Is there a way to force asterisk to take care only of sip signaling without
forcing it to take care of rtp traffic?
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Il Neofita wrote:
Is there a way to force asterisk to take care only of sip signaling
without forcing it to take care of rtp traffic?
Yes. The canonical way is to enable canreinvite=yes on both SIP peers
(incoming and outgoing legs), which will cause Asterisk to send a new
INVITE within
Hello,
we're in process of testing an interconnection with a trans-european
carrier. But the carrier wants the SIP signalling to server 1 and the
RTP stream to server 2. How do I configure asterisk to work with that
type of installation. It seems they are using NexTone as SIP Signaling
and RTP
