Davida,
You would also want to look at canreinvite option in sip.conf
http://www.voip-info.org/wiki-Asterisk+SIP+canreinvite
cheerz
- Ben.
Eric "ManxPower" Wieling wrote:
> David Alcott wrote:
>
>>
>> Is there a way to configure the Asterisk so that the RTP goes
>> directly between the Endpoints as opposed to going through the
asterisk?
>
>
> That is the default if Asterisk believes it will work. Things that
> might not make it work is tTwW options to Dial, protocol transation
> (one leg is SIP, the other is IAX2, transcoding, NAT, or many other
> things that make the two legs of the call not compatible with reinvites.
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