Hi, <div>I think this step is also mentioned in any tutorial for integrating asterisk with opensips. But anyway don't forget to reply the exact config set which worked for you here for others to get help from.</div><div>
<br></div><div>Regards,<br>Sammy</div><div><br><div class="gmail_quote">On Mon, Dec 5, 2011 at 8:49 PM, Matt Hamilton <span dir="ltr"><<a href="mailto:mistral9999@hotmail.com">mistral9999@hotmail.com</a>></span> wrote:<br>
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<br><div><div><div class="im">>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></div>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>
<div class="im"><br><div>On Sun, Dec 4, 2011 at 9:28 PM, Matt Hamilton <span dir="ltr"><<a href="mailto:mistral9999@hotmail.com" target="_blank">mistral9999@hotmail.com</a>></span> wrote:<br>
<div><blockquote style="border-left:1px #ccc solid;padding-left:1ex">
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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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