Sure, you can initiate any dial plan applications with AGI.
Rawshan Iajdani wrote:
How about AGI scripting? Cant I dial with that script?
*From:* users-bounces@lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] *On Behalf Of *Neill Wilkinson *Sent:* Saturday, January 24, 2009 4:00 PM *To:* users@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@provati.com mailto:iajdani@provati.com>
Well that is what I am trying to do.. To originate the 2^nd 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@lists.kamailio.org mailto:users-bounces@lists.kamailio.org [mailto:users-bounces@lists.kamailio.org mailto:users-bounces@lists.kamailio.org] *On Behalf Of *Neill Wilkinson *Sent:* Friday, January 23, 2009 6:10 PM *To:* users@lists.kamailio.org mailto:users@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@aliax.net mailto:ibc@aliax.net>
2009/1/23 Rawshan Iajdani <iajdani@provati.com mailto:iajdani@provati.com>:
UA----->OpenSer(Outbound Proxy)---------Register Server |
|| Asterisk(codec converion)----------------------The UA will register to Register server through outbound proxy
OpenSer. When
UA makes call it first comes to Openser, OpenSer should route the
media to
Register server through Asterisk for codec conversion. OpenSer will
not hold
any User account rather it will act as a proxy.
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@aliax.net mailto:ibc@aliax.net>
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org mailto:Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users