[asterisk-users] SIP Connection Question

[Kenneth Noisewater]

Hi All,

I have a question about how a particular situation would work between two
PBX systems:

If I were to connect my Mitel box to my Asterisk box via a SIP trunk (same
rack, same network), and then pass a call from the Mitel to Asterisk to
perform some functions (lookups, maybe routing), and then pass the call back
to the Mitel to be routed to it's endpoint, would Asterisk stay in that loop
after the call was passed back to the Mitel? Or, does the call leave
Asterisk completely when passed back?

If it does leave/stay in the loop, is there a way to force it to leave/stay
based on what my needs are?

Thanks,

Kenny
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100401/c9c0e17e/attachment.htm