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
On 11 January 2012 15:43, Kevin P. Fleming kpflem...@digium.com 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
From: Steve Davies davies...@gmail.com
Sent: Wednesday, January 11, 2012 12:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SIP and NAT best practices since recent
@lists.digium.com
*Subject*: Re: [asterisk-users] SIP and NAT best practices since recent
changes?
On 11 January 2012 15:43, Kevin P. Fleming kpflem...@digium.com 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
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
Hendersongordon+aster...@drogon.net wrote:
On Mon, 3 Aug 2009,
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
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
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
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
_
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
_
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 there,
I
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
Still i cannot resolve this issue, please anyone can help me with this?
Thanks in advance
--
_
Facundo Agustin Barrera
--
www.openlabs.com.ar
Let the penguins do the work
-
Buenos
Dear list:
I have the typical one way audio problem, as far as i know
it's a nating problem, my hosts inside my lan can call to outside
internet hosts, but can't listen a thing, i read a lot about sip and
rtp and protocols and the problem it seems to be with NAT, this is the
config i
Change To canreinvite=no
On 1/6/07, Facundo Barrera - GMail [EMAIL PROTECTED] wrote:
Dear list:
I have the typical one way audio problem, as far as i know
it's a nating problem, my hosts inside my lan can call to outside
internet hosts, but can't listen a thing, i read a lot about
NAT changes address of the packet, but does not go inside of the SIP
packet itself. And SIP packet contains address as well. If you look at
debug output, you will see that SIP packets have remote host local
address in them, not the public IP as one would expect. At least this
is the problem I
Isn't that what
externhost=sip.server.com.ar my server name on the internet
localnet=192.168.5.0/255.255.0.0 my LAN
is supposed to do?
Bob...
Rudolf Ladyzhenskii wrote:
NAT changes address of the packet, but does not go inside of the SIP
packet itself. And SIP packet contains address as
Thanks for the answers , tried canreinvite=no , but still cannot
listen any soung from the outside, any other idea??
Thanks in advance
--
_
Facundo Agustin Barrera
--
www.openlabs.com.ar
Let the penguins do the work
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
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
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
Hello All;
I have a Grandstream 496 ATA and it is behind a NAT Router. The phone
service works well, but it is setup to support Call-Waiting, which it
does not do. When I am on the phone and someone calls, instead of
getting a ring, they go straight to Voicemail with the busy message. I
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
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,
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
method that
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
traverse
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
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
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
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
Hi everyone My problem is the following: I'm trying to make a call from a sip phone (SJphone) behind a Restricted Cone NAT towards and Asterisk behind another NAT (a freeBSD 3.3 using pf). By now I'm only trying to play a record set in the remote Asterisk. My soft phone registers without
--- Blake Krone [EMAIL PROTECTED] wrote:
What is the best solution? I dont want to have
modify firewall's at all or
do port fowarding. Ideally I would like a solution
that with either a
softphone or wireless hardphone one could connect
via friends, family, or
hotspots without reconfiguring
What is the best solution? I dont want to have modify firewall's at all or do port fowarding. Ideally I would like a solution that with either a softphone or wireless hardphone one could connect via friends, family, or hotspots without reconfiguring their devices.
What are people using? STUN?
Mensaje citado por: Blake Krone [EMAIL PROTECTED]:
What is the best solution? I dont want to have modify firewall\'s at all or
do port fowarding. Ideally I would like a solution that with either a
softphone or wireless hardphone one could connect via friends, family, or
hotspots without
Hi All,
I have a question for you:
- "SIP doesn't work behind NAT very
well"
Do you agree with this sentence?
regards
César
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Hi All,
I have a question for you:
- SIP doesn't work behind NAT very well
Do you agree with this sentence?
regards
César
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To
César Davi Ávila do Nascimento wrote:
Hi All,
I have a question for you:
- SIP doesn't work behind NAT very well
Do you agree with this sentence?
Depends on the NAT/firewall in question, you can also alleviate some of
these issues using STUN and sip proxing...
--
Best regards,
Duane
I have a question for you:
- SIP doesn't work behind NAT very well
Do you agree with this sentence?
Depends. Asterisk behind a nat box tends to be an implementation
problem limited by the knowledge of the person doing the implementation
and somewhat by the functionality implemented within
Thanks a lot!
Regards
César
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, January 31, 2005 9:18 AM
Subject: Re: [Asterisk-Users] SIP x NAT
I have a question for you
I have a question for you:
- SIP doesn't work behind NAT very well
Do you agree with this sentence?
Complete and utter crap (if you assume a few things).
SIP w/NAT works just fine if:
Asterisk itself is not behind NAT
You do not want to use SIP reinvites
You use some form of NAT Keepalive*
: [Asterisk-Users] SIP x NAT
Hi All,
I have a question for you:
- SIP doesn't work behind NAT very well
Do you agree with this sentence?
regards
César
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2005 09:50:31 -0600
Subject: RE: [Asterisk-Users] SIP x NAT
I'll agree with that sentence. There are many times when even STUN and
so on
isn't going to help. In Guatemala, a lot of people end up with private
IPs,
behind two NATs, etc. I've seen them aggressively timeout connections,
limit
Eric Wieling wrote:
I have a question for you:
- SIP doesn't work behind NAT very well
Do you agree with this sentence?
Complete and utter crap (if you assume a few things).
SIP w/NAT works just fine if:
. . . .
Hardly complete and utter crap when it has to be followed by a laundry
list of
, 31 Jan 2005 09:50:31 -0600
Subject: RE: [Asterisk-Users] SIP x NAT
I'll agree with that sentence. There are many times when even STUN and
so on
isn't going to help. In Guatemala, a lot of people end up with private
IPs,
behind two NATs, etc. I've seen them aggressively timeout connections
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 told
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
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
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??
Assuming that getting H323 to work over NAT is almost really hard
What is
about having both SIP clients venid different NATs ¿ is it posible or as
hard as H.323?
Thanks!
Marc.
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SIP works fine behind NAT if you have externip, localnet localmask
defined in sip.conf. I believe it was committed since 0.7.1.
-Heison
On Mon, Feb 23, 2004 at 08:51:23PM +0100, Marc Fargas wrote:
Assuming that getting H323 to work over NAT is almost really hard? What is
about having both SIP
- Original Message -
From: Heison Chak [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, February 23, 2004 7:50 PM
Subject: Re: [Asterisk-Users] SIP over NAT
SIP works fine behind NAT if you have externip, localnet localmask
defined in sip.conf. I believe it was committed since 0.7.1
Hello Users,
I am attempting to create a sip connection in the following network:
Sipgate.de -- Internet -- Gate -- Asterisk PBX -- Some Extension
Gate, the gateway and nat'ing firewall has sip udp (5060) traffic and
rtm udp (8000 to 8020) traffic forwarded to the asterisk pbx machine.
Both
Hello List,
Just thought I would post an update, I got asterisk to register with
sipgate.de.
I was wrong, it was my firewall (maybe).
Here is the way a normal nat under openbsd pf works:
udp 192.168.1.100:5060 - 24.102.192.227:(random port) - 217.10.79.9:5060
but add this line to pf.conf
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Patrick Lidstone (Personal E-mail)
Sent: 26 January 2004 18:31
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP behind NAT - use of externip option
I am having difficulty configuring SIP behind NAT
I am having problems in a couple of installations where I have SIP
phones (both GS101 and ATA186) connecting to an asterisk box that has a
public IP address, where the stations are behind NAT.
I'm still testing to make sure I have all the permutations looked at,
but from what I can tell, what
I am having problems in a couple of installations where I have SIP
phones (both GS101 and ATA186) connecting to an asterisk box that has a
public IP address, where the stations are behind NAT.
I'm still testing to make sure I have all the permutations looked at,
but from what I can tell,
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
...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.
Well, I happen to be one of those very specific cases... ;) and looks
like
will have experiment with it myself. Although I'd hate to re-invent
the
wheel.
Checking e-mail this morning it looks like we have two independent
fixes that both do what has been suggested in this thread.
No
Peter Zeltins wrote:
Checking e-mail this morning it looks like we have two independent
fixes that both do what has been suggested in this thread.
No need for a third except posibly a merge of the two.
Would you care to elaborate? I don't see anything in asterisk-users, and no
mention of
http://lists.digium.com/pipermail/asterisk-users/2003-October/024968.html
Any idea when these hacks will appear in CVS?
We should all hope never. That's why you call it a hack
because it works for only one very specific case and would break
SIP under Astrisk for most people. It even
--- Peter Zeltins [EMAIL PROTECTED] wrote:
Well, I happen to be one of those very specific cases... ;) and looks
like
will have experiment with it myself. Although I'd hate to re-invent
the
wheel.
Peter
Checking e-mail this morning it looks like we have two independent
fixes that
That's for pointing out Walter Snel hack.
Adding his two additional features would not be
hard.
http://lists.digium.com/pipermail/asterisk-users/2003-October/024968.html
Any idea when these hacks will appear in CVS?
Peter
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Peter Zeltins wrote:
That's for pointing out Walter Snel hack.
Adding his two additional features would not be
hard.
http://lists.digium.com/pipermail/asterisk-users/2003-October/024968.html
Any idea when these hacks will appear in CVS?
Probably when they have been properly tested and
Hello,
my next problem is with SIP device behind NAT.
First few seconds of the call are OK. Astrisk is sending the packets to
the public IP address of the FW/NAT (62.152.224.3). But this change in
10 second and packets are send to the my public addres.(192.168.1.163).
in the sip.conf for the
Hi!
First few seconds of the call are OK. Astrisk is sending the packets to
the public IP address of the FW/NAT (62.152.224.3). But this change in
10 second and packets are send to the my public addres.(192.168.1.163).
in the sip.conf for the phone(X-Lite) is
I think you should check
--- Peter Zeltins [EMAIL PROTECTED] wrote:
That's for pointing out Walter Snel hack.
Adding his two additional features would not be
hard.
http://lists.digium.com/pipermail/asterisk-users/2003-October/024968.html
Any idea when these hacks will appear in CVS?
We should all hope never.
PROTECTED]
Date: Wed, 29 Oct 2003 09:13:31 -0800 (PST)
Subject: Re: [Asterisk-Users] SIP behind NAT, workaround to make W Snel's
very welcome fix work both for inside *and* outside clients
--- Peter Zeltins [EMAIL PROTECTED] wrote:
That's for pointing out Walter Snel hack.
Adding his two additional
Hello everyone and welcome to my first post to the list!
After studying for a couple of weeks, I finally built * for the first
time last night, and of course had the same SIP-behind-NAT woes that
plague all of us who use NATted connections.
It was therefore with no small joy that I read the fix
Honestly I can't see all these NAT woes people speak of... I have * on a
public ip .. sip.conf entries with nat=yes load em up.. and they work. So
I have yet to see why everyone has SO MANY problems.
bkw
On Tue, 28 Oct 2003, Christopher Stephens wrote:
Hello everyone and welcome to my first
That's for pointing out Walter Snel hack.
Adding his two additional features would not be
hard.http://lists.digium.com/pipermail/asterisk-users/2003-October/024968.html
in sip.conf nat=1 means the _client_ that Asterisk is
talking with is NAT'd. We could add a line like below
to sip.conf
Can anybody explain me what does canreinvite=yes really does?
Not sure how technical an answer you want becasue it look slike you know whats going on but as I unterstand it canreinvite=no tells the UA that reinvite is not supported and so causes all the RTP traffic to be routed via the * server..
WipeOut . wrote:
Any ideas on the client A to C (same LAN, same NAT box, unique
outside IP, same * server)?
Only thing that springs to mind is to install another * box
internally and then use IAX to connect the internal * box to the
external one.. then the internal phone will call each other
I have been trying to get SIP UA work with NAT but i have no been
successful has any one got NATed ATA working(i.e an ATA witha private IP
working with NAT).
Asterisk registers the 192.168.0.3 Ip but no call go through at all,
infact there is no log of any call made on asterisk console.
can
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
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 have
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
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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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
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