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

[Alex Balashov]

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 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 <mailto:iajdani at 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 at lists.kamailio.org 
> <mailto:users-bounces at lists.kamailio.org> 
> [mailto: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 <mailto: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 <mailto:ibc at aliax.net>>
> 
> 2009/1/23 Rawshan Iajdani <iajdani at provati.com 
> <mailto: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 <mailto:ibc at aliax.net>>
> 
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> 
>  
> 
>  
> 
> 
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-- 
Alex Balashov
Evariste Systems
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