[asterisk-users] The call is established but without exchanged voice packets

[Matthew J. Roth]

Asmaa, 

You're getting ahead of yourself.  How do you expect audio to work if
your firewall/NAT settings aren't even configured correctly to
establish SIP sessions?

Go back and read the message that I sent yesterday.  Fix the SIP 
three-way handshake problem.  That is step 1 and you'll know you have
it right when you stop seeing 'Retransmission timeout reached on
transmission' errors.

You still won't have audio but that's step 2.  It requires properly
configuring Asterisk's NAT settings and the firewall(s) between the
phones and the server to allow RTP traffic to flow, but don't worry
about it until step 1 is complete.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer