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<div class="moz-cite-prefix">Hi Ram,<br>
First, excuse me on my English which is not well. I'm sending
files with two calls. The first call from PSTN (20624338 to
901001) is on SIP client on my mobile phone and it's OK. Second
call from PSTN (20624338 to 901000) is on X-lite client behind
ADSL router and it hasn't one rtp stream. Please, write if you
need some other files and logs.<br>
<br>
Zoran<br>
<br>
On 10.2.2015 11:58, ram wrote:<br>
</div>
<blockquote
cite="mid:CA+XvL-PfPbNLePpAqyKkgeq2mR9Jh05HhSryuZhu+Fh8Q_r84A@mail.gmail.com"
type="cite">
<div dir="ltr">Hi Zoran,
<div><br>
</div>
<div>Post the details call logs to assist you better.</div>
<div><br>
</div>
<div>Ram</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Mon, Feb 9, 2015 at 8:15 AM, Zoran
Brajovic <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:zoranbrajovic@t-com.me" target="_blank">zoranbrajovic@t-com.me</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"> <font>Hi,</font>
<p style="margin-bottom:0in" lang="sh-YU"><font>We have
installed SPCE mr.3.7.7 besides our Soft Switch which
works in our case as Class 4 Switch for calls to and
from PSTN. Calls between UAs on SPCE and from UA to
PSTN are OK regardless is UA nated or not. But call
from PSTN to UA, when UA is nated, has only rtp from
UA to PSTN and call drops after 30 sec. If I press
HOLD and RESUME on X-Lite client during the call, it
sends new INVITE message and rtp goes in both
direction without time limit. In this scenario if UA
is not nated call is OK, too. Please for help.</font></p>
</div>
<br>
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</blockquote>
</div>
<br>
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</blockquote>
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