<br><pre><tt><tt>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<br> most of the users are behind PBXes (having individual extensions)<br> themselves and almost all of the PBXes send a 200 OK and then play out the<br> 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.</tt></tt></pre><br><p> 
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