Hi All,from what i'm understanding, Asterisk is back to back user agent. Base on this my initial thought was even if we enable reinvite in sip.conf, asterisk still will be in sip path after transfer. But i read some information in asterisk using refer to transfer a call completely to another sip or per say, a call comes in from voip provider and get transferred by asterisk to a cell phone number by using same provider and then asterisk will not be in SIP path anymore. is it doable ?
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