[Asterisk-Users] Outgoing SIP trunk

[Juan J. Sierralta P.]

On Mon, 2003-09-15 at 00:10, John Brown wrote:
> Hi Juan,
> 
> basicly you would have the * system "call" the SIP channel
> 
> exten => _91800XXXXXXX,1,Dial(SIP/${EXTEN:1}@fwd) 
> 
> in your extensions.conf  file would cause AST to dial via SIP
> the number I dialed by using  fwd
> 
> in my sip.conf
> 
> [fwd]
> type=friend
> secret=<MY PASSWORD>
> username=<MY USERNAME>
> host=fwd.pulver.com
> context=fwd-in
> 
> hope this helps

	Thanks it worked. Now another questions does * support being behind
NAT, I saw in sip.conf that it cant support clients which are behind NAT
but I don´t know if * can be behind a NAT.
	The only thing that need to be behind NAT is to register it´s WAN IP
instead of its own IP an change the <Contact> header accordingly. Even
most SIP server replies REGISTER with a "received=" header so the client
can get its NAT IP automagically.

 
-- 
Juan J. Sierralta P. <juanjo at atmlab.utfsm.cl>
UTFSM