In case of IAX phones, this is not appliacble. IAX uses same port for both
control and voice.
I have not tried CISCO phones, but I beleive you do need port forwarding if
they are SIP phones. Otherwise, they will not accept calls. At least this is
the case with Polycom phones.
Rudolf
----- Original Message -----
From: "Tom Rymes" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>
Sent: Sunday, August 14, 2005 12:30 PM
Subject: RE: [Asterisk-Users] Why NAT problem
As a followup to my own post, AFAIK, my comments apply to SIP clients,
but you always have to forward the ports to the asterisk server...
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tom Rymes
Sent: Saturday, August 13, 2005 10:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Why NAT problem
This is not technically true. For instance, you can take a
Cisco 79X0 and put it behind NAT and it will work without
port forwarding. You do, however, have to program the phone
to enable the NAT features. (There are two, I can't remember
their names, though.) I have generally left the WAN IP
address blank, with no noticable ill effects, but that might
not be a good idea.
Also, I believe that you can do this with multiple phones, so
long as you use different port numbers for each phone (5061,
5062, etc)
Tom
> -----Original Message-----
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Rudolf Ladyzhenskii
> Sent: Saturday, August 13, 2005 9:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Why NAT problem
>
>
> At firewall/NAT you have to do port forwarding.
>
> If your phone is at port 5060, NAT device will receive a
> connection and has
> to know that it is destined for your SIP phone. So, forward
> port 5060 to the
> phone.
>
> Rudolf
>
>
> ----- Original Message -----
> From: "Kamran Ahmad" <[EMAIL PROTECTED]>
> To: <[email protected]>
> Sent: Sunday, August 14, 2005 6:52 AM
> Subject: [Asterisk-Users] Why NAT problem
>
>
> > hello
> >
> > i am using asterisk-1.0.9. i have a NAT problem.
> > without NAT registration is ok. and if user is bhind
> > NAT it is registring on asterisk. but SJPhone is
> > showing "not registered". i think asterisk is properly
> sending request
> > to UA. any comments............this sip.conf setting was working
> > previously
> >
> > -- Registered SIP '5000' at 0.0.0.0 port 5060
> > expires 120
> > -- Saved useragent "SJLabs-SJphone/1.40.258" for
> > peer 5000
> >
> > [general]
> > context=default
> > port=5060
> > bindaddr=0.0.0.0
> > srvlookup=yes
> > nat=yes
> > canreinvite=no
> >
> > [5000]
> > type=friend
> > port=5060
> > canreinvite=no
> > host=dynamic
> > nat=yes
> > insecure=yes
> > auth=plaintext
> >
> >
> >
> >
> >
> > ____________________________________________________
> > Start your day with Yahoo! - make it your home page
> > http://www.yahoo.com/r/hs
> >
> > _______________________________________________
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> > [email protected]
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