[asterisk-users] re-invite (bypass asterisk) post call establishment

[Steve Davies]

On 21/04/2008, Benjamin Jacob <ben4asterisk at yahoo.com> wrote:
>
>
> Hello ppl,
> Any way to do a re-invite and make RTP bypass Asterisk, after call
> establishment.
> In other words, I would like to control when to do the bypass work for
> peer-peer RTP flow.
> The issue is that I need to send DTMFs after dialing the user because most
> of the users are behind PBXes (having individual extensions) themselves and
> almost all of the PBXes send a 200 OK and then play out the PBX messages.
> So I need to send the extension DTMFs first, bridge the calls and then
> re-invite users for them to do a peer-peer rtp conversation.
>
> TiA,
> - Ben.

You don't say what you've tried already, but as long as
canreinvite=yes is set against the SIP peer, the RTP stream should be
redirected once the connection is open.

As far as DTMF to dial an extension at the remote end, have you looked
at the D() parameter to the Dial command?

Regards,
Steve