Re: Simple Asterisk to Asterisk SIP Call Setup?

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Simple Asterisk to Asterisk SIP Call Setup?
\ (Russell Brown) (30 Nov 2007)
. \ Vivek Shrivastava (30 Nov 2007)

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Subject:	Re: Simple Asterisk to Asterisk SIP Call Setup?
Group:	Asterisk-users
From:	Vivek Shrivastava
Date:	30 Nov 2007

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> 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/90,30)
>
> 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|30") in new stack
> -- Called 90
> -- 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 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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