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Asterisk Mystery Incoming

I've just put 'fail2ban' on my asterisk box and it has cleared a constant stream of registration attempts, but this keeps coming up:
== Using SIP RTP CoS mark 5
[2013-12-05 17:05:54] NOTICE[29712]: chan_sip.c:22622 handle_request_invite: Call from '' (54.224.145.76:5076) to extension '+448458673552' rejected because extension not found in context 'CallsComingIn'.

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I'm trying to work out what's happening.
The context 'CallsComingIn' is in the [general] section of sip.conf. There is no registration attempt, so is it a sip invite coming straight into port 5060 trying to dial an extension '+448458673552' and asterisk uses the 'CallsComingIn' context but can't match the extension?
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Silas2
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Silas2
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1 Solution
 
PhonebuffCommented:
So is this a valid call not being handled, or is port 5060 wide open and someone is probing to see where they can call through you to ?  

I don't recognize your Context from the standard FreePBX builds,   What exactly are you supporting, a roll your own or one of the many ISO packages.

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Silas2Author Commented:
its rollyourown
No its not a valid call, it is 5060 wide open but it has to be open so the asterisk box can function doesn't i?
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Silas2Author Commented:
I'm asking the question cos i think it exposing a glaring weakness in my knowledge (i'm only supporting this asterisk box for a tiddly number of users and the sip trunk provider has banned premium calls so there's nothing really at stake),
But what i don't understand is how/why are 5060 port call-initiations possible from a SIP extension which is not registered?
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PhonebuffCommented:
Well,

    If your external SIP is only your provider I would set rules in your firewall to only permit traffic to/for him on 5060.   You should always use White lists when possible.

    You can also do a catch all rule to Hangup for any attempt to make an incoming call to an undefined DID.

    Post your context within a code block an I will try and find time to look at it for you.

    ++++++++++++++
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Silas2Author Commented:
Well  the SIP registrations are all over the place with dynamic IP's (they work from home) so a white list isn't really easy, unless you have a clever idea...

So it is true that a non-registered SIP extension can enter into a phone call with Asterisk thru 5060? (I thought the point of registering was for the uname/pwd security check)
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PhonebuffCommented:
Well,

    I would use a PBXinaFlash build and Wards Travelingman3 for your project, as Ward as already built the magic into TravelingMan to handle roaming & Dynamic IP users.

    =======

  Yes, if you allow Guests and a number of other optional settings your system could route SIP traffic from 'Non Registered" sources.
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Silas2Author Commented:
Ah, allowguest, that would seam to be what i'm missing...
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