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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>How about AGI scripting? Cant I dial with that script? <o:p></o:p></span></p>
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color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
users-bounces@lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] <b>On
Behalf Of </b>Neill Wilkinson<br>
<b>Sent:</b> Saturday, January 24, 2009 4:00 PM<br>
<b>To:</b> users@lists.kamailio.org<br>
<b>Subject:</b> Re: [Kamailio-Users] Openser-Asterisk Codec conversion..<o:p></o:p></span></p>
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<p class=MsoNormal style='margin-bottom:12.0pt'>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.<o:p></o:p></p>
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<p class=MsoNormal style='margin-bottom:12.0pt'>Neill...;o)<o:p></o:p></p>
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<p class=MsoNormal>2009/1/23 Rawshan Iajdani <<a
href="mailto:iajdani@provati.com" target="_blank">iajdani@provati.com</a>><o:p></o:p></p>
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<p><span style='font-size:11.0pt;color:#1F497D'>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><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p>
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<p><b><span style='font-size:10.0pt'>From:</span></b><span style='font-size:
10.0pt'> <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><o:p></o:p></p>
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<p> <o:p></o:p></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.<o:p></o:p></p>
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<p style='margin-bottom:12.0pt'>Neill....;o)<o:p></o:p></p>
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<p>2009/1/23 Iņaki Baz Castillo <<a href="mailto:ibc@aliax.net"
target="_blank">ibc@aliax.net</a>><o:p></o:p></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>
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<a href="http://lists.openser-project.org/cgi-bin/mailman/listinfo/users"
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