Hello,

My target system is :
PSTN <---> Sip Provider <---IP/ADSL---> Router with fw/NAT <--- SIP/IP/Eth --> Asterisk <--- SIP/IP/Eth --> SIP Phones

Asterisk is configured to keep NAT connection with SIP provider open (with qualifyfreq) so I don’t have any problem (yet) with either casual incoming or outgoing calls.

To work around a possible No Audio when an incoming call is forwarded to an external number (because of NAT issues), I would like to configure Asterisk so that :

whenever a call comes in from SIP trunk, Asterisk starts to play a Ringing tone (while endpoint is ringing) as an RTP flux so that router opens appropriate NAT translation.

My questions are:

1. Is the method above recommended to work around fw/NAT issues ?

2. Are the setings bellow sufficient to implement the above method (from experience, I’ve gathered mixed results and I would appreciate any input that would confirm I’m on the right or the wrong track) or shall add more magic somewhere (Answer(), Progress(), …) ?

sip.conf:
progressinband=yes prematuremedia=no

extensions.conf exten => _X.,1,Dial(SIP/foo)

Regards