Seem like my past msg did not go thru .... second attempt ....<br><br><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div><div class="gmail_quote">
<div class="im"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">At the moment the code basically works and now I want to figure out how<br>
to best fit it in Asterisk.<br><br></blockquote></div></div></div></blockquote><div> </div><div><div>Hi Tzafrir,</div><div><br></div><div>This might be a silly question and I'm sure you had contemplated this option before, but I'd like to know what made you discard it (notice I am not familiar with res_monitor and this suggestion is more oriented to mix monitor/audio hooks). </div>
<div><br></div><div>From the very first time I saw your branch seemed like a cool project to me, something I did not understand though, is why not create a regular SIP call instead of a dummy call (that is calling ast_request(), ast_call() etc to a configured SIP peer), then anything read from the monitored channel would be sent using ast_write() and audio coming back from the SIP call would be silently discarded ( may be you decided is a waste? ). </div>
<div><br></div><div>As I said, however, this thought I had was more oriented to using it inside mix monitor with audio hooks sending the mixed audio already to the monitoring SIP call (or any other asterisk tech channel for that matter) instead of writing it to a file. </div>
<div><br></div><div><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; ">-- <br>Moises Silva<br>Software Developer<br>Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada<br>
t. 1 905 474 1990 x 128 | e. <a href="mailto:moy@sangoma.com" target="_blank" style="color: rgb(66, 99, 171); ">moy@sangoma.com</a></span></div></div></div>