I currently run an Asterisk server on a NetBSD system. It mostly works
but sometimes I have weird issues. As far as I can tell they are
usually NAT issues.
I have a Cisco SPA-2102 with two phone numbers installed. I have NAT
Mapping and NAT Keepalive enabled. No STUN server. Both are using
5060. This is behind an ADSL through a WRT54GL with no special port
handling.
The server is 11.15.1. My sip.conf includes this:
[general]
context=unauthenticated
allowguest=yes
udpbindaddr=0.0.0.0
nat=force_rport,comedia
srvlookup=yes
qualify=yes
All of this works fine. It also works fine with the few clients that I
have connected. However, certain changes cause failures, usually one
way audio suggestion NAT issues.
First experience - I add a softphone on my laptop and assign a third
number. I can register but one way audio. It also messes up the
working lines.
Second - My local carrier provides SmartRG ADSL modem/routers. Right
now I have it set to bridge mode and do everything in thw WRT. If I
switch to using the router in the SmartRG I have problems with the
existing two lines again.
I really need this to work with whatever hardware the client has. They
may have different ATAs, soft phones or SIP phones. Are my server
settings reasonable? Do I need to make specific requirements for the
client settings? Using a STUN server didn't seem to help. Is it a
good idea to specify it anyway?
Any help appreciated.
Cheers.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net
VoIP: sip:darcy at Vex.Net
