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

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@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@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
--------------------------------------------------------------------
| 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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