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 <[EMAIL PROTECTED]> 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
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-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
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.
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