23 Jan
2009
23 Jan
'09
10:24 a.m.
My scenario is like that..
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.
The way I am thinking.. Openser could be used as outbound proxy that's fine.
If it were used as outbound proxy, can it also route the media? In the
asterisk side I may need username/password pair of UA to dial to Register
Server in the second leg. Can I read the User/Pasword from Openser after UA
is registered to OpenSer. I can have both OpenSer and Asterisk to reside on
the same machine with different IPs so they can access the database if
required..
23 Jan
23 Jan
12:34 p.m.
2009/1/23 Rawshan Iajdani <iajdani(a)provati.com>om>:
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>
1:09 p.m.
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>
2009/1/23 Rawshan Iajdani
<iajdani(a)provati.com>om>:
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>
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
2:38 p.m.
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(a)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
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>
2009/1/23 Rawshan Iajdani <iajdani(a)provati.com>om>:
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>
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
24 Jan
24 Jan
10:59 a.m.
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>
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(a)lists.kamailio.org [mailto:
users-bounces(a)lists.kamailio.org] *On Behalf Of *Neill Wilkinson
*Sent:* Friday, January 23, 2009 6:10 PM
*To:* users(a)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>
2009/1/23 Rawshan Iajdani <iajdani(a)provati.com>om>:
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>
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
9:20 p.m.
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>
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(a)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
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>
2009/1/23 Rawshan Iajdani <iajdani(a)provati.com>om>:
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>
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
9:21 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>>
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org>
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
------------------------------------------------------------------------
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5574
days inactive
5575
days old
6 comments
4 participants
participants (4)
-
Alex Balashov -
Iñaki Baz Castillo -
Neill Wilkinson -
Rawshan Iajdani