[Freeswitch-users] SIP Trunk with Private Static IP?

[Steven Ayre]

Gateways do not require usernames and passwords. You are required to
set the parameter, but if no authentication is needed they are ignored
so you can put anything in the field, so that is not a reason to avoid
them.

-Steve


On 25 February 2010 08:57, Steven Ayre <steveayre at gmail.com> wrote:
> Create two SIP profiles, each bound to one of your local IPs.
>
> You may create a gateway on the profile for the SIP trunk IP for the
> 10.42.0.1 server, but this is optional.
>
> You can then bridge calls via the SIP server using one of:
> <action application="bridge" data="sofia/gateway/gatewayname"/>
> <action application="bridge" data="sofia/profilename/number at 10.42.0.1"/>
>
> The advantages of using a gateway are:
> - supports authentication
> - will monitor the gateway to detect if it goes down (so calls fail
> instantly rather than after a timeout)
>
> As for the default gateway, it is the IP you send via to reach IPs
> that are not on a network you are connected directly to - you should
> probably only have one set, and it should be the one you go via to
> reach the Internet.
>
> -Steve
>
>
> On 25 February 2010 08:33, Roly Maz <rm at callrica.co.za> wrote:
>> Hi Community
>>
>>
>>
>>
>>
>> My Provider provides the following info when they supply a SIP trunk:
>>
>>
>>
>> ·         A direct connection into their network. i.e. they provide private
>> IPs:
>>
>> ·         An IP address I must use for my FS box e.g. IP: 10. 42.0.66 MASK:
>> 255.255.255.248 GW: 10. 42.0.68
>>
>> ·         An IP address for their SIP server 10.42.0.1
>>
>>
>>
>> I have setup a dual homed FS box (Windows Server 2008, latest FS version)
>>
>>
>>
>> NIC 1 – Internal LAN IP: 10.0.2.185 MASK: 255.255.255.0 GW: 10.0.2.253
>>
>> NIC 2 – SIP Provider (WAN) IP: 10. 42.0.66 MASK: 255.255.255.248 GW: 10.
>> 42.0.68
>>
>>
>>
>> Windows complains about multiple gateways – which I ignore? I can ping
>> internal addresses  and the SIP Server
>>
>>
>>
>> When I fire up FS, I can register Xlite phones on my LAN. I can dial and
>> hear the test IVR (5000)
>>
>>
>>
>> This means my Internal SIP Profile is ok.
>>
>>
>>
>> Now, how do i route a call out to the 10.42.01 SIP Server?
>>
>>
>>
>>  Creating a gateway doesn’t make sense, because I am not supplied a
>> username/password?
>>
>>
>>
>> Any pointers would be most appreciated, I am sure I am missing something
>> really simple.
>>
>>
>>
>> Roland
>>
>>
>>
>>
>>
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>>
>