Hello, I’m new to Asterisk and the server configuration. I’ve got TrixBox distro running and a linksys pap2 with a single trunk registered. However the problem I’m having is that I keep getting random calls from extensions that I had not created and when you pick the phone up there’s no one there. I have read that this is happening because of jerks running SIP scanners or something like that. Also I have ran TCP dump to analyze the packets of these calls and I’m able to intercept there sip invites. What I’ve noticed is that there sip invites are large digget number unlike a normal US phone number would look like. So what im wondering is if there is some way to set a filter on what kind of calls can come in and that there criteria would match what a normal incoming phone number would be formatted like?
This is what my TCP dumps look like and then on the phone’s caller id they just show up as some random extension like 101, or 5001 or something. Also when I check the call log at the voip service theres noting there.
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22:57:01.123621 IP (tos 0x20, ttl 110, id 22475, offset 0, flags [none], proto: UDP (17), length: 789) 192.168.1.1.sdl-ets > 192.168.1.100.sip: [udp sum ok] SIP, length: 761
INVITE sip:[email protected] SIP/2.0
The INVITE sip:[email protected] is the jerk. Is there a way to filter these out so that only a normal phone number containing 10 diggets is allowed? Something like 1234567890@myserveraddress would only be sent to the ATA. Any help would be appreciated thanks.
Also I’ve done the change the port on the ATA thing and that didn’t work.