[Freeswitch-users] Freeswitch inbound logic match

[Michael Nielsen]

My reason for this kind of issue is that I have a case where users are
charged for incoming calls...

So I want to make sure, that ONLY registrered users and my SIP gateway can
make calls with our FreeSWITCH.

On Mon, Oct 12, 2015 at 1:18 PM, Michael Nielsen <mic.niel84 at gmail.com>
wrote:

> I see that scenario make sense.
> But what if I only want to have registered users call each other?
>
> I've tried with external_auth_calls=true in vars.xml, but that still allow
> completely external parties to call me...
>
>
>
> On Thu, Oct 8, 2015 at 9:44 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
>> 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 at 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 at 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 at 54.172.60.0>
>> >
>> > Contact
>> > <sip:gw+itsp-inbound-us1 at 52.2.15.172:5060
>> ;transport=udp;gw=twilio-inbound-us1>
>> >
>> > Exten   1000
>> >
>> > To      sip:not-required at 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 at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
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>>
>> _________________________________________________________________________
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>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
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>> http://www.cluecon.com
>>
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>
>
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