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<br><div><div>>You can add a OpenSIPS peer in you asterisk sip.conf without any username and password with param "insecure=invite,port" <br>>that way your asterisk would not worry about asking for credentials on incoming INVITES from opensips. <br><br>That's what I was looking for, thanks for both of your answers.<br><br>Regards,<br>Matt<br><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"></span><br></div><br><div>On Sun, Dec 4, 2011 at 9:28 PM, Matt Hamilton <span dir="ltr"><<a href="mailto:mistral9999@hotmail.com">mistral9999@hotmail.com</a>></span> wrote:<br>
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<br>I'm a little confused about Opensips doing the registrations for Asterisk.<br><br>I integrated Opensips with Asterisk Realtime (Asterisk sipusers/peers point to Opensips subscribe table via a view). Opensips does the authorization and saves the location. However, when a call comes in (INVITE is routed to Asterisk), it seems like Asterisk doesn't know about the user (or sees the users as not authorized), so can't create the SIP channel.<br>
<br>If I route the REGISTER to Asterisk after authorizing in Opensips (and doing a consume_credentials()), Asterisk does the Authorization again from scratch. In that case call goes through, but I end up duplicating the authorization process. If I don't use consume_credentials(), Asterisk's authorization challenge doesn't match with Opensips, the user is not authenticated.<br>
<br>I was hoping to take the load of handling registrations from Asterisk. I'm fairly new to both Asterisk and Opensips - what am I missing?<br><br>Thanks,<br>Matt<br><br><br><br> </div></div>
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