[Asterisk-Users] Re: Why NAT problem
Kamran Ahmad
p_kami at yahoo.com
Tue Aug 16 02:29:48 MST 2005
hello
but this conf was working for me when i installed
asterisk last time. and UA was successfully reg and
working.
i think port forwarding is not the solution. because
it was working without port forwarding in my last
installation.
it is a simple case UA behind NAT and Asterisk is on
public ip
[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
>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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page
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>>> > >
>>> > >
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>>
>>
>>
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