HI Michael,
I would say you can get each sip messages (REGISTER,INVITE,ACK,BYE) routed to
your asterisk server using rewritehostport("<your_ip>:5060");
you can configure proxy as your opensips in sip clients with port 5260.
if(is_method("REGISTER")){
rewritehostport("<your_ip>:5060"); # This will forward any
register packet to your asterisk.
route(1); # <-- this route actually send the sip packet to
asterisk
}
route[1] {
if (!t_relay()) {
xlog("L_INFO", "(Rewriting) t_relay 1 - M=$rm RURI=$ru F=$fu
T=$tu IP=$si ID=$ci\n");
sl_reply_error();
}
t_on_reply("1");
t_on_failure("1");
}
Just as above you can route other sip messages as well to your asterisk box.
________________________________
From: Michael <[email protected]>
To: OpenSIPS users mailling list <[email protected]>
Sent: Thursday, July 21, 2011 2:56 PM
Subject: Re: [OpenSIPS-Users] Using OpenSIPS as Proxy to Asterisk
On Thu, Jul 21, 2011 at 12:20 PM, Gareth Blades <[email protected]>
wrote:
>The only documentation I know of is
>http://www.opensips.org/Resources/DocsTutAsterisk
>
In this tutorial, the extensions register to OpenSIPS, while we would like to
use it only as a Proxy and not as a Registrar, so that a few of the extensions
would be able to register to Asterisk and make/receive calls to/from it, going
through OpenSIPS, as a simple relay only. I assume it should be possible.
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