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
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 Iñaki 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).
-- Iñaki Baz Castillo ibc@aliax.net
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