[Asterisk-Users] Why NAT problem

[Rudolf Ladyzhenskii]

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" <trymes at rymesheating.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
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: asterisk-users-bounces at lists.digium.com
>> [mailto:asterisk-users-bounces at lists.digium.com] 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: asterisk-users-bounces at lists.digium.com
>> > [mailto:asterisk-users-bounces at lists.digium.com] 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" <p_kami at yahoo.com>
>> > To: <asterisk-users at lists.digium.com>
>> > 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
>> > >
>> > >
>> > >
>> > >
>> > >
>> > > ____________________________________________________
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