[asterisk-users] Asterisk hangs up call after 20s

[Gilles]

On Wed, 22 Dec 2010 13:22:47 -0500, Bruce B <bruceb444 at gmail.com>
wrote:
>This is a NAT issue like noted before.
>
>Try:
>localnet=192.168.0.0/ <http://192.168.0.0/24>255.255.255.0
>instead of:
>localnet=192.168.0.0/24
>
><http://192.168.0.0/24>Also, make sure you have all your VPN connections as
>localnet and other side subnet as localnet as well if you are using VPN.
>Otherwise, open the neccessary ports needed for SIP and RTP. If you note
>your router type someone might be able to help more specifically.

Thanks Bruce for the tip, but Asterisk still hangs up after 20s when
the call originates from the remote user on the Net.

The router is built by my ISP, so it has no brand/model. I believe
it's based on OpenBSD. I have no VPN: The remote SIP user connects out
using STUN.

While going through the debug messages, I can see that at some point
in the call, Asterisk tries to send SIP messages to the remote user...
using Asterisk's public IP address instead of the remote user's IP
address :-/

But then, I'm not clear at how to set things up so that remote users
can register with Asterisk and be part of the dialplan just like they
were on the LAN, with both Asterisk/local and remote users behind
their respective NAT firewall, so I would have been very lucky to get
this working without more investigation :-)

If someone has a working configuration where...
1) Asterisk and some users are on a private LAN behind a NAT firewall
2) some roadwarriors, behind their own NAT firewall, are allowed to
register with Asterisk, and make/receive calls just like they were in
the office
3) the NAT firewall protecting the Asterisk server has SIP and
RTP/RTCP ports mapped, while the NAT firewall protecting the remote
user has its ports open dynamically using STUN

... I'm interested in how you set things up.

Thank you.