<br><br><div class="gmail_quote">On Thu, Oct 22, 2009 at 3:59 PM, Tihomir Culjaga <span dir="ltr"><<a href="mailto:tculjaga@gmail.com">tculjaga@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div class="im"><div class="gmail_quote"><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>
<div>TC>Hi, here is the FS log without crash-protection:<br>
</div>TC><a href="http://pastebin.freeswitch.org/10796" target="_blank">http://pastebin.freeswitch.org/10796</a> and here is the backtrace:<br>
TC><a href="http://pastebin.freeswitch.org/10797" target="_blank">http://pastebin.freeswitch.org/10797</a><br>
<br>
i fix this crash already, please download latest version from same link<br>
as previous, recompile and try again.<br>
<br></blockquote></div><br></div>outgoing works, I can place an end-to-end call ... except the RTP is
realy delayed ... after approx 30 sec of conversation the audio is
delayed more than 10 seconds.... but i have 2 way audio for outgoing
calls:)<br>
<br></blockquote><div><br>one more thing ... it is H323 endpoint => SIP phone audio that is delayed. SIP phone => H323 endpoint is ok!<br>
<br>
T.<br> </div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Do you need some logs ?<br><br><br>Inbound cals still the same... i suppose you didn't have a chance working on that as well ...<br><font color="#888888"><br>T.<br>
</font></blockquote></div><br>