If I can dial out on SIP, why can't anyone dial in?

Asterisk Asterisk Support
1

I’m running asterisk behind nat. I can make an outgoing sip call.
But if anyone calls in on fwd, they get an unavailable message.

sip.conf:

[general]
context=default
srvlookup=yes
language=en
disallow=all
allow = gsm ; what we deem is necessary
allow = ilbc
allow = speex
allow = g726
allow = ulaw
tos_sip = cs3
tos_audio = ef
tos_video = af41
register => fwdnumber:fwdpassword@fwd.pulver.com
context = fwd-incoming
nat=yes
insecure=port,invite

extensions.conf:

[fwd-incoming]
exten => s,1,Answer()
exten => s,2,Dial(Zap/1)

The register seems to work:

– parse_srv: SRV mapped to host fwd.pulver.com, port 5060

My asterisk server is behind a nat, but if I can make a sip connection one way, why not the other? I’ve also tried it not behind the nat. Same result.

puzzled.

sean

2

Are you forwarding the right sip ports, rtp ports through your natd server. My * server is behind a natd gateway like yours and it did not work until I started forwarding ports

5060-5082 for sip
8000-20001 for rtp

Mine is setup to forward udp and tcp ports for all these.

.

3

Does it work as soon as asterisk registers with your ITSP ? I had a router that would ‘forget’ where the server was and no calls came in.

4

Sorry i have to laugh at this ROFLMAO damn things almost have as bad ofa memory as me at times :wink:

How did you manage to fix the problem? I am interested to know.

Cheers,

David.

5

Simple. Switched routers.