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[ben4asterisk at yahoo.com]

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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[davies147 at gmail.com]

On 21/04/2008, Benjamin Jacob <ben4asterisk at yahoo.com> wrote:

[ben4asterisk at yahoo.com]

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:

[davies147 at gmail.com]

2008/4/22 Benjamin Jacob <ben4asterisk at yahoo.com>:
[snip]

[ben4asterisk at yahoo.com]

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]