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475 posts

Ultimate Geek
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  Reply # 1078181 1-Jul-2014 19:43 Send private message

Just got it set to 1Gb a month.

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  Reply # 1078184 1-Jul-2014 19:49 Send private message

Try a Draytek router, they are pretty good at handling SIP traffic. I've never had to port forward to get Asterisk or other VoIP configurations working.




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475 posts

Ultimate Geek
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  Reply # 1078186 1-Jul-2014 19:50 Send private message

Ok, looking at tcpdump logs I see traffic from

SIP
107.150.45.66 New York
188.138.5.114 Berlin
202.180.76.182 Auckland ? WxC
27.111.14.65 Auckland
198.204.231.123 Sydney

TCP
74.125.31.125 Atlanta
216.52.233.71 Los Angeles




475 posts

Ultimate Geek
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  Reply # 1078193 1-Jul-2014 20:12 Send private message

Anyone tried the Sippot RBL client http://www.opentek.ca/sippot



475 posts

Ultimate Geek
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  Reply # 1078254 1-Jul-2014 21:24 Send private message

Oh well, I installed the sippot RBL client which added about 1400 addresses to my iptables
I then added the 5 other ip addresses that came from overseas to iptables as well ( none was in their honeypot list).
Shall run tcpdump again and see what else pops up.


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  Reply # 1078286 1-Jul-2014 21:46 Send private message

27.111.14.65 is 2Talk's proxy.



475 posts

Ultimate Geek
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  Reply # 1078296 1-Jul-2014 21:52 Send private message

And the other Auckland address belongs to callplus.

Yep, I have another pap2t or something device on premises which is turned off ... and I guess they're trying to find it.




475 posts

Ultimate Geek
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  Reply # 1078339 1-Jul-2014 22:51 Send private message

And 74.125.31.125 is xmpp traffic.

I must have googlevoice installed on my asterisk box.

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  Reply # 1078342 1-Jul-2014 22:57 Send private message

5060 is not the port that is used for audio, so fowarding it will not help with one way audio, there is another range that does the audio which can help, but even with it fowarded I was geting one way audio too often when going via asterisk.

Only reason to foward 5060 is to allow off site registration.

I gave up on asterisk and just have several analog phones on several ATA's to get all the numbers I need registered on. I take phone calls so seldom its not a big deal.

IMO asterisk is just not reliable enough in a natted configuration. It does dumb stuff that no ATA has ever done like totally lose all registrations when one dynamic dns name stopped resolving etc.




Richard rich.ms



475 posts

Ultimate Geek
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  Reply # 1078346 1-Jul-2014 23:16 Send private message

Well, I haven't had those sorts of issues.
I did have 629,000 calls in June .. though none made it outside except the local calls I personally made.
I've got hold of a router flashed with DD-WRT which allows me to set the source for NAT so once I'm up in the office I can try and install that.
In the meantime I'll see if these 1400 RBL addresses help.



475 posts

Ultimate Geek
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  Reply # 1078411 2-Jul-2014 08:17 Send private message

Only had 2 calls overnight, and already 22mb of traffic.

Have now disabled Truenet checking and also disabled the WiFi on their probe in case someone has been using my WiFi.

If I still have traffic over the day, will try another tcpdump.

29 posts

Geek
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  Reply # 1078432 2-Jul-2014 09:12 Send private message

Try an IP filter for WXC proxy server IP, rather than a port forward.

http://i.imgur.com/z5ZnW9e.png

This will allow all communication from WXC sip proxy to your asterisk and dump every other IP:5060 at the WAN port.

You'll still get charged for traffic sent to your connection but whoever is sending you traffic will move on once you start dropping packets.

If you're on a dynamic IP with your service provider I'd recommend rebooting after the setting is in to get a different IP.
If its a static check with ISP about changing the address.




475 posts

Ultimate Geek
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  Reply # 1078538 2-Jul-2014 11:53 Send private message

Thanks for looking at that for me.

I only have ssh access to my Asterisk box, and no GUI.  
And I'm not aware of any command line tool that I can use to change the Netcomm settings.
So, I'll have to wait until I can get to the office to try this out.


^__^
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  Reply # 1078545 2-Jul-2014 12:09 Send private message

gchiu: Thanks for looking at that for me.

I only have ssh access to my Asterisk box, and no GUI.  
And I'm not aware of any command line tool that I can use to change the Netcomm settings.
So, I'll have to wait until I can get to the office to try this out.



how about an ssh tunnel, should be able to access your routers webinterface from there.



475 posts

Ultimate Geek
+1 received by user: 2


  Reply # 1078781 2-Jul-2014 17:16 Send private message

Yes, I do have ssh running but not sure how  I can bring up a GUI to the router.

Anyway, in the meantime, I have altered my iptables by

Removing these lines:

ACCEPT     all  --  0.0.0.0/0            0.0.0.0/0
ACCEPT     udp  --  0.0.0.0/0            0.0.0.0/0           udp dpts:5000:5082 

And then:

/sbin/iptables -A INPUT -s 182.154.16.150,58.28.20.150,202.180.76.182,27.111.14.65,192.168.1.0/24,25.107.0.0/16 -p udp --dport 5000:5082 -j ACCEPT

So, I think SIP traffic is now just limited to WxC and 2talk externally, and allowed on Hamachi VPN, and local lan.

I'll see what happens now.

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