24 Jan
2009
24 Jan
'09
10:21 p.m.
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(a)lists.kamailio.org
[mailto:users-bounces@lists.kamailio.org] *On Behalf Of *Neill Wilkinson
*Sent:* Saturday, January 24, 2009 4:00 PM
*To:* users(a)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(a)provati.com <mailto:iajdani@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(a)lists.kamailio.org
<mailto:users-bounces@lists.kamailio.org>
[mailto: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(a)lists.kamailio.org <mailto: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(a)aliax.net <mailto:ibc@aliax.net>>
2009/1/23 Rawshan Iajdani <iajdani(a)provati.com
<mailto:iajdani@provati.com>>:
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
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(a)aliax.net <mailto:ibc@aliax.net>>
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