Hi ,
I have a question about the canreinvite option in the sip extention configuration.
If i understand correctly the canreinvite option of an extention will enable
asterisk to connect 2 extentions directly instead of passing the audio via asterisk. ( and if needed it can reinvite those extentions in order to pass the audio channel through the pbx ), this seems like a good option in order to take the load off asterisk and also to avoid some bandwidth problems in case the extentions are “far” from the PBX.
my questions:
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What if i want an extention to be able to use this option for local ( inside network calls ) and for other calls not to be able to do that… ?
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Am i right about the canreinvite option ? how often is it used ?