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

[Benjamin Jacob]

Apologies for not explaining the set up .

Using AstMan API, I Originate a call to user A. User A is a conference bridge which needs pin authentication. So post 200 OK, I need to send DTMFs for that pin. 
After sending the pin, I Dial (using the Originate context) user B. Now user B is behind a PBX, so I need to dial the extension for user B. I send the extension digits using DTMFs again.

So, if I set canreinvite=yes, as soon as I get a 183/200 OK from user B, re-Invites are sent to both participants with the other's SDP. 

So, my question : once the SDPs are exchanged, what will happen to the DTMFs sent by Asterisk using sendDTMF or the D option in dial.

Another scenario would be to call user B first and then user A first. The same case applies over there as well.

Is there any other way to tell asterisk when to do a re-Invite/control the timing of the re-Invite?

Hope I am clear this time.

cheerz
- Ben.


Steve Davies <davies147 at gmail.com> wrote: On 21/04/2008, Benjamin Jacob  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

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