[Asterisk-Users] extravagant behavior, nat problem ?

[Alessio Focardi]

Dear friends,

anyone has some experience of cisco PIX configuration with sip, I
suspect a nat problem, but i'm not sure ....

My situation:

Ser/Asterisk server on a real ip address, no firewall, open ip

Grandtream phone, real ip address

Xlite client, natted by pix with a internal/external direct conduit,
allow any rule for inside/outside connections

Tests made:

direct call

Grandstrem ---> ser ---> Xlite  = ok
Xlite ---> ser ---> Grandstream = ok

ok now to the problem, I know this is a long explanation, but I badly
need help, this looks to complex for me ! :)

Grandstream calls 10, ser has this rule

if (uri=~"sip:10+@") {

                rewritehostport("host.domain:5090");
                t_relay_to_udp("host.domain", "5090");
                break;
                };


call is forwarded to Asterisk, a menu is played, then an extension is
called

exten => *,1,Background(beep)
exten => *,2,Dial,SIP/Xliteextention at domain|30|mt
exten => *,3,Voicemail(u10)
exten => *,102,Voicemail(b10)

Xlite rings, but if I pick up calls are not connected, asterisk shows
ringing, Xlite shows Connected.

If I press Hangup on Xlite asterisk senses that and brings the
grandstream call to voicemail, I hear the voicemail prompt from
grandstream phone.

Reversing the process everithing works absolutely fine: if Xlite calls
asterisk, then a call is made to grandstream ext calls are connected.

To make some more tests I tried using a non natted xlite, it works,
that's the reason why i suspect that nat is the problem here.

Any idea ?


-- 
Best regards,
 Alessio                          mailto:alessiof at interconnessioni.it