Greetings -

I have a peculiar problem which I was wondering if anyone had some 
insight into.

I have an asterisk server with 2 NICs (1 extern IP, one intern IP). 
Asterisk V1.4.19

I have a set of SIP phones on the internal LAN which houses the intern 
IP of the pbx.

I also have 3 SIP phones that reside outside of the local network, (i.e. 
people’s homes) behind there own firewalls.

The pbx forwards local calls (3 digit) between the extensions, and 
relays (10 digit) external calls up to an ITSP.

All phones are registered and are reachable (i.e qualify=yes) I can call 
any extension from any other extension. All channels are configured to 
use ulaw, so there should be not transcoding issues.

 From an outside (PSTN) phone, I can call the pbx and route to any 
extension.

 From a LAN phone I can complete outbound calls to the PSTN.

 From any one of the 3 outside phones, I cannot complete PSTN calls, The 
phones ring, they pick up, they disconnect. SIP works, RTP doesn’t.

/Using tcpdump, I can see the RTP stream from the phone to the pbx, from 
the pbx to the ITSP, from the ITSP back to the pbx, BUT NOT back to the 
phone. I see the RTP stream from the ITSP coming back into the pbx, so 
it’s not like it’s trying to establish a direct media path to the phone. /

/Using tcpdump on one of the LAN phones, I can see all 4 RTP streams 
working fine. /

I’ve fiddled with the NAT settings on both the phones and in the 
extension definitions, and only succeeded in making it worse.

Clearly this is NAT related, but is complicated by that that local calls 
between extensions work fine.


Any clues on how to investigate this further. ?

Oh, btw, this worked fine on a prior asterisk PBX running 1.2. Not 
thinking this is a Asterisk1.4 problem, but it has worked. It shouldn’t 
be that hard to get it working again.

Thanks for any thoughts...

rb


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