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

[Jeroen Eeuwes]

Hi Gilles,

> 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 have (almost) that and it is working fine. One user needed to use a
different port than 5060 because his modem really loves to interfere
with packets on 5060. Probably because their ISP can also provide a
SIP/phone line and the ISP modem is assuming that all packets on 5060
are for that phone line even though it is not enabled.

I don't use STUN anywhere.

Anyway, in all cases it has worked as soon as the phone was able to
register on my server (that was usually the hard part!).

In sip.conf I have added for all the remote users the setting
canreinvite=no. The downside to that setting is that Asterisk is
always in the audio path. For my situation that does not really
matter.

Best regards,
Jeroen Eeuwes