<div dir="ltr"><div dir="ltr"><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small">Roberto</div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small"><br></div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small">Check your router if ALG or similar feature is enabled. Disable and test.<br></div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small">Also, on SNGREP check if both parties are getting ACK correctly after RTP starts.<br></div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small"><br clear="all"></div><div><div dir="ltr" class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><b><font face="monospace, monospace">--</font></b></div><div><b><font face="monospace, monospace">Atenciosamente,</font></b></div><font face="monospace, monospace"><b><br>Luciano Moreira<br></b><b>(85)99974-2750</b></font></div><div dir="ltr"><b><font face="monospace, monospace">__<br>Logic Telecom<br></font></b></div><div><font face="monospace, monospace"><b>0800-085-7799 | (85)4042-7799 | </b><b>(11)4210-7799</b></font></div></div></div></div></div></div></div><br></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">Em ter., 22 de set. de 2020 às 13:35, Roberto <<a href="mailto:roberto.medola@gasparimsantos.com.br">roberto.medola@gasparimsantos.com.br</a>> escreveu:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div>
Hello.<br>
Thanks for the reply.<br>
<p>Yes. In the traffic analyzed, the BYE is sent by the originator
of the call, but there is no "human" hangup, but the asterisk one.</p>
BYE is sent, received and confirmed.<br>
<br>
I don't know how I could investigate the reason for this BYE.<br>
<div><br>
</div>
<div>Em 21/09/2020 17:12, Dovid Bender
escreveu:<br>
</div>
<blockquote type="cite">
<div dir="ltr">Is there anything in the Asterisk logs? Which side
sends the BYE? Were you able to capture the traffic with
sngrep/wireshark to see if any side stopped sending/getting RTP?
What did the other side see?
<div><br>
</div>
</div>
<br>
<div class="gmail_quote">
<div dir="ltr" class="gmail_attr">On Mon, Sep 21, 2020 at 3:22
PM Roberto <<a href="mailto:roberto.medola@gasparimsantos.com.br" target="_blank">roberto.medola@gasparimsantos.com.br</a>>
wrote:<br>
</div>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">Hello<br>
I have an asterisk 16.2.1 on an ubuntu on AWS, which is
experiencing a <br>
drop in call. It does not have a certain time, it is random.
The audio <br>
is flowing normally and the call is dropped.<br>
Has anyone ever experienced this?<br>
<br>
My settings changed below:<br>
<br>
allowoverlap = no<br>
udpbindaddr = 0.0.0.0<br>
tcpenable = no<br>
tcpbindaddr = 0.0.0.0<br>
<br>
transport = udp, ws, wss<br>
<br>
srvlookup = yes<br>
<br>
directmedia = no<br>
<br>
rtcachefriends = yes<br>
<br>
externaddr = my ip address<br>
<br>
externhost = my domain address ; <a href="http://foo.dyndns.net" rel="noreferrer" target="_blank">foo.dyndns.net</a>;
refreshed periodically<br>
externrefresh = 180<br>
<br>
localnet = 172.31.40.21 / 255.255.240.0; AWS NETWORK<br>
localnet = 192.168.0.0 / 255.255.0.0; RFC 1918
addresses<br>
localnet = 10.0.0.0 / 255.0.0.0; Also RFC1918<br>
localnet = 172.16.0.0 / 12; Another RFC1918 with CIDR
notation<br>
localnet = 169.254.0.0 / 255.255.0.0; Zero conf local
network<br>
localnet = 200.0.0.0 / 24<br>
localnet = 191.0.0.0 / 24<br>
localnet = 201.0.0.0 / 24<br>
localnet = 177.0.0.0 / 24<br>
<br>
localnet = 179.0.0.0 / 24<br>
<br>
<br>
Thanks<br>
<br>
Roberto.<br>
<br>
<br>
-- <br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" rel="noreferrer" target="_blank">http://www.api-digital.com</a>
--<br>
<br>
Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/" rel="noreferrer" target="_blank">https://community.asterisk.org/</a><br>
<br>
New to Asterisk? Start here:<br>
<a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" rel="noreferrer" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Getting+Started</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" rel="noreferrer" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a></blockquote>
</div>
<br>
<fieldset></fieldset>
</blockquote>
<p><br>
</p>
</div>
-- <br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" rel="noreferrer" target="_blank">http://www.api-digital.com</a> --<br>
<br>
Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/" rel="noreferrer" target="_blank">https://community.asterisk.org/</a><br>
<br>
New to Asterisk? Start here:<br>
<a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" rel="noreferrer" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Getting+Started</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" rel="noreferrer" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a></blockquote></div></div>