Alessio Focardi
2004-Feb-17 05:10 UTC
[Asterisk-Users] extravagant behavior, nat problem ?
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@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@interconnessioni.it
Does it work if you don't re-write the port to 5090, but rather leave it at
5060?..
What is the 'ser'?
Please post the relevant portions of your PIX config too.
-E
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Alessio Focardi
Sent: Tuesday, February 17, 2004 7:10 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] extravagant behavior, nat problem ?
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@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@interconnessioni.it
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users