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]
for file in /Database/custom/*
cp $(file) /root/kerbynet.cgi/scripts/
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
NAT yes, RTP reinvite no, no other SIP settings to speak of really…
Occasional call quality/drops, moreso on the end of people calling in.
Quality degrades and seems daily reboots help (usually just the asterisk box).
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.