<br>Apologies for not explaining the set up .<br><br>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. <br>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.<br><br>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. <br><br>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.<br><br>Another scenario would be to call user B first and then user A first. The same case applies over there as well.<br><br>Is there any other way to tell asterisk when to do a re-Invite/control the timing of the re-Invite?<br><br>Hope I am clear this time.<br><br>cheerz<br>-
Ben.<br><br><br><b><i>Steve Davies <davies147@gmail.com></i></b> wrote:<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> On 21/04/2008, Benjamin Jacob <ben4asterisk@yahoo.com> wrote:<br>><br>><br>> Hello ppl,<br>> Any way to do a re-invite and make RTP bypass Asterisk, after call<br>> establishment.<br>> In other words, I would like to control when to do the bypass work for<br>> peer-peer RTP flow.<br>> The issue is that I need to send DTMFs after dialing the user because most<br>> of the users are behind PBXes (having individual extensions) themselves and<br>> almost all of the PBXes send a 200 OK and then play out the PBX messages.<br>> So I need to send the extension DTMFs first, bridge the calls and then<br>> re-invite users for them to do a peer-peer rtp conversation.<br>><br>> TiA,<br>> - Ben.<br><br>You don't say what you've tried already, but as long
as<br>canreinvite=yes is set against the SIP peer, the RTP stream should be<br>redirected once the connection is open.<br><br>As far as DTMF to dial an extension at the remote end, have you looked<br>at the D() parameter to the Dial command?<br><br>Regards,<br>Steve<br><br>_______________________________________________<br>-- Bandwidth and Colocation Provided by http://www.api-digital.com --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-users<br></ben4asterisk@yahoo.com></blockquote><br><p> 
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