Re: [Asterisk-Users] sip+nat+asterisk

2005-08-02 Thread Eric Wieling aka ManxPower

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.
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[Asterisk-Users] sip+nat+asterisk

2005-08-02 Thread mohammad



 HI ALL;
 
 
 I have problem with incoming Sip call to 
users behind Nat.I set the following for my users behind Nat:
 
 nat=yes
 canreinvite=no
 
 
Appreciate any help
Mohammad
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Re: [Asterisk-Users] SIP, NAT & Asterisk

2003-06-08 Thread Tilghman Lesher
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.it/guidetoasterisk.htm
> > http://asterisk.gnuinter.net/
>
> I found a lot of helpful hints.
>
> > My suggestion is that if your Asterisk server is behind a NAT, the
> > only hosts it should talk with are devices that are also behind the
> > NAT, or other Asterisk servers via IAX or IAX2 which may be on the
> > outside of the NAT.
>
> All my SIP devices in my home network are behind the NAT. The
> asterisk server at our institute has a official public IP address.
> Maybe the existing firewall ca make some trouble ??

Two solutions:  your existing asterisk server could be given an
interface on the internal network (without forwarding packets).
Second, you could deploy an additional internal Asterisk server
whose sole purpose would be to translate SIP calls to IAX, to relay
to the outside Asterisk server.

-Tilghman

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Re: [Asterisk-Users] SIP, NAT & Asterisk

2003-06-08 Thread Olaf Menzel
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 found a lot of helpful hints.


> My suggestion is that if your Asterisk server is behind a NAT, the
> only hosts it should talk with are devices that are also behind the
> NAT, or other Asterisk servers via IAX or IAX2 which may be on the
> outside of the NAT.  

All my SIP devices in my home network are behind the NAT. The asterisk server 
at our institute has a official public IP address. Maybe the existing 
firewall ca make some trouble ??

Thank you for your fast reply and help


Olaf
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Re: [Asterisk-Users] SIP, NAT & Asterisk

2003-06-07 Thread John Todd
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, there are a variety of home-built pages.
http://www.automated.it/guidetoasterisk.htm
http://asterisk.gnuinter.net/

2. Does Asterisk work as a standard SIP Proxy ?
No.  Asterisk can perform basic SIP redirection, but it is not a 
standard SIP proxy in many of the ways that you might expect.  It can 
do _some_ of the features of a SIP proxy, but to call it a SIP proxy 
would be an overstatement.

3. I am just installing a Asterisk PBX in our institute and additionally I
purchased some ot the Snom 100  SIP devices. For SIP account I have
registered with iptel.org and fwd.pulver.com. Both work fine with the Snom
100 device. I tried the Snom 100 from home via a Linksys DSL router wich
supports UPnp and port range forwarding to the private network. I faced some
troubles with registering the Snom with IPTEL.ORG from my NAT'ed network.
Because I'd like to setup a Asterisk PBX to my private Network as well my
approach is to connect the SNOM to the asterisk at home network and establish
a server-server connection via IAX to the institutes Asterisk server. The
institute's Asterisk can login at iptel.org and fwd.pulver.com. If soembody
wants to call me he should first try to reach me at my Snom phone at
institutes premisses and after a defined time this call should be forwarded
to my home asterisk server to reach the SNom 100 at home.  BTW - for calling
out I want to use the AVM Fritz ISDN card with ISDN4LINUX. I could not find
any doc for Asterisk- ISDN4LINUX configuration.  Later I'd like to upgrade to
a Digium device to support conferencing. Wich of the Digium's devices you
propose for a normal EuroISDN line (2 B-channels) ? I have attached a small
figure for my SIP configuration aaproach.
My suggestion is that if your Asterisk server is behind a NAT, the 
only hosts it should talk with are devices that are also behind the 
NAT, or other Asterisk servers via IAX or IAX2 which may be on the 
outside of the NAT.  Anything else causes more headaches than it's 
worth.  I'm afraid I have no experience with the EuroISDN equipment, 
so I am not qualified to answer that part of your questions.

JT

Thank you for your help

regards

Olaf

--
Dipl. Ing. Olaf Menzel - System Engineer
FOKUS - Fraunhofer Institute for Open Communication Systems:
- Competence Center for Advanced Satellite Communication
Schloss Birlinghoven, 53754 Sankt Augustin, Germany
Phone: +49-2241-14-3494   Mobile: +49-175-2616161Fax: +49-2241-14-43494
email: [EMAIL PROTECTED]   Internet: http://www.fokus.fhg.de/satcom
Attachment converted: PrivateSpace:asterisk-conf.gif (GIFf/prvw) (000324C8)
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[Asterisk-Users] SIP, NAT & Asterisk

2003-06-07 Thread Olaf Menzel
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 ?

2. Does Asterisk work as a standard SIP Proxy ?

3. I am just installing a Asterisk PBX in our institute and additionally I 
purchased some ot the Snom 100  SIP devices. For SIP account I have 
registered with iptel.org and fwd.pulver.com. Both work fine with the Snom 
100 device. I tried the Snom 100 from home via a Linksys DSL router wich 
supports UPnp and port range forwarding to the private network. I faced some 
troubles with registering the Snom with IPTEL.ORG from my NAT'ed network. 
Because I'd like to setup a Asterisk PBX to my private Network as well my 
approach is to connect the SNOM to the asterisk at home network and establish 
a server-server connection via IAX to the institutes Asterisk server. The 
institute's Asterisk can login at iptel.org and fwd.pulver.com. If soembody 
wants to call me he should first try to reach me at my Snom phone at 
institutes premisses and after a defined time this call should be forwarded 
to my home asterisk server to reach the SNom 100 at home.  BTW - for calling 
out I want to use the AVM Fritz ISDN card with ISDN4LINUX. I could not find 
any doc for Asterisk- ISDN4LINUX configuration.  Later I'd like to upgrade to 
a Digium device to support conferencing. Wich of the Digium's devices you 
propose for a normal EuroISDN line (2 B-channels) ? I have attached a small 
figure for my SIP configuration aaproach.

 
Thank you for your help

regards

Olaf


-- 
Dipl. Ing. Olaf Menzel - System Engineer
FOKUS - Fraunhofer Institute for Open Communication Systems:
- Competence Center for Advanced Satellite Communication
Schloss Birlinghoven, 53754 Sankt Augustin, Germany
Phone: +49-2241-14-3494   Mobile: +49-175-2616161Fax: +49-2241-14-43494
email: [EMAIL PROTECTED]   Internet: http://www.fokus.fhg.de/satcom<>