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) wo
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 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 dial command in use as well a
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
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=
localnet=
Commercial 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?
>
>
] 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 t
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: [aste
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
PROTECTED]
[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
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 PROTE
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 s
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 we
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
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
___
Ast
>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
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 iptables,
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 is
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
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:
> [40
t; [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,
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
> 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 * receiv
ED]
> [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, L
.
-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
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
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 i
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
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, sometim
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
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.
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 id
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 collis
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"
> 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 t
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
>
>
To: <[EMAIL 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 softph
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 up
Chris Albertson wrote:
X-Lite _can_ dail out to FWD through a firewall but Asterisk can't.
SO this gives us the perfect chance to compare the content of
outbound packets where we have a working and non-working example.
X-lite when configured for FWD behind a NAT uses an outbound proxy
for that con
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
> > cha
--- 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) Bot
to be more specific, I only managed to get xten softphone register to *
behind the nat fw, but nothing else.
- Original Message -
From: "Tomica Crnek" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, October 20, 2003 7:44 PM
Subject: Re: [Asterisk-Users]
--- 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
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
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
> a
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 he
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 inter
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 intertex range but I
don't know how much mil
> 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
> > Sent: Friday, September 19, 2003 5:27 PM
> > To: [EMAIL PROTECTED]
> > Subject: RE: [Asterisk-Users] SIP + NAT Howto?
> >
> >
> > > ClientServer
> > >
> > > XTEN <--> */Firewall(NAT) <---IAX---> Firewall
iday, September 19, 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
> 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 mail
; */Firewall(NAT) <---IAX---> Firewall(NAT)/*
More reading to do now :)
Thanks fellas
-cj
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Stephen Varga
> Sent: Friday, September 19, 2003 3:37 PM
> To: [EMAIL PROTECTED]
&
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. Johnso
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 sc
> 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 Geor
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 questio
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 i
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.
> > http://www.automated.i
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 foun
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, the
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