Tiago Stein D`Agostini wrote:
Hi,
Ie been looking for some time how to use asterisk to initiate SIP
connections between 2 IP phones, but afetr initiated the
communication making the RTP go directly from one telephone to the
other, without passing by asterisk. Unfortunately I found no
explanations of how to do it.
Does anyone care to give a pointer to any explanation about how to do it?
canreinvite=yes
and look at the options for dial()
Thanks in advance
bye
Ronald Wiplinger
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