[Asterisk-Users] Asterisk as a SIP/H.323 Router

[Abdul Hakeem]

Hi,
Extensions are different from H323 endpoints.
My understanding of Cisco routers and Sip/H323 dictates that there
should be a Proxy when routing from 1 Sip/H323 gateway to another.
We have tried it before and the best we ever achieved is a one-way audio
in the most minimalist form.. i.e. Cisco A---->Cisco B------Cisco C.
In your configs below, you will be routing to Gateways not endpoints,
and that's where the problem is.
Perhaps, someone else has tried this with better luck, but I doubt it.
Cheers,
Abdul
-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Nir
Simionovich
Sent: Monday, March 17, 2003 3:33 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Asterisk as a SIP/H.323 Router


Ok, 
 
  Lets say that I have something like the following:
 
  On the Asterisk box I have the following definitions in sip.conf:
 
[locationA]
type=friend
host=router.locationA.foobar.com
defaultip=1.1.1.2
context=router
 
[locationB1]
type=friend
host=router.locationB1.foobar.com
defaultip=1.1.2.2
context=router
 
[locationB2]
type=friend
host=router.locationB2.foobar.com
defaultip=1.1.2.3
context=router

  Now, in extensions.conf I would have something with the following
logic:
 
[router]
;
; We start with what to do when a call first comes in.
;
exten => s,1,Wait,1
exten => s,2,[AGI program that checks the originator and routes
according to pre-defined settings]
 
  Again, this is just a concept, not the actual configuration. But in
theory, I don't a reason why this can't
work.
 
Nir Simionovich
 
 

----- Original Message ----- 
From: Abdul  <mailto:alhakeem at softhome.net> Hakeem 
To: asterisk-users at lists.digium.com 
Sent: Monday, March 17, 2003 4:19 PM
Subject: RE: [Asterisk-Users] Asterisk as a SIP/H.323 Router

Hi,
I took a look at the architecture.
The way the Cisco boxes will work if Asterisk in the middle is a Proxy,
but it's not.
You cannot re-direct an incoming Voip call from one gateway to another,
only a proxy can do that.
The Asterisk in this mode can only terminate the calls via the PSTN. If
it attempts to re-direct the call to the Cisco, it has to be via it's
PRI interface (i.e. Cisco PRI0-3 is connected to the PRI interface of
the Asterisk Location B.
Cheers,
Abdul
-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Nir
Simionovich
Sent: Monday, March 17, 2003 10:08 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Asterisk as a SIP/H.323 Router


Hi All,

  I've been spending the last month experimenting with Asterisk, and I
must say that all results point
to a very positive outcome. 

  Now, i've been asked the following question: Is it possible to put an
Asterisk box between 2 Cisco
routers or other SIP complianet equiment, then routing SIP/H.323 calls
between the two routers?
 
  Here's a drawing that will explain:
 
      +--------------+                   +----------+
+--------------+
 PRI  |              |1.1.1.2     1.1.1.3|          |1.1.2.1   1.1.2.2|
|
 ---->+ Cisco Router +-------------------+ Asterisk +--------+--------+
Cisco Router +--> PRI0-3
      | Location A   |                   |Location B|        |        |
|
      +--------------+                   +----------+        |
+--------------+
                                                             |
                                                             |
+--------------+
                                                             | 1.1.2.3|
|
                                                             +--------+
Cisco Router +--> PRI4-7
                                                                      |
|
 
+--------------+
 
The question here is this:
 
A phone call is is made to the router in location A.
The Cisco router routes the call via SIP to the Asterisk Box at Location
B.
According to a set of rules on the Asterisk box, the box would route the
SIP/H.323
to one of the the other Cisco boxes, in order to terminate the call.
 
What do you think, is this possible?
 
Nir Simionovich
 
P.S.
  Excuse me for the poor ASCII art, I didn't want to attach a Visio or a
PDF file.

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