Carmelo,<div>If you have an SIP peer that matches the host and port of the opensips server.. ie:</div><div>[opensips]</div><div>type=friend</div><div>host=<ip of opensips.</div><div>port=<port of opensips> (can be omitted if port 5060)</div>
<div><br></div><div>Then it'll match that.. typically if it's coming from opensips you'll want to add:</div><div>insecure=invite</div><div><br></div><div>so that opensips won't be challenged to authenticate. Also be sure there is no secret set.</div>
<div><br></div><div>I personally wouldn't do this using the default context as the other posters had recommended as that will allow *anyone* to send traffic to your asterisk server. Which I don't believe is what you really want to do. Instead, create a peer that is limited by IP and PORT allowed to send invites without a secret.</div>
<div><br></div><div>Also be sure that the context for that peer is set to the right context and that if from the asterisk CLI you type:</div><div>dialplan show <RURI username>@<opensips context></div><div>that it matches something you'd expect.</div>
<div><br></div><div>On another note, are you performing a consume credentials? I think it *might* be possible that opensips is forwarding your UAC's credentials on to Asterisk if you are not..</div><div><br></div><div>
-Brett</div><div><br></div><div><br><div class="gmail_quote">On Tue, May 4, 2010 at 8:02 AM, wüber <span dir="ltr"><<a href="mailto:leone81@gmail.com">leone81@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<br>
Hi Bogdan,<br>
<br>
connecting Opensips with Asterisk I can see that if a client registered on<br>
Opensips server tries to make a call to a client in Asterisk domain, after<br>
the INVITE, it receives a "forbidden" message from asterisk. I have set the<br>
forwarding functionality in Opensips (rewriteuri function) and I'm pretty<br>
sure it's something related to asterisk.<br>
<br>
Perhaps this is not the right section, but anyway could you help me? Do you<br>
know what I should set in the sip.conf of Asterisk config file?<br>
<br>
Thanks a lot,<br>
Carmelo<br>
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