[Asterisk-Users] SIP Nat Issue

2003-10-20 Thread Mark Evans
Hi All

Has anything been done to fix the issue where the * box is sat behind a
nat firewall?

Regards

Mark


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Re: [Asterisk-Users] SIP Nat Issue

2003-10-20 Thread John Todd
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 intertex range but I 
don't know how much milage you will get from that..

Later..


This has been exhaustively covered in the archives as recently as a 
few days ago.

See http://bugs.digium.com/bug_view_page.php?bug_id=104  and 
http://bugs.digium.com/bug_view_page.php?bug_id=359  for more 
details.

JT
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Re: [Asterisk-Users] SIP Nat Issue

2003-10-20 Thread Chris Albertson

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 help is try this and report
findings:

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.

I'd look into hacking Asterisk to mimic the X-Lite generated
SIP protocol.  

Put the result on the wiki.  Doing this will greatly speed up
the fix.



--- WipeOut [EMAIL PROTECTED] 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 intertex range but I 
 don't know how much milage you will get from that..
 
 Later..
 
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Re: [Asterisk-Users] SIP Nat Issue

2003-10-20 Thread Eric Wieling
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 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 help is try this and report
 findings:
 
 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.
 
 I'd look into hacking Asterisk to mimic the X-Lite generated
 SIP protocol.  
 
 Put the result on the wiki.  Doing this will greatly speed up
 the fix.
 
 
 
 --- WipeOut [EMAIL PROTECTED] 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 intertex range but I 
  don't know how much milage you will get from that..
  
  Later..
  
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   Cell:   310-990-7550
   Office: 310-336-5189  [EMAIL PROTECTED]
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Re: [Asterisk-Users] SIP Nat Issue

2003-10-20 Thread Tomica Crnek
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 behind a
 nat firewall?
 
 Regards
 
 Mark
 
 
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Re: [Asterisk-Users] SIP Nat Issue

2003-10-20 Thread Chris Albertson

--- 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 different IP addreses get written into some of the fields?

Can you be _specific_ about what is done?




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Re: [Asterisk-Users] SIP Nat Issue

2003-10-20 Thread Chris Albertson

--- 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 and 2.

I think #1 canbe made to work but I've not herd of anyone getting
#2 to work yet.

What did you have to do to get it to work?

=
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  Home:   310-376-1029  [EMAIL PROTECTED]
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  Office: 310-336-5189  [EMAIL PROTECTED]
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Re: [Asterisk-Users] SIP Nat Issue

2003-10-20 Thread Eric Wieling
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 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 different IP addreses get written into some of the fields?
 
 Can you be _specific_ about what is done?
 
 
 
 
 =
 Chris Albertson
   Home:   310-376-1029  [EMAIL PROTECTED]
   Cell:   310-990-7550
   Office: 310-336-5189  [EMAIL PROTECTED]
   KG6OMK
 
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Re: [Asterisk-Users] SIP Nat Issue

2003-10-20 Thread Olle E. Johansson
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 what is required at the bit and byte
level?   One thing that could help is try this and report
findings:
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.
I'd look into hacking Asterisk to mimic the X-Lite generated
SIP protocol.  

Put the result on the wiki.  Doing this will greatly speed up
the fix.


--- WipeOut [EMAIL PROTECTED] 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 intertex range but I 
don't know how much milage you will get from that..

Later..

 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.

Well, it requires changing messages exchanged in the SIP protocol.
The outbound proxy may want to change data in the SDP body when
doing an invite, so the media path for the incoming media stream
has to be changed.
/O

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Re: [Asterisk-Users] SIP Nat Issue

2003-10-20 Thread Tomica Crnek
opened static nat for both * and client, one on each firewall
configured xten to connect to external (nat) address of asterisk
configured sip on asterisk on external (nat) address of client
I think this was all

- Original Message - 
From: Chris Albertson [EMAIL PROTECTED]
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 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 and 2.
 
 I think #1 canbe made to work but I've not herd of anyone getting
 #2 to work yet.
 
 What did you have to do to get it to work?
 
 =
 Chris Albertson
   Home:   310-376-1029  [EMAIL PROTECTED]
   Cell:   310-990-7550
   Office: 310-336-5189  [EMAIL PROTECTED]
   KG6OMK
 
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 Do you Yahoo!?
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