[Asterisk-Users] SIP QUESTION

[Jamie Carl]

Seeing as no one else has replied, I figured I may give it 
a shot.  At least it'll start something.

Now, correct me if I'm wrong someone, but as far as I 
understand in this situation you can do both.  Normally 
the RTP packets would be swtiched through *, but you can 
set in you sip.conf file the 'canreinvite=yes' option 
which will allow the RTP stream to be direct if a 
compatible codec is negotiated.

I'll double check if I ever get my server up and running 
again.

J

On Tue, 19 Aug 2003 11:17:20 -0500
  "Jorge Cisneros Flores" <jorge at redenlaces.com.mx> wrote:
>Hi 
>
>
>   Is posible to make a call from site A to Site C, and 
>my question is, the rtp data is from A to C or is from A 
>to B to C
>
>
>
>
>    Site A                             Site B 
>                                     Site C
>    ata186<-------->FW<--------->Asterisk<--------->FW<----------->ata186
>
>Thanks

Regards,

Jamie Carl
Jazz Inc.
Email:  me at jazz-inc.net
Web:    www.jazz-inc.net
Phone:  +61-414-365-466
Jabber: jazz at netmindz.net