<p>Freeswitch is turning into a mean, mean event-driven voip panacea...<br>
Congrats on the new commit.</p>
<div class="gmail_quote">On Feb 22, 2013 6:49 AM, "Emrah" <<a href="mailto:lists@kavun.ch">lists@kavun.ch</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi guys,<br>
<br>
Sorry was disconnected from the list for a while.<br>
Indeed it looks like the latest commit fixed it.<br>
The issue I see now is that the channel info are not updated and a show channels will only list the initial codecs.<br>
<br>
Great job and thanks a bunch,<br>
Emrah<br>
On Feb 21, 2013, at 8:42 PM, Dmitry Lysenko <<a href="mailto:dvl36.ripe.nick@gmail.com">dvl36.ripe.nick@gmail.com</a>> wrote:<br>
<br>
> Now this cool feature is fully working! Today Anthony committed the bugfix!<br>
> Tested with 2 different SIP UA<->FS<->Callcentric. Both legs.<br>
> Now possible to write application that will switch codecs on the fly, regarding on any event, such as switching to low bandwidth backup internet line or too high cpu load, etc.<br>
> BTW, is there way to get API access to statistic of jitter buffer?<br>
> Thanks.<br>
><br>
> P.S. As I know, asterisk can't do such thing.<br>
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</blockquote></div>