<div>Asterisk would need credentials configured in SIP.CONF for the user to do this. Otherwise you would really need to be using a proper SBC or B2BUA with Media conversion capability.<br><br></div>
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<div>Neill...;o)<br><br></div>
<div class="gmail_quote">2009/1/23 Rawshan Iajdani <span dir="ltr"><<a href="mailto:iajdani@provati.com" target="_blank">iajdani@provati.com</a>></span><br>
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<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">Well that is what I am trying to do.. To originate the 2<sup>nd</sup> leg.. I need the username/password for authentication to the terminating server. Can I get that from OpenSer??? Because UA already logged into OpenSer..</span></p>
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<p><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;"> <a href="mailto:users-bounces@lists.kamailio.org" target="_blank">users-bounces@lists.kamailio.org</a> [mailto:<a href="mailto:users-bounces@lists.kamailio.org" target="_blank">users-bounces@lists.kamailio.org</a>] <b>On Behalf Of </b>Neill Wilkinson<br>
<b>Sent:</b> Friday, January 23, 2009 6:10 PM<br><b>To:</b> <a href="mailto:users@lists.kamailio.org" target="_blank">users@lists.kamailio.org</a><br><b>Subject:</b> Re: [Kamailio-Users] Openser-Asterisk Codec conversion..</span></p>
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<p>Or Put another Way Asterisk acts in SIP terms as a Back2Back User Agent, to terminate one side of the call let and originate a new call leg with a different codec profile in the SIP/SDP. Asterisk then terminates the inbound media, transcodes it an originates a new media stream on a completely different call leg.</p>
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<p style="margin-bottom: 12pt;">Neill....;o)</p></div>
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<p>2009/1/23 Iņaki Baz Castillo <<a href="mailto:ibc@aliax.net" target="_blank">ibc@aliax.net</a>></p>
<p>2009/1/23 Rawshan Iajdani <<a href="mailto:iajdani@provati.com" target="_blank">iajdani@provati.com</a>>:<br>><br>> UA----->OpenSer(Outbound Proxy)---------Register Server<br>> | |<br>
> |<br>> Asterisk(codec converion)----------------------<br>><br>> The UA will register to Register server through outbound proxy OpenSer. When<br>> UA makes call it first comes to Openser, OpenSer should route the media to<br>
> Register server through Asterisk for codec conversion. OpenSer will not hold<br>> any User account rather it will act as a proxy.<br><br>Asterisk cannot receive *just* the media, it needs to receive the SIP<br>signalling so then it can handle the media (and do the codec<br>
conversion).<br><br>--<br>Iņaki Baz Castillo<br><<a href="mailto:ibc@aliax.net" target="_blank">ibc@aliax.net</a>><br><br>_______________________________________________<br>Kamailio (OpenSER) - Users mailing list<br>
<a href="mailto:Users@lists.kamailio.org" target="_blank">Users@lists.kamailio.org</a><br><a href="http://lists.kamailio.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.kamailio.org/cgi-bin/mailman/listinfo/users</a><br>
<a href="http://lists.openser-project.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.openser-project.org/cgi-bin/mailman/listinfo/users</a></p></div>
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