[OpenSIPS-Users] Proxy everything to asterisk

[Olle E. Johansson]

17 apr 2013 kl. 13:04 skrev Bogdan-Andrei Iancu <bogdan at opensips.org>:

> Hello Arthur,
> 
> The OpenSIPS script allows you to implement whatever logic you want, so the answer is : yes, you can do that. 
> 
> Reuse the part for handling the sequential requests from the default opensips script and for initial requests (after handling CANCEL and retransmissions) you can simply do :
>     $du = "sip:asterisk_ip:asterisk_port";
>     t_relay;
> 
> This will send the INVITE to asterisk without changing the RURI at all.

Just to add a little bit more:

Depending upon the version of Asterisk, you might want to select transport as well, like

>     $du = "sip:asterisk_ip:asterisk_port;transport=udp";

/O
> 
> Regards,
>  Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> 
> On 04/17/2013 12:51 PM, Arthur Titeica wrote:
>> 
>> Hello,
>> 
>> I'm rather new to opensisps.
>> 
>> >From what I read this is not possible but I thought I should ask just to make 
>> sure.
>> 
>> Is there a configuration setup so that opensips doesn't handle 
>> users/extensions but just forwards everything that matches the configured 
>> domains to an asterisk gateway?
>> 
>> Something like any_ext at sip.example.com should be allowed to go to asterisk and 
>> let asterisk to deal with auth.
>> 
>> Thank you all for any input.
>> 
>> 
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