<div>Hi Folks,</div>
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<div>I had a need to detect SITs, at the time, just on SIP calls in early media, but, expected that the same requirement could be present on any channel driver... So... I modified app_dial, added an option to enable inband progress detection, plugged in a dsp call and during early media forced transcoding to slin for the dsp's consumption, switching transcoding back on answer... This was with Asterisk 1.2... It's been running pretty successfully for the past 6 months, so I want to contribute this functionality back. </div>
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<div>What I could use some feedback on is that the way I've implemented this, while fine for my needs, probably isn't the 'right way' and does cause problems with early media being bridged elsewhere when the dsp is attached due to forcing the transcoding, so while looking to rewrite for trunk I'm first looking at doing it the right way ...</div>
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<div>My thoughts on this are that perhaps duplicating the frames would make sense and feeding one half through transcoding leaving the original audio frames intact, but, also wondering whether I'd even need to do this with additions since 1.2 such as transcode_via_sln and audiohooks perhaps usable to do the main work there... </div>
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<div>Thanks,</div>
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<div>d</div>
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