Hello,
I have very simple dial plan like that:
exten => _011XXXX.,1,Dial(SIP/sipprov/00${EXTEN:3})
exten => _011XXXX.,2,Hangup
I have canreinvite=yes and nat=no for both incoming and outgoing less.
I expected that upon call establishment Asterisk will send reINVITE ob both legs to remove itself from RTP path.
It doesn’t happen. I don’t have SDP in initial incoming INVITE (SDP coming later in ACK as SDP answer). Can it be a reason, that Asteisk doesn’t try to renegotiate media upon call establishment? Is there any way to solve that, assuming that I can not provide SDP in initial INVITE?
Thank you,
Boris