asterisk users - Apr 2008 - re-invite (bypass asterisk) post call establishment

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.








     
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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

Hi again,
I tried this again, but the reInvite happens immediately after the 200 OK/ACK.
And then the D() specified DTMF is sent.

Attached is the SIP trace for the calls.
I call (from Asterisk) - 0119198807xxxxx 
After connect, I dial - 31927xxxxx.
This number 31927xxxxx is the conference bridge and I need to send DTMF (the
bridge PIN) to it after connection. But alas, the reinvite happens before the
D() is executed.
The SIP gateway is MySIPGateway at 204.aaa.bbb.ccc. 

cheers
- Ben.




Steve Davies <davies147 at gmail.com> wrote: 2008/4/22 Benjamin Jacob :
[snip]
>
> 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.
>
[snip]

As far as I can tell, the D() option will be executed before the
re-invite takes place, so Asterisk will still be in-line. I believe
that the dial is not considered "complete/connected" until the D() is
finished.

Cheers,
Steve

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