Hello<div><br></div><div>I've checked out the latest trunk, the problem is still left.</div><div>Im using a Linksys SPA8000 analogue telephone adapter as one device. The other call comes through the sip gateway (from PSTN).<br>
<br></div><div>I'll try to specify 30ms.</div><div><br></div><div>Regards,</div><div> Jonas</div><div><br><div class="gmail_quote">On Thu, Dec 18, 2008 at 6:20 PM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">It seems to be related to 20ms vs 30ms ptime.<br><br>What are the 2 devices and what rev of FS are you on?<br><br>There was more code added in the last few weeks to smooth out this occurrence.<br>
<br>you can also opt to declare your codec prefs at 30ms<br>
<br>PCMU@30i instead of just PCMU in the conf.<br><br><br><br><div class="gmail_quote"><div><div></div><div class="Wj3C7c">On Thu, Dec 18, 2008 at 11:05 AM, Jonas Gauffin <span dir="ltr"><<a href="mailto:jonas.gauffin@gmail.com" target="_blank">jonas.gauffin@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="border-left:1px solid rgb(204, 204, 204);margin:0pt 0pt 0pt 0.8ex;padding-left:1ex"><div><div></div><div class="Wj3C7c"><span><div>Hello</div><div><br></div>I got problems with choppy voice. I just happens1 time of 10 or something like that. </span><div>
<span></span><br><div><span>Incorrect call: <a href="http://pastebin.freeswitch.org/6479" target="_blank">http://pastebin.freeswitch.org/6479</a></span></div>
<div><span>working call: <a href="http://pastebin.freeswitch.org/6481" target="_blank">http://pastebin.freeswitch.org/6481</a></span><br><div><br></div><div>Any idea what is wrong?</div><div><br></div>
<div>Thanks,</div><div> Jonas</div></div></div>
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