<div dir="ltr">Yea, update to master helps. <div>Thanks.</div></div><div class="gmail_extra"><br><div class="gmail_quote">2016-05-05 17:43 GMT+03:00 Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Confirm if this is still an issue with master. I recall fixing something around this space. If it is still an issue in master, please file a jira.<br>
<span class=""><br>
> On May 5, 2016, at 10:06 AM, Igor Olhovskiy <<a href="mailto:igorolhovskiy@gmail.com">igorolhovskiy@gmail.com</a>> wrote:<br>
><br>
> Hi!<br>
><br>
> I have an issue with a mod_conference. Now profile is set to default. But when one phone is connecting with G722 and other with PCMA, one with G722 has very distorted sound. This all is within virtual environment (VmWare).<br>
> I know same problems with Asterisk MeetMe was about timings and modules. Is there any option for FreeSwitch to set in conference profile desired codec string or so? Cause even now it set to rate 8000 and still uses 16000 for G722 in same room.<br>
> Freeswitch 1.6.6<br>
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</blockquote></div><br><br clear="all"><div><br></div>-- <br><div class="gmail_signature"><div dir="ltr">Best regards,<div>Igor</div></div></div>
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