[asterisk-users] Looking for best practices

[D'Arcy J.M. Cain]

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