canreinvite=yes
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Convergent Technologies Core
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brazil
+55 54 2104-7000
- Arno Scholz [EMAIL PROTECTED] escreveu:
Hello,
I'm implementing a VoIP client and using Asterisk 1.4. The RTP
transfer
should be handled in a direct connection from client to client. But
the
Asterisk server does not reveal the IP address of the peer in the
contact header field of a SIP request nor in the connection header
field
of the SDP message. Instead he always writes its own address.
So the clients are forced to handle the RTP transfer over the Asterisk
server.
Is there a possibility to configure the Asterisk server that he does
not
replace the peer IP address with his own?
I hope I could describe my problem properly.
Regards,
Arno
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