[Asterisk-Users] Transfer

Tomislav Parcina tparcina at lama.hr
Fri Dec 23 04:17:38 MST 2005


Can * transfer call if I use canreinvite=yes in sip.conf?
Can * start "automon" (recording) if I use canreinvite?

If answers are no, then which one did you chouse for your configuration? 
Do you use "canreinvite=yes" so you can't do those stuff or you don't 
use this so you have high processor load?


-- 

Tomislav Parcina
ime.prezime at email.t-com.hr




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