[Kamailio-Users] Openser-Asterisk Codec conversion..
Alex Balashov
abalashov at evaristesys.com
Sat Jan 24 21:21:51 CET 2009
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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--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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