[OpenSIPS-Users] Integrating OpenSIPs against Asterisk

[James Lamanna]

Hi Sergio,
I think I also want OpenSIPS to handle UA registrations as well, so I can remove
that burden from the Asterisk boxes. Then OpenSIPS would rewrite the
URI to send it to Asterisk. (Though maybe this isn't possible or makes things
infinitely more complicated?)
In addition, almost 100% of my current UAs are behind NAT (my asterisk box
is not behind NAT), so all the SIP extensions in asterisk have nat=yes set.
I assume if I want OpenSIPS to handle registrations, then it also
needs to handle
NAT as well before forwarding it to Asterisk.

I know that having OpenSIPS handle UA registrations makes things a bit more
complicated, but I think its important for scalability to remove that
burden from
Asterisk. I'm guessing that I would need to turn qualify off on each
SIP extension
in Asterisk as well, so Asterisk still tries to complete the call
because it doesn't
monitor it as well, though I suppose the OPTIONS messages sent out by Asterisk
could be rewritten by OpenSIPs to qualify would still work as well?

Thanks.

-- James

On Sat, Mar 7, 2009 at 10:30 AM, Sergio Gutierrez <saguti at gmail.com> wrote:
> Hi James.
>
> Your case sounds like you would use your OpenSIPS just as a proxy. Asterisk
> would be your UAS.
>
> If that is your situation, you could start from the example config file
> which is installed with source code;
>
> From that file you can take validations for SIP signaling; you would rewrite
> uri of SIP requests to route to your asterisk, or in case you have several
> asterisks, you could try loadbalancing if you need.
>
> Also, you could choose whether your OpenSIPS would operate as stateless or
> stateful proxy according to your particular needs.
>
> Finally, if you decide to perform accounting at proxy, some actions should
> be included in your config file (The comments in the example file would
> help).
>
> Feel free to ask any other thing you need.
>
> Best regards.
>
> Sergio.
>
> On Sat, Mar 7, 2009 at 1:18 PM, James Lamanna <jlamanna at gmail.com> wrote:
>>
>> Hi,
>> Does anyone have some good examples of an OpenSIPs
>> configuration that integrates with Asterisk?
>> Essentially I want to use OpenSIPs as the UA, but still run all
>> the calls through Asterisk (for dialplan, etc..)
>>
>> I've tried searching for some good examples, but I haven't found any
>> for Asterisk yet.
>>
>> Thanks.
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> --
> Sergio Gutiérrez
>