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@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
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 Iņaki Baz Castillo <ibc@aliax.net>
2009/1/23 Rawshan Iajdani <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).
--
Iņaki Baz Castillo
<ibc@aliax.net>
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