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

[Gilles]

On Thu, 23 Dec 2010 15:54:59 +0100, Jeroen Eeuwes
<jeroeneeuwes at gmail.com> wrote:
>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.

Thanks Jeroen.

After more reading, I found what it was: To cut down on the hacking
attempts I saw (since posting in this mailing list...), I decided to
reconfigure my NAT router to use another port than UDP5060 for SIP
while leaving it as-is on the inside so internal SIP clients would
still connect to the usual port.

But www.smartvox.co.uk/astfaq_configbehindnat.htm says:

"When configuring your NAT/firewall/router device, you will probably
need to find the settings for "port forwarding" or "one-to-one" NAT.
Make sure your NAT device does not use port address translation. i.e.
if your Asterisk server expects to receive SIP messages on port 5060,
make sure you also use port 5060 on the WAN port of your NAT device to
forward these messages. Similarly, make sure the same range of port
numbers are forwarded on the WAN port for RTP as will receive the RTP
on the Asterisk server."

Reconfiguring the NAT router back to UDP5060 solved the problem: Calls
originating from the remote SIP client are not longer cut off at 20s
by Asterisk.

Thanks everyone for the help.