Friday, November 30, 2007

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

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@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@from-sip:1] Dial("SIP/112-0071f650", "SIP/90@10.10.111.13|30") in new stack
-- Called 90@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
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| Russell Brown | MAIL: russell@lls.com PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com

|
| Peterborough, England | WWW Play: http://www.ruffle.me.uk

|
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