Reply To: Asterisk goes offline when connected to ZS

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#51368

lip
Member

Hi Guys, if I could join in.. I was about to start a topic asking for voip/sip help, as there are so surprisingly few to reference, then I see this one just started. And, with no less than atheling responding, who seems the resident expert from all previous posts.

atheling regarding your setup recommendations, I have:
Setup: zeroshell, DSL modem, cable modem, (no pstn lines), switch, asterisk box, 10 PCs, 10 SIP phones, and up 6 occasional remote SIP phones.

Script: [Pre boot]
modprobe nf_nat_sip
for file in /Database/custom/*
do
cp $(file) /root/kerbynet.cgi/scripts/
done

Virtual server:
ppp0/any UDP 5000-5082 192.168.1.2:5000-5082
ppp0/any UDP 10000-20000 192.168.1.2:10000-20000

Firewall: Forward table needs a rule corresponding the VS rule correct?
Accept UDP opt — in ppp0 out * 0.0.0.0->192.168.1.2 udp dpts:5000:5084
Accept UDP opt — in ppp0 out * 0.0.0.0->192.168.1.2 Layer7 RTP udp dpts:10000:20000

Asterisk:
NAT yes, RTP reinvite no, no other SIP settings to speak of really…

Issues:
Occasional call quality/drops, moreso on the end of people calling in.
Quality degrades and seems daily reboots help (usually just the asterisk box).

Questions:
First, I’ve had horrible call quality for the last few days, and restarted the asterisk box several times which changed nothing. So I rebooted zeroshell and it’s all clear again. Any ideas why this could be? Anything in zeroshell I can check, or setup to monitor? I’d at least like to determine if it is a hardware, software, or configuration related.

As the system is all voip (no pstn/pri), my primary concern is the voip provider sip trunk connection(5060,10000-20000). Does your setup cover this as well as the occasional remote phones?

So should I erase the RTP rules, and make SIP only 5060 (or is it 1 port per remote phone or something)? Could these extra open ports cause a problem or just less secure?

THANKS, IN ADVANCE, FOR ANY HELP, SERIOUSLY!

If I can’t figure this out, I’m going to get a DLink WBR-2310 router (because they are supposed to work perfectly for SIP) and separate the voice LAN, then consider trying zeroshell on different (atom?) hardware in the future. I can’t find the link but it’s currently running on a small Lite-ON ‘Book PC’ VIA C3 533MHz 500ram 40hdd, but maybe that’s not enough, or there is an incompatibility.

** As an aside, I believe the moderators should start a VOIP/SIP section in the forum index, as it’s a growing indispensable component.