[Asterisk-Users] Transfer

[Jean-Michel Hiver]

Tomislav Parcina a écrit :

>Can * transfer call if I use canreinvite=yes in sip.conf?
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My understanding is that canreinvite only redirects the media path. 
Signaling and media are separate with SIP (which is what makes it so 
nice by the way).

>Can * start "automon" (recording) if I use canreinvite?
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Dunno. If it works at all, asterisk will have to redirect the media 
stream to itself. Have you tried?