I'm having some problems getting an ATA186 behind NAT working. When I had
it on the same subnet as the Asterisk server, it worked fine. Now Ive
taken the ATA on the road with me, and it's behind a Dlink router+firewall,
doing NAT. I pick it up, hear a dialtone .. the firewall on the asterisk
hi there,
I 'm a newbie in "VOIP technologies" ; i 'm implementing asterisk and i 'm
wonder what is the best way to resolving "the Asterisk/NAT problem" : some
clients are behind a NAT.
anyone could help me?
thanks
johanna
_
App
hi there,
I 'm a newbie in "VOIP technologies" ; i 'm implementing asterisk and i 'm
wonder what is the best way to resolving "the Asterisk/NAT problem" : some
clients are behind a NAT.
anyone could help me?
thanks
johanna
_
App
I recently did a set up where I replaced a simple D-link home router
that was having trouble processing a T1's worth of bandwidth with a
linux machine running iptables. the kernel was 2.6.29-r5 and I chose
the SIP connection tracking modules from the menuconfig.
Router worked fine for norma
Hello all. I'm having a little problem here with NAT, and I already read
a lot of documentation on web, but I still cant understand how to get
asterisk and "external (on internet)" sip clients connected.
Could anybody give me a tip ?
Thanks
Lincoln
_
Hi All,
i found an article that explains SIP NAT woes.
http://www.sipcenter.com/files/SIPNATtraversal.pdf
It is a great read for all people in the mailing list that have problems
with SIP when * is behind NAT or when there is NAT between asterisk and a
SIP phone.
Serge
Oh, and yes, the * is current as of a few days ago .. so it should have
that new SIP code mark was working on a while back.
Thanks
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
John,
Client Behind a NAT should not be problem. What are your issues? If you post
your scenario and more details about your problem only then some can help
you better.
Jai
"Buy SIP DID at www.didforsale.com"
On Wed, Oct 22, 2008 at 12:24 AM, Johanna NIRINA <[EMAIL PROTECTED]>wrote:
>
> hi ther
I'm using asterisk 1.4 . There is some sip clients is behind a NAT : the
asterisk server can't send request to these client. I'm looking for a solution
to solve that in the server (asterisk) side. (sorry for my english).
thanks,
johanna
_
Johanna NIRINA wrote:
I'm using asterisk 1.4 . There is some sip clients is behind a NAT : the
asterisk server can't send request to these client. I'm looking for a solution
to solve that in the server (asterisk) side. (sorry for my english).
thanks,
johanna
__
On Mon, 2009-08-03 at 13:29 -0400, Ketema Harris wrote:
> I recently did a set up where I replaced a simple D-link home router
> that was having trouble processing a T1's worth of bandwidth with a
> linux machine running iptables. the kernel was 2.6.29-r5 and I chose
> the SIP connection tra
On Mon, 3 Aug 2009, Ketema Harris wrote:
> my questions are: What is the correct way(or resource to find a way)
> to get a linux firewall to work with SIP so that the NAT issue is not
> an issue ?
Remove all SIP ALG/connection tracking modules and use old fashioned port
forwarding on the router
Hello!
What are the nat_sip modules you mention?
When I set up a linux router some time ago and configured sip.conf
with net=yes, everything went smoothly just like any other router.
Elliot
On Mon, Aug 3, 2009 at 8:45 PM, Gordon
Henderson wrote:
> On Mon, 3 Aug 2009, Ketema Harris wrote:
>
>> m
Could you please explain what the network configuration you want to try? it would be really helpful.
you can be as simple as: SIPphone--> internet --> NAT--> asterisk
or whatever your particular scenario is.Alyed
Return-Path: <[EMAIL PROTECTED]> Mon Jul 31 11:43:16 2006Rece
Lincoln Zuljewic Silva a écrit :
Hello all. I'm having a little problem here with NAT, and I already
read a lot of documentation on web, but I still cant understand how to
get asterisk and "external (on internet)" sip clients connected.
So you have an Asterisk that is behind NAT, and you want
have you tried nat=1 in your friend declaration? I notice in your dump it
says "non-NAT"
Mark
On Fri, 21 Mar 2003, denon wrote:
> Oh, and yes, the * is current as of a few days ago .. so it should have
> that new SIP code mark was working on a while back.
>
> Thanks
>
>
Thanks -- I didn't realize that needed to be set. It works now, but
there's a horrible echo on the sip client side. (I dont know about the
other side, as I havent called any humans yet :)
I don't, however, hear an echo when I call voicemail or such .. so I'm
assuming it's something with the br
On Fri, 21 Mar 2003, Mark Spencer wrote:
> have you tried nat=1 in your friend declaration? I notice in your dump it
> says "non-NAT"
>
I´m in the same situation, trying to debug an ATA 186 behing a NAT.
And i´m stuck with "SIP/2.0 407 Proxy Authentication Required" debug
messages. Does anyone
Thanks Moises. I was kind of hoping that, at least if I hosted my Asterisk
server somewhere where there was no NAT for the * box that the SIP phones
wouldn't create any issues.
How do you people with Hosted PBX handle the deployment of SIP phones behind
NAT firewalls? Is it just elbow grease an
In response to the SIP and NAT discussion, I have updated the ticket
on the subject that seemed to be getting the most attention: #104.
There are enough clueful people here that perhaps someone can come up
with a patch that handles NAT in the elegant way that I describe in
the bugnotes, as I am
Hi all,
Seriously, I've tried to read everything I could find (& search for) on
voip-info.org and other sites about this problem, but have been unsuccesful.
Equipment:
xten lite
X100P
Whitebox linux running Asterisk / AMP
D-Link DI-804HV (VPN router)
I have installed another DI-804HV at a second
Most often the simple addition of nat=yes in the relevant sip.conf
stanza is all that's required to make a remote SIP phone work from
behind a firewall.
for example
[2201]
user=blah
secret=blah
auth=blah
allow=blah
host=dynamic
nat=yes
I've been running 4 remote SIP phones across the internet
How about when you have four or five SIP devices at a single location? Do
you manually assign each phone a separate port and add firewall/router
rules? I am looking for an inexpensive device or method that will allow
this happen automatically. Rather than going that route, my current
solution is
Trevor G. Hammonds wrote:
How about when you have four or five SIP devices at a single location? Do
you manually assign each phone a separate port and add firewall/router
rules? I am looking for an inexpensive device or method that will allow
this happen automatically. Rather than going that
I thing, that configuring nat device/firewall at consumer site isn't
always possible, thus simplest (but not optimal) way is to configure
phone in sip.conf as nat=yes & canreinvite=no, this should work in most
cases even if multiple phones are behind same nat, like adsl router.
disadvatage is, t
Leo Ann Boon wrote on Saturday, 21 January 2006 6:21 PM:
> Trevor G. Hammonds wrote:
>
>> How about when you have four or five SIP devices at a single
>> location? Do you manually assign each phone a separate port and add
>> firewall/router rules? I am looking for an inexpensive device or
>> met
Trevor G. Hammonds wrote:
While I have not used siproxd, I have read a bit about it. From my
understanding of the docs, the local SIP agents register to siproxd, but
siproxd does not register to Asterisk. So the calls will traverse the NAT
properly, but features like MWI will not work in this
Leo Ann Boon <> wrote on Sunday, 22 January 2006 4:32 PM:
> Trevor G. Hammonds wrote:
>
>> While I have not used siproxd, I have read a bit about it. From my
>> understanding of the docs, the local SIP agents register to siproxd,
>> but siproxd does not register to Asterisk. So the calls will
>
Apart of what everyone writes with the NAT=YES I would suggest using
canreinvite=no as well as normally asterisk cans the reinvite and this
might cause the audio not to get through the NAT and cause dead air for
the users specially if the users are behind 2 seperate NAT servers eg.
different p
...and to solve another problem, there's my suggestion on support for outbound SIP
proxy.
http://bugs.digium.com/bug_view_page.php?bug_id=359
There are corporate networks that use a "SIP proxy proxy" as an ALG, application layer
gateway,
for all outbound and inbound SIP traffic in the DMZ. Th
> Seriously, I've tried to read everything I could find (& search for) on
> voip-info.org and other sites about this problem, but have been unsuccesful.
>
> Equipment:
> xten lite
> X100P
> Whitebox linux running Asterisk / AMP
> D-Link DI-804HV (VPN router)
>
> I have installed another DI-804H
> each vendor for rtp. Cisco uses one range, xlite another, asterisk
> another, etc, etc. Mapping the sip port (udp 5060) is easy; mapping
> the rtp ports and using the proper nat statements (possibly at both
> the phone location and asterisk location) tends to be difficult. Then
X-Lite can be tol
I was testing 1.0.2 with one phone behind a nat.
I have it also setup in the sip.conf for nat=yes, but after the phone
has registered with asterisk and you look at 'sip show peers' is shows
the sip phone Nat=no
Has anyone experienced this problem??
Hi,
Since the recent update to the NAT configuration options and defaults
in chan_sip.so, I am interested in any SIP/NAT best practices advice.
What I've always done in the past is:
Global: nat=no
SIP handsets that are local: nat=no
SIP handsets that are remote: nat=yes
ITSP SIP trunks: nat=yes
On 01/11/2012 05:29 AM, Steve Davies wrote:
Hi,
Since the recent update to the NAT configuration options and defaults
in chan_sip.so, I am interested in any SIP/NAT best practices advice.
What I've always done in the past is:
Global: nat=no
SIP handsets that are local: nat=no
SIP handsets that
On 11 January 2012 15:43, Kevin P. Fleming wrote:
> On 01/11/2012 05:29 AM, Steve Davies wrote:
>>
>> Hi,
>>
>> Since the recent update to the NAT configuration options and defaults
>> in chan_sip.so, I am interested in any SIP/NAT best practices advice.
>>
>> What I've always done in the past is:
From: "Steve Davies"
Sent: Wednesday, January 11, 2012 12:51 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-users] SIP and NAT best practices since recent
changes?
On 11 January
On 01/11/2012 12:09 PM, Bryant Zimmerman wrote:
*From*: "Steve Davies"
*Sent*: Wednesday, January 11, 2012 12:51 PM
*To*: "Asterisk Users Mailing List - Non-Commercial Discussion"
*Subject*: Re: [ast
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