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Posted on June 10, 2017 by Paul

New feature! Faultsmans Ringback on VoIP Phones

A "faultsmans ringback" facility is useful for testing that you're able to receive incoming calls. The idea is that you dial a special number, hang up, and the exchange calls you back. On our UAX13 strowger exchange, we use this relay set: and a similar relay set on our PABX4 based exchange: - but on the asterisk phones we didn't have a solution.

Until now! If you dial '9#' from any of our asterisk phones, you should hear a 3 tone "doo... dah... Dit!" sequence, which will repeat until you hang up. There should then be a short pause before your phone rings to indicate an incoming call. If you pick the handset up you should hear dial tone (although you won't be able to dial any numbers). The facility is still new, and we're still ironing out a few wrinkles, but try it and let Paul know how you get on.

For anyone interested in asterisk, it's based on the following dialplan contexts:

exten == 9,1,Goto(SIP-ringback,s,1) ; Faultsmans ringback
[SIP-ringback] ; This context (and SIP-ringback-complete) do faultsmans ringback
exten == s,1,Answer()
exten == s,n,Set(RINGBACK=${CALLERID(num)})
exten == s,n,Log(NOTICE, Ringback requested for ${RINGBACK})
exten == s,n,Wait(1) ; Wait 1s for the audio to connect
exten == s,n,Playtones(950/330,0/15,1400/330,0/15,1800/330,0/1500) ; "Dooh Dah Dit!"
exten == s,n,Wait(10) ; Play 10S of the above tone before hanging up
exten == s,n,Hangup()
exten == h,1,Log(NOTICE,Executing ringback for ${RINGBACK})
exten == h,n,Wait(3)
exten == h,n,Originate(SIP/${RINGBACK},exten,SIP-ringback-complete)
[SIP-ringback-complete] ; Used in conjunction with [SIP-ringback]
exten == s,1,Answer()
exten == s,n,Playtones(350+440) ; Dialtone
exten == s,n,Wait(10)
exten == s,n,Hangup()
There are a few knotty issues in the above, which mean that it's not quite as predictable as it might seem at first - but I've got plan for ironing those out

Posted on August 11, 2017 by Paul

For the non-techie audience, I've been tightening our security. There is a small chance I've tightened it a bit too far and stopped your VoIP phone at home from working. If that's the case, do let me know!

Posted on August 12, 2017 by Paul

It looks as though the asterisk is currently offline, as of 2017-08-12 14:25:01 (BST). I'm not sure yet the cause of this problem. It could be anything from a power failure, to the the router being offline, or the router having been reset losing the port forwarding configuration. More information when I've got it.

Update 2017-08-12 22:11 - I don't think it's the Dynamic DNS this time. The asterisk regularly phones home to my monitoring server (a process that isn't reliant on the Dynamic DNS) and it hasn't reported in since 14:25.

Update 2017-08-13 12:00 Sam very helpfully went in, and found everything switched off. He powered up the UPS and we're back in business, for now!

Posted on August 23, 2017 by Paul

What does SIP scanning look like?

Following on from my post two weeks ago about enhancing SIP Security I've been running a tool that's allowing me to profile the SIP brute force attacks we see, which are an unavoidable cost of our being able to have VoIP phones at home. Every time anything tries to talk to our Asterisk server over the internet (be that legitimate traffic from our ATA's at home, or scanning by "bad guys" trying to gain illegitimate access to our Asterisk) it creates a line in a log file. I've been through the results and have classified each connection attempt as legitimate, or unexpected. The graph is quite interesting:

About two thirds of the traffic we see is legitimate. The next largest chunk is 'friendly-scanner' which is a known SIP account brute force kit, based on sipvicious. The rest are mostly scans that are masquerading as legitimate devices. I've tweaked our blocking to cover most of the illegitimate traffic, but it's possible that I've widened the net a little too widely, so if your phone at home has stopped working, let me know!

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