i have a problem with SIP and NAT. I have an Asterisk server behind a NAT and i would talk with a SIP client ouside server’s NAT.
When i try to call a SIP client from another client (SIP or IAX anyway inside the server’s NAT), the softphone rings but we can’t talk each other in both directions.
I exactly followed what asteriskguru tutorial explains about it. Here it is my configuration in sip.conf:
externip= [my pubblic ip]
localnet=[subnet ip where the server is in]/[subnet mask]
Then, in rtp.conf i set
From the CLI i can see (while ringing):
SIP/601-081fd258 is ringing
Call on SIP/601-081fd258 left from hold
SIP/601-081fd258 answered SIP/602-081f92f0
Packet2Packet bridging SIP/602-081f92f0 and SIP/601-081fd258
And after hangup i can see:
== Spawn extension (macro-stdexten,s,1) exited non-zero on ‘SIP/602-081f92f0’ in macro ‘stdexten’
== Spawn extension (macro-stdexten,s,1) exited non-zero on ‘SIP/602-081f92f0’