[Asterisk-Users] SIP Reinvites or changing the codec mid-call

[Nathan]

Is it possible to change the codec for a SIP session, midway through a call? 
I.E. I wish to start a session off as ULAW, detect a fax (using 
NVFaxDetect), if one is detected keep the codec as ULAW, if one is not 
detected change the codec to G723.. I wish to do this transparently (no 
dialing extra digits, etc...)

Thanks!

Nathan