[asterisk-users] DUNDi and reinvites...

[Bryan Laird]

I'm talking out my rear so someone please apply an attitude  
adjustment if I'm way off base.

But, if you are using Dundi as a lookup engine it should know the  
contact information both endpoints and how to reach them perhaps not  
ONLY knowing how to comunicate via another asterisk box.
Much like simply initializing a base dns infrastructure for the CPE  
devices.  If the CPE devices are configured to accept SIP  
transactions from $domain or both asterisk servers server A should be
able to send a invite directly to client B and bring up the "inbound"  
call.  As far as the client knows it's still
talking and placing outbound calls with server B.

IE:
	Client A calls Client B
	Client A hits Serv A.
	Serv A does lookup finds it knows about Client B
	Serv A sends the call direct to Client B's IP.

	I'm assuming that both servers are acting as mirrors of eachother,  
in that voicemail and all that is a //shared// resource.. so if  
Client B rings unavail/busy that your serv A knows
	what to do with the call.  In general as long as a client device  
knows to understand and accept sip messages from $host an inbound  
call does not have to come from the server they registered to.

	If you look at a linksys adapter this is one of the reasons they  
have that "domain" parameter which controls the list of hosts that  
are allowed to send SIP transactions to the unit.


Am I wrong on this?  The only other artifact I can think of is the  
fact of NAT traversal, where if client B that's to recieve the call  
is behind a NAT firewall and you are not doing port forwarding of the  
SIP signaling
then ofcourse it won't get the call because server A has not  
established the NAT association.  But assuming you are using a common  
'sbc' or gatekeeper (ser) that box would know the association and things
would be happy.



On Jun 7, 2007, at 7:11 PM, Jared Smith wrote:

> On 6/7/07, Douglas Garstang <DGarstang at interainc.com> wrote:
>> That's all fine and good until
>> it becomes the receiving phone, and the other phone (as an  
>> originator)
>> also has canreinvite set to yes. Then, your back to both Asterisk
>> servers being completely taken out of the loop again!
>
> While I haven't taken the time to actually try this, I might suggest
> that you could set up separate  user and peer sections in sip.conf, so
> that you can handle inbound calls differently that outbound calls.
>
> -Jared
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
Bryan Laird, Sr. Manager CM Operations
                        -+-
Cablemodems are the gateway to the Internet.
The Internet is a gateway to some things that are  better left un-seen.