<html><head><style>body{font-family:Helvetica,Arial;font-size:13px}</style></head><body style="word-wrap:break-word;line-break:after-white-space"><div id="bloop_customfont" style="font-family:Helvetica,Arial;font-size:13px;color:rgba(0,0,0,1.0);margin:0px;line-height:auto">I had similar problem and it turned out the problem was loose routing </div><div id="bloop_customfont" style="font-family:Helvetica,Arial;font-size:13px;color:rgba(0,0,0,1.0);margin:0px;line-height:auto"><br></div><div id="bloop_customfont" style="font-family:Helvetica,Arial;font-size:13px;color:rgba(0,0,0,1.0);margin:0px;line-height:auto">the provider I had did not support it. I changed provider…</div><div id="bloop_customfont" style="font-family:Helvetica,Arial;font-size:13px;color:rgba(0,0,0,1.0);margin:0px;line-height:auto"><br></div> <br> <div id="bloop_sign_1533228914925479936" class="bloop_sign"></div> <br><p class="airmail_on">Il giorno
2 agosto 2018 @ 11:22:20, Daniel Grotti (<a href="mailto:dgrotti@sipwise.com">dgrotti@sipwise.com</a>) ha scritto:</p> <blockquote type="cite" class="clean_bq"><span><div text="#000000" bgcolor="#FFFFFF"><div></div><div>
<title></title>
<tt>Hi,<br>
start to have a look at the kamailio-lb.log and the
kamailio-proxy.log in order to understand why the call is
drop.<br>
As Jon suggested already, it could related to the missing ACK
received from the callee party.<br>
The log (and a SIP trace with sngrep on SPCE) will help you to
understand where could be the problem.<br>
Most likely SPCE does not receive the ACK at all from caller
side.<br>
<br>
Cheers,<br>
<br></tt>
<pre class="moz-signature" cols="72">--
Daniel Grotti
Head of Customer Support
Sipwise GmbH, Campus 21/Europaring F15
AT-2345 Brunn am Gebirge
Office: +43(0)130120332
Email: <a class="moz-txt-link-abbreviated" href="mailto:dgrotti@sipwise.com">dgrotti@sipwise.com</a>
Website: <a class="moz-txt-link-freetext" href="https://www.sipwise.com">https://www.sipwise.com</a>
</pre>
<div class="moz-cite-prefix">On 08/02/2018 06:22 AM, Alfirus Ahmad
wrote:<br></div>
<blockquote type="cite" cite="mid:em6c72e4bc-0aa8-4cbe-9ca8-883ea346518b@laptop-ly700">
Hi,
<div><br></div>
<div>I installed Sipwise CE at home for testing purpose. I'm using
dynamic IP and my router model is D-link model name:
DIR-850L.</div>
<div><br></div>
<div>I already disable SIP Application Level Gateway (ALG)
Configuration and set DMZ to the Sipwise server.</div>
<div><br></div>
<div>User A inside LAN -- call -- User B inside LAN = Fine</div>
<div>User A outside LAN -- call -- User B inside LAN = 32 sec drop
& can't do video call</div>
<div>User A outside LAN -- call -- User B outside LAN = 32 sec
drop & can't do video call</div>
<div><br></div>
<div>Is there a solution? If need to change router, what model i
should use? I read setting STUN server can overcome the problem and
how to do that in sipwise?</div>
<div><br></div>
<div>Thanks,</div>
<div>Alfirus Ahmad</div>
<div><br></div>
<div><br></div>
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