actually typically you don't want to limit your inbound calls.

For example, I get a Twilio PSTN number 1555123456, and configure
twilio to send the calls to sip.example.com:5080.  So, FreeSWITCH will
receive the calls in public context.

Now, what I can do is give my SIP URI
(1555123456@sip.example.com:5080) to third parties -- for example,
register it at some ENUM directories. Or maybe my overseas partner
configures their PBX to send calls to me directly via Internet,
instead of sending it to their local PSTN.

So, what you need to make sure is that you only accept calls to your
own numbers in public context. If you receive an unauthenticated call
for some arbitrary destination number, it should be rejected (you
don't really want to send it to PSTN, do you?).

If there's only a small quantity of PSTN numbers landing on your
FreeSWITCH, you can simply configure a sequence of condition
statements in the public context. If your setup is bigger, there are
many ways to look up the number in some database and make a decision
where to route it. One of the simplest ways is to have your directory
users, one per DID, and use "user_exists" call to check if this is our
number.

I don't see much of a reason in sending inbound calls to the internal
profile (port 5060) and making an exclusion ACL to disable the
authentication. It only makes things complex and it doesn't add
security.







On Wed, Oct 7, 2015 at 5:15 PM, Gonzalo Gasca Meza
<gascagonzalo@gmail.com> wrote:

> I have a SIP Provider which has 4 IP addresses in US. They send SIP calls
> from any of those 4 IPs to my Freeswitch. **They do not require
> authentication, nor SIP Trunk registration, just purely send a SIP INVITE**
>
> Incoming calls work fine except that I want to assign a context when I
> receive an incoming call from this ITSP at gateway level.
> Gateway xml file is configured under external folder.
> (../conf/sip_profiles/external/)
>
> I can see in packet capture and in freeswitch.log call comes from correct ip
> and port, but is always routed to context default. Hence I need to configure
> something there. Is it possible to define context at gateway level?
>
> Console trace:
> http://pastebin.com/NzzLAK8U
> Freeswitch trace
> http://pastebin.com/YUYVLfyY
>
> I defined my 4 SIP Gateways (status up) as follows: (1 for each IP address)
>
> <gateway name="itsp-inbound-us1">
>
>   <param name="auth-calls" value="false"/>
>
>   <param name="proxy" value="54.172.60.0"/>
>
>   <param name="register" value="false"/>
>
>   <param name="context" value="itsp"/>
>
>   <param name="username" value="not-required"/>
>
>   <param name="password" value="not-required"/>
>
>   <param name="from-user" value="not-required"/>
>
>   <param name="expire-seconds" value="600"/>
>
>   <param name="extension" value="1000"/>
>
> </gateway>
>
>
>>sofia profile external gwlist
>
> itsp-inbound-us4 itsp-inbound-us3 itsp-inbound-us2 itsp-inbound-us1
>
>> sofia status gateway itsp-inbound-us1
>
> Name    itsp-inbound-us1
>
> Profile external
>
> Scheme  Digest
>
> Realm   54.172.60.0
>
> Username not-required
>
> Password yes
>
> From    <sip:not-required@54.172.60.0>
>
> Contact
> <sip:gw+itsp-inbound-us1@52.2.15.172:5060;transport=udp;gw=twilio-inbound-us1>
>
> Exten   1000
>
> To      sip:not-required@54.172.60.0
>
> Proxy   sip:54.172.60.0
>
> Context itsp
>
> Expires 600
>
> Freq    600
>
> Ping    0
>
> PingFreq 0
>
> PingTime 0.00
>
> PingState 0/0/0
>
> State   NOREG
>
> Status  UP
>
> Uptime  536s
>
> CallsIN 0
>
> CallsOUT 0
>
> FailedCallsIN 0
>
> FailedCallsOUT 0
>
> =================================================================================================
>
> Any suggestion?
>
>
>

> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting@freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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