[asterisk-users] Passing call status/progress between protocols

[Michelle Dupuis]

We have a * box with sip in, and h.323 out.  When the H.323 call setup is
underway, will Asterisk translate the progress/status/result codes to SIP
automatically?  
 
Or....do we have create our own result codes in SIP headers?
 
Thanks,
MD
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