asterisk users - Jan 2014 - How to tell Asterisk to to send Ringing signals as into RTP

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
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On Wed, Jan 15, 2014 at 03:11:46PM +0100, Olivier wrote:
> 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(), ...) ?
Dial(SIP/foo,45,r(ring))

Asterisk will start sending audio packets to the caller before the
callee answers.

> sip.conf:
> progressinband=yes
progressinband=never



-- 
Stefan Tichy  ( asterisk3 at pi4tel dot de )