[Asterisk-Users] Asterisk 1.2.7.1 with Polycom 501 on SIP -> Conf Calling

[Mike Staver]

I have one extension setup for each Polycom 501 I have, and when I try 
to call out on a conference call, I get "all circuits busy" for the 
second call.  I have one sip trunk set up for each DID that I have 
through our VoIP provider.  Each trunk is capable of having one call 
placed on it at one time.  So, I'm thinking I need a way to tell 
Asterisk to have the second call go out on one of the other empty trunks 
at the time if one exists, which more than likely, it will.  Is this 
possible?
-- 

                                 -Mike Staver
                                  staver at fimble.com
                                  mstaver at globaltaxnetwork.com