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

[Vivek Shrivastava]

looks like something wrong with the dial plan in the extensions.conf.. i
would recommend start debug on and see the content of "full" log  may be
that give some clue.

Thanks,

Vivek


On 11/30/07, Russell Brown <russell at lls.lls.com> wrote:
>
>
> 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         |
> --------------------------------------------------------------------
>
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