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Sip Scanner... can't get rid of it!

Started by threehappypenguins, March 10, 2013, 03:16:44 PM

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threehappypenguins

Hi, I read this topic here: http://www.obitalk.com/forum/index.php?topic=4067.0

And I tried to implement some of the changes. I thought that I did it right, and it "seemed" to have worked, but I just received another call from a 4 digit number again: 1000.

Here is what my call history says:



Call 1
18:37:04
18:37:04
18:38:08
03/10/2013    18:37:04
From '1000' SP1(1000)

Fork to:
SP3(username@sip2sip.info);(Failed: No Service) PH2
Ringing
Call Ended




Call 2
18:36:54
18:36:54
18:37:03
18:37:07
03/10/2013    18:36:54   
From '1000' SP1(1000)

Fork to:
SP3(username@sip2sip.info);(Failed: No Service) PH2
Ringing
Call Connected
Call Ended

I had forgotten that during the last bout, I had disabled sp3 (and all the others) while I was making changes to the X_InboundCallRoute. I just forgot to enable sp3 again (which is why it said "No Service").

Originally (with the help of ianobi), I had in X_InboundCallRoute:

{sp3(username@sip2sip.info),ph,ph2}

Because I had no idea what I was doing, and it was about 1 o'clock in the morning (WHY?!?!), I copied and pasted what the forum post said (see my above link):

{(?|x|xx|xxx|xxxx|xxxxx|xxxxxx|un@@.|anon@@.):},{ph,ph2}

I added the ph2 on the end because the next day I found only one phone was ringing, and I was like, "Oh yeah, I have two phone ports!"

So that didn't stop the scanner, apparently... because it happened again today!

How do I stop this thing? I also want to make sure 911 is still functioning, and I want to be able to route my calls to my android again as well (hence, the sip2sip). Sp3 is what I am using so I can making outgoing calls via Android on Wi-Fi (which, isn't working right now because I can't figure out how to get a dynamic dns on my new router, nor can I figure out how to do a static ip on it either).

ianobi

#1
You may want to have a look at items 1 and 2 in this post:

http://www.obitalk.com/forum/index.php?topic=5455.msg35330#msg35330

If you have this:
Voice Services > SP1 Service > X_InboundCallRoute:
{(?|x|xx|xxx|xxxx|xxxxx|xxxxxx|un@@.|anon@@.):},{ph,ph2}

It really should have blocked a call from CallerID "1000" coming in on sp1, so that's hard to explain.

For all users who want to test their IncomingCallRoute "Scanners trap". Here is a simple way to do it:

Download PhonerLite softphone onto a pc in the same subnet. (It's a free download).
Set up a very basic account on the PhonerLite softphone. Leave proxy/registrar blank, leave register checkbox unchecked. Fill in "User Name" - this is CallerID.
Send calls to your OBi by calling Ip address:port. Say 192.168.1.10:5061 (default for sp2)

Keep sending the same calls, but change the "User Name" to any CallerID that you wish to test as an incoming call to your OBi. This only takes about two seconds per test call - change "User Name", then hit "save" each time.

For calls that fail to get into your OBi, the Phonerlite Logbook will record "486:Busy Here"

threehappypenguins

For some reason the  X_InboundCallRoute change back in SP1. Maybe it is AcroVoice's autoprovisioning? I changed it to what I wanted a couple of days ago, and it is still the same... so I guess that's good? Not sure why it changed because autoprovisioning usually changes it within a few minutes. So... I don't know...

BUT... how do I implement this: {sp3(username@sip2sip.info),ph,ph2}

Into this: {(?|x|xx|xxx|xxxx|xxxxx|xxxxxx|un@@.|anon@@.):},{ph,ph2}

(mash them together so I can do both)

Shale

How did you change X_InboundCallRoute?

noahliam

I am happy to find this post very useful for me, as it contains lot of information.

ianobi

No mashing required - simply use both rules:

Voice Services > SP1 Service > X_InboundCallRoute:
{(?|x|xx|xxx|xxxx|xxxxx|xxxxxx|un@@.|anon@@.):},{sp3(username@sip2sip.info),ph,ph2}

Any CallerID that matches the first rule will be blocked. All other CallerIDs will use the second rule, which will call the sip2sip account using sp3, phone 1 and phone 2. The first of those three to answer will be connected to the incoming call.