<div dir="ltr">I had this recently... and i bet if you use wireshark/tcpdump youll see a dns lookup for the server's own hostname right before the cutout, and audio again after response is received. quick fix is to add the hosts name and ip to /etc/hosts<br><br><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26280">https://issues.asterisk.org/jira/browse/ASTERISK-26280</a><br></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Aug 23, 2016 at 12:20 PM, Brent Davidson <span dir="ltr"><<a href="mailto:brent@texascountrytitle.com" target="_blank">brent@texascountrytitle.com</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">
I'm having an issue with some Snom 300s on a server running Asterisk
version 13.9.1, Dahdi 2.11.1 w/OSLEC and pjsip 2.5.1. There is <u><b>NO
NAT</b></u> involved. Phones and server are plugged into the
same network switch, all on the same IP range. The server is
running a Wildcard AEX410 analog card with 2 FXO modules receiving
incoming analog lines.<br>
<br>
Occasionally, in the middle of a call, the audio will drop out for
between 15 and 20 seconds before suddenly coming back. I've tried
running u-Law as the codec and licensed g.729 version 13.0_3.1.7
with exactly the same results. I have tried turning on every
logging option I can think of to troubleshoot this but have not been
able to find a solution. I'm troubleshooting by remote, so haven't
been able to run a wireshark capture yet.<br>
<br>
pings to the phones from the Asterisk server show no packet loss
during the cut-outs.<br>
<br>
Any ideas?<br>
<br>
Thanks,<br>
<div>
<b>Brent Davidson</b> <br>
<div><br>
<br>
</div>
</div>
</div>
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