<div dir="ltr"><div>FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git ae069dc 2014-09-29 19:49:48Z 64bit)<br><br></div>Current version on the server<br></div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Nov 3, 2014 at 11:44 AM, Joel White <span dir="ltr"><<a href="mailto:joelewhite@gmail.com" target="_blank">joelewhite@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>I will work on getting a sip trace. I am seeing this happen when monitoring the system. The calls sound fine and then it sounds like the audio slows to a crawl. It does not appear to happen regularly and I have seen it happen on individual calls and on conferences. <br><br></div>The conferences have external DIDs for outside participants. The internal extensions dial a 5 digit extension.<br><br><br><br></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Nov 3, 2014 at 10:42 AM, Brian West <span dir="ltr"><<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">What rev are you on? How are you replicating this? Can you get a siptrace and logs and file a JIRA so we can get this bug fixed. Sounds like the codec is probably changing during the call, are you doing a hold/unhold operation?</div><div class="gmail_extra"><br><div class="gmail_quote"><span>On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge <span dir="ltr"><<a href="mailto:blasterjr@gmail.com" target="_blank">blasterjr@gmail.com</a>></span> wrote:<br></span><div><div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Joel is there any way you can test with one polycom set to g711u.<br><br>This sounds to me like a 16k vs 8k transcoding issue, maybe setup your codecs on the freeswitch side to only allow 8k g722 (if that's even possible)<br></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Tue, Oct 21, 2014 at 8:22 AM, Joel White <span dir="ltr"><<a href="mailto:joelewhite@gmail.com" target="_blank">joelewhite@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr">Also, just to mention. The CPU is never above 2% and the current load is about 20 simultaneous calls <br></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Oct 21, 2014 at 10:16 AM, Joel White <span dir="ltr"><<a href="mailto:joelewhite@gmail.com" target="_blank">joelewhite@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div><div>Is this an issue with transcoding?<br><br></div>I am wondering how I lock it down so that transcoding only happens at a minimum<br><br></div>Also, would it be bad to completely remove G722 from the codec list?<br><br></div>I think the issue is that our provider is sending G711 and all of the Polycom Phones are trying to use G722<br></div></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Oct 20, 2014 at 11:43 AM, Joel White <span dir="ltr"><<a href="mailto:joelewhite@gmail.com" target="_blank">joelewhite@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div>I am having an issue where some calls start out fine and then go into slow motion (distorted slow speech)<br><br></div>I noticed something in the logs stating <br><br> Activating write resampler<br><br><br></div>What does this mean and how do I avoid this in the future?<br><br><br><br></div>Thank you in advance <br></div>
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<p><font face="courier new, monospace"><b><i><font size="4">Brian West</font></i></b><br><span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
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