It is my understanding that the canreinvite command within Asterisk is used to directly connect the RTP audio to the destination IP phone. So if an IP phone is remotely registered to TrixBox and a SIP call comes in from another Trixbox phone or from SIP provider, the canreinvite=yes command would then directly connect the RTP traffic between the two.
There may be more to it, but is the essentially correct?
 
Member Since:
2006-06-25