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. 

The conferences have external DIDs for outside participants.  The internal extensions dial a 5 digit extension.

On Mon, Nov 3, 2014 at 10:42 AM, Brian West <brian@freeswitch.org> wrote:

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?

On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge <blasterjr@gmail.com> wrote:

Joel is there any way you can test with one polycom set to g711u.

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)

On Tue, Oct 21, 2014 at 8:22 AM, Joel White <joelewhite@gmail.com> wrote:

Also, just to mention.  The CPU is never above 2% and the current load is about 20 simultaneous calls

On Tue, Oct 21, 2014 at 10:16 AM, Joel White <joelewhite@gmail.com> wrote:

Is this an issue with transcoding?

I am wondering how I lock it down so that transcoding only happens at a minimum

Also, would it be bad to completely remove G722 from the codec list?

I think the issue is that our provider is sending G711 and all of the Polycom Phones are trying to use G722

On Mon, Oct 20, 2014 at 11:43 AM, Joel White <joelewhite@gmail.com> wrote:

I am having an issue where some calls start out fine and then go into slow motion (distorted slow speech)

I noticed something in the logs stating

 Activating write resampler

What does this mean and how do I avoid this in the future?

Thank you in advance

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