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
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 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:
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
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
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
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
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
__
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
_
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
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
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
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
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
>
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 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
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
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
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
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
> 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
> 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
...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
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 -- 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
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
>
>
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