<div dir="ltr">About a call not being hang up for asterisk while the client hang up, please remember SIP is based on UDP and UDP packets get easily lost... they are retransmitted but sometime they are lost as the previous...<div>
<br></div><div>For the ghost calls, are the SIP port of the phones reachable from the Internet... maybe it is just someone trying to place some free calls</div><div><br></div><div>Leandro</div></div><div class="gmail_extra">
<br><br><div class="gmail_quote">2014-02-12 19:05 GMT+01:00 Mike Diehl <span dir="ltr"><<a href="mailto:mdiehlenator@gmail.com" target="_blank">mdiehlenator@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr"><div><div><div>Hi all,<br><br></div>I've got a customer who's reporting "ghost calls." Essentially, the phone rings, they pick up, and there's no body there.<br><br></div>It is NOT one-way audio, and it doesn't happen all the time.<br>
<br></div>We use voipmonitor to watch calls, and this is what we saw for the call in question:<br><br><span style="font-family:courier new,monospace">| calldate | caller | called | duration | whohanged |<br>
+---------------------+------------+----------------+----------------+-----+<br>| 2014-02-12 09:28:06 | 575xxxxxxx | CCD539F38...-1 | 60 | NULL |<br>| 2014-02-12 09:29:06 | 575xxxxxxx | CCD539F38...-2 | 1 | NULL |<br>
</span><br><div>So, it looks like my customer received a call, which lasted a minute, and then they hung up. Then their phone rang again, but there was no one there.<br></div><div>Based on what I'm seeing in my log, the first call was never hung up, even though both parties claim to have hung up the call. My logs only indicate that the 'h' extension was called once, at 9:29:07<br>
<br></div><div>My question is, how can a call not get hung up when both parties hang up the call? I know that sounds odd, but that's what I'm seeing.<br><br>Any ideas?<span class="HOEnZb"><font color="#888888"><br>
<br>Mike.<br></font></span></div><div><br></div></div>
<br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</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" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br></div>