<br>Hi again,<br>I tried this again, but the reInvite happens immediately after the 200 OK/ACK. And then the D() specified DTMF is sent.<br><br>Attached is the SIP trace for the calls.<br>I call (from Asterisk) - 0119198807xxxxx <br>After connect, I dial - 31927xxxxx.<br>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.<br>The SIP gateway is MySIPGateway at 204.aaa.bbb.ccc. <br><br>cheers<br>- Ben.<br><br><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;"> 2008/4/22 Benjamin Jacob <ben4asterisk@yahoo.com>:<br>[snip]<br>><br>> So, my question : once the SDPs are exchanged, what will happen to the DTMFs<br>> sent by Asterisk using sendDTMF or the D option in dial.<br>><br>[snip]<br><br>As far as I can tell, the
D() option will be executed before the<br>re-invite takes place, so Asterisk will still be in-line. I believe<br>that the dial is not considered "complete/connected" until the D() is<br>finished.<br><br>Cheers,<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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