[asterisk-users] Simple Asterisk to Asterisk SIP Call Setup?

[Russell Brown]

I have two Asterisk systems that can route to each other via a VPN with
firewalls disabled for testing purposes.

Each Server can see (tested via nmap) UDP port 5060 on the other.

So...  I thought that I could simply use a Dial command in Server A's
config to place a SIP call to Server B...  but it doesn't seem to work.

Server A (192.168.1.33) has:   

	exten => *136,1,Dial(SIP/90 at 10.10.111.13,30)

but whenever a user on Server A dials '*136' the call doesn't complete
and the CLI shows:

	Executing [*136 at from-sip:1] Dial("SIP/112-0071f650", "SIP/90 at 10.10.111.13|30") in new stack
	-- Called 90 at 10.10.111.13
	-- SIP/10.10.111.13-00793520 is circuit-busy
	== Everyone is busy/congested at this time (1:0/1/0)

I can't see anything in Server B's logs from 192.168.1.33

What am I missing?

Any pointers to help me get this working?

-- 
 Regards,
     Russell
 --------------------------------------------------------------------
| Russell Brown          | MAIL: russell at lls.com PHONE: 01780 471800 |
| Lady Lodge Systems     | WWW Work: http://www.lls.com              |
| Peterborough, England  | WWW Play: http://www.ruffle.me.uk         |
 --------------------------------------------------------------------