<span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; ">Ok, so as you can see from the trace, the 200OK is retransmitted.. you'll see:<div>200OK</div><div>ACK</div>
<div>200OK</div><div>200OK</div><div>200OK</div><div>200OK</div><div>BYE</div><div><br></div><div>This is because the ACK never made it to Astersk. It's a scripting error in Opensips. You may want to check your loose route block for errors. </div>
<div><br></div><div>Also, it's worth mentioning that the behavior to hang up in 17 seconds is correct. It's saying "I never got confirmation that the call really got completed, so I'm just going to end it". What's really happening is that the 200OK is timing out.</div>
</span><br><div class="gmail_quote">On Tue, Oct 6, 2009 at 8:48 AM, Peter den Hartog <span dir="ltr"><<a href="mailto:peterdenhartog@gmail.com">peterdenhartog@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<br>
I understand you can find it under this text.<br>
as you can see, the call just disapeare, i see now that the bye appears when<br>
i hang up the polycom phone.<br>
<br>
I hope this information helps.<br><br></blockquote></div>