<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:times new roman, new york, times, serif;font-size:12pt"><div><br>Hello ppl,<br>Any way to do a re-invite and make RTP bypass Asterisk, after call establishment.<br>In other words, I would like to control when to do the bypass work for peer-peer RTP flow. <br>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. <br>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.<br><br>TiA,<br>- Ben.<br><br><br><br><br></div></div><br>
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