[Asterisk-Users] SIP QUESTIO

[Brian West]

On Wed, 20 Aug 2003, Jamie Carl wrote:

>
> 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
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