[Asterisk-Users] Grandstream SIP phones across NAT

[asterisk at emptyhole.net]

I have a Grandstream Budge Tone 100 SIP phone connected through a NAT
firewall to an Asterisk server. I successfully connected the phone via
NAT to the server but when I dial the extension to an AGI script, it
does not kill the process as soon as I hang up. As a result, the next
time I pickup, it gives me multiple streams of audio. It turns out that
when I hang up, it does not kill the last AGI process. The question is
why and how do I resolve this problem. This problem does not occur if I
do a direct connect without NAT.