless
I've got Grandstreams (SIP devices) working behind double NATs, none the
less.
I recommend turning STUN off and make sure that your SIP devices are
generating random port numbers. If they generate static port numbers,
you'll get port collisions.
The other parameter to watch is the keep alive
On Thu, 2005-01-27 at 16:06, Kim Lux wrote:
I've got Grandstreams (SIP devices) working behind double NATs, none the
less.
I recommend turning STUN off and make sure that your SIP devices are
generating random port numbers. If they generate static port numbers,
you'll get port collisions.
On Thu, 2005-01-27 at 16:57 -0500, David Boyd wrote:
Will you Please share your configuration, I was ready to give up,
thinking no one had been successful.
I am not using Asterisk, so I can only give you the Grandstream part of
things. Maybe some of the Grandstream parameters will twig an
On Thu, 2005-01-27 at 17:25, Kim Lux wrote:
On Thu, 2005-01-27 at 16:57 -0500, David Boyd wrote:
Will you Please share your configuration, I was ready to give up,
thinking no one had been successful.
I am not using Asterisk, so I can only give you the Grandstream part of
things. Maybe
Here is another way to approach the problem: if you are using SIP phones
to connect to a SIP provider and are using * in between, try turning off
* and setting the * computer up as a simple NATing server and make sure
you don't have some sort of network issue that has nothing to do with
*.
Once
Jean-Michel Hiver wrote:
Hi Guys,
After days of fiddling, I can't really get my SIP device to work
communicate with Asterisk behind NAT. Sometimes the STUN server is
flaky, sometimes the device isn't reachable if the connection is
dropped and then put back on, sometimes it registers OK,
I was expecting to have to port forward too and yet our setup doesn't
require it, not on the laptop nor on the wireless router.
I think as long as the SIP clients open a port on the NATing device and
keep them open so the SIP provider can connect to it, all is well, even
if STUN isn't used.
I
Kim Lux wrote:
I was expecting to have to port forward too and yet our setup doesn't
require it, not on the laptop nor on the wireless router.
I think as long as the SIP clients open a port on the NATing device and
keep them open so the SIP provider can connect to it, all is well, even
if STUN
Comments below.
On Fri, 2005-01-28 at 08:18 +0800, Leo Ann Boon wrote:
Kim Lux wrote:
I was expecting to have to port forward too and yet our setup doesn't
require it, not on the laptop nor on the wireless router.
I think as long as the SIP clients open a port on the NATing device and
I have asterisk running on a public ip and a client running behind a natting
firewall with a Grandstream bt-100.Is there something special I should do to
get it working,I got other users working using host=dynamic in sip.conf
There are many answers to this in the list, it comes up all the time.
Dave,
Should it work to have a multi-homed asterisk server with grandstream
phones on the internal network and another grandstream phone on the
internet and be able to call between them? I set the bindaddr to the
external IP and pointed the internal and external grandstream phones to
At 4:01 PM +0100 10/20/03, WipeOut wrote:
Mark Evans wrote:
Hi All
Has anything been done to fix the issue where the * box is sat behind a
nat firewall?
Regards
Mark
As far as I know it can't be done.. The server has to be on a public IP..
You could try using a SIP aware router like the
Asterisk works perfectly fine in back of a NAT firewall, as long
as all of your SIP phones are also in back of that same fire
wall ;-)
Seriously, I'd fix this if I knew enough about SIP protocol.
Is anyone willing to write up what is required at the bit and byte
level? One thing that could
Actually it requires CHANGING the SIP protocol. Asterisk already
changes the SIP protocol when you use nat=yes and many clients also
change the SIP protocol to work with NAT.
On Mon, 2003-10-20 at 11:31, Chris Albertson wrote:
Asterisk works perfectly fine in back of a NAT firewall, as long
as
yes, regarding sip, but I have stil problems with rtp
- Original Message -
From: Mark Evans [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 20, 2003 4:49 PM
Subject: [Asterisk-Users] SIP Nat Issue
Hi All
Has anything been done to fix the issue where the * box is sat
--- Eric Wieling [EMAIL PROTECTED] wrote:
Actually it requires CHANGING the SIP protocol. Asterisk already
changes the SIP protocol when you use nat=yes and many clients also
change the SIP protocol to work with NAT.
Is it really a change to the format of what is sent or is it that
only some
--- Tomica Crnek [EMAIL PROTECTED] wrote:
to be more specific, I only managed to get xten softphone register to
*
behind the nat fw, but nothing else.
Where was the firewall?
1) Between xten X-Lite and the public Internet or,
2) Between Asterisk and the Publict Internet or
3) Both 1
You should start by reading the specific SIP and RTP RFCs. SIP is less
of an issue than RTP (as someone else pointed out)
On Mon, 2003-10-20 at 12:47, Chris Albertson wrote:
--- Eric Wieling [EMAIL PROTECTED] wrote:
Actually it requires CHANGING the SIP protocol. Asterisk already
changes
Eric Wieling wrote:
On Mon, 2003-10-20 at 11:31, Chris Albertson wrote:
Asterisk works perfectly fine in back of a NAT firewall, as long
as all of your SIP phones are also in back of that same fire
wall ;-)
Seriously, I'd fix this if I knew enough about SIP protocol.
Is anyone willing to write
PROTECTED]
Cc: Tomica Crnek [EMAIL PROTECTED]
Sent: Monday, October 20, 2003 8:37 PM
Subject: Re: [Asterisk-Users] SIP Nat Issue
--- Tomica Crnek [EMAIL PROTECTED] wrote:
to be more specific, I only managed to get xten softphone register to
*
behind the nat fw, but nothing else.
Where
Hi,
Is there anyway to use xlite though a nat
I have a xlite - nat- asterisk.
* is on a public IP.
When I do this, I get an error on the asterisk server because it is trying
to use the dirty ip of the computer running xlite.
All of the settings in xlite seem to have no effect!
I am new to * and I have been attempting to solve this same issue, but
have come to the conclusion that they only way to make it work is for *
to have a real reachable IP address or place another * box at the second
site and use IAX trunking. This second * box, unfortunately is
unsuitable for my
Don't know yet if it helps, but if you read the link at:
http://www.voip-info.org/tiki-index.php?page=NAT+and+VOIP
it will point you to:
http://www.sipcenter.com/files/SIPNATtraversal.pdf
However has the voip-info.org site; your stuff ROCKS!!
On Fri, Sep 19, 2003 at 03:11:31PM -0500, C.
ClientServer
XTEN -- */Firewall(NAT) ---IAX--- Firewall(NAT)/*
If you are going to use IAX, I don't think you have to put * on the
firewall boxes, only if you wish to use SIP.
Steve
___
Asterisk-Users mailing
, 2003 5:27 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] SIP + NAT Howto?
Client Server
XTEN -- */Firewall(NAT) ---IAX--- Firewall(NAT)/*
If you are going to use IAX, I don't think you have to put *
on the firewall boxes, only if you wish
: [Asterisk-Users] SIP + NAT Howto?
ClientServer
XTEN -- */Firewall(NAT) ---IAX--- Firewall(NAT)/*
If you are going to use IAX, I don't think you have to put *
on the firewall boxes, only if you wish to use SIP.
Steve
Hi George,
Do you have qualify=yes set in sip.conf for your phones?
When you check sip show peers, does it give you an OK (X ms) or does it
say UNREACHABLE or UNMONITORED?
If you enable qualify=yes or qualify=[some number] then Asterisk will
poll the SIP UA every once in a while to make sure
Forgot to mention that we have specified the nat=yes for all sip entries in
sip.conf.
Regards
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of George Lin
Sent: Wednesday, August 13, 2003 10:32 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP NAT
From: George Lin [mailto:[EMAIL PROTECTED]
I want to deploy multiple SIPs phone in our office. And we have
shutdown
the
firewall at our office router(with ip 211.x.x.x). we have deployed the
asterisk with IP 218.x.x.x.
All SIP phones have 192.x.x.x.
We have something similar George, *
Hi John,
thank you for pointing me to to some of the additional Asterisk documentation
stuff.
http://www.digium.com/handbook-draft.pdf
In addition, there are a variety of home-built pages.
http://www.automated.it/guidetoasterisk.htm
http://asterisk.gnuinter.net/
I found a
On Sunday 08 June 2003 08:45, Olaf Menzel wrote:
Hi John,
thank you for pointing me to to some of the additional Asterisk
documentation stuff.
http://www.digium.com/handbook-draft.pdf
In addition, there are a variety of home-built pages.
Hi all,
beacause I am a newbie in the asterisk ralm and the existing documentation
could not satisfy I'd like to ask you some Questions:
1. Does somewhere in the Internet exist additional documentations for asterisk
configuration ?
http://www.digium.com/handbook-draft.pdf
In addition,
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