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.