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

[Rawshan Iajdani]

How about AGI scripting? Cant I dial with that script? 

 

From: users-bounces at lists.kamailio.org
[mailto:users-bounces at lists.kamailio.org] On Behalf Of Neill Wilkinson
Sent: Saturday, January 24, 2009 4:00 PM
To: users at 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 at 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 at lists.kamailio.org
[mailto:users-bounces at lists.kamailio.org] On Behalf Of Neill Wilkinson
Sent: Friday, January 23, 2009 6:10 PM
To: users at 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 at aliax.net>

2009/1/23 Rawshan Iajdani <iajdani at 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 at aliax.net>

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