Re: [Kamailio-Users] Openser-Asterisk Codec conversion..

How about AGI scripting? Cant I dial with that script? From: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] On Behalf Of Neill Wilkinson Sent: Saturday, January 24, 2009 4:00 PM To: users(a)lists.kamailio.org Subject: Re: [Kamailio-Users] Openser-Asterisk Codec conversion.. 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. Neill...;o) 2009/1/23 Rawshan Iajdani <iajdani(a)provati.com> Well that is what I am trying to do.. To originate the 2nd leg.. I need the username/password for authentication to the terminating server. Can I get that from OpenSer??? Because UA already logged into OpenSer.. From: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] On Behalf Of Neill Wilkinson Sent: Friday, January 23, 2009 6:10 PM To: users(a)lists.kamailio.org Subject: Re: [Kamailio-Users] Openser-Asterisk Codec conversion.. 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. Neill....;o) 2009/1/23 Iñaki Baz Castillo <ibc(a)aliax.net> 2009/1/23 Rawshan Iajdani <iajdani(a)provati.com>om>: | When hold Asterisk cannot receive *just* the media, it needs to receive the SIP signalling so then it can handle the media (and do the codec conversion). -- Iñaki Baz Castillo <ibc(a)aliax.net> _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users