<div dir="ltr">Hi All,<div>from what i'm understanding, Asterisk is back to back user agent.</div><div>Base on this my initial thought was even if we enable reinvite in sip.conf, asterisk still will be in sip path after transfer.</div>
<div>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.</div>
<div>is it doable ?</div><div><br></div></div>