Thanks for the input. Long ago the CDR showed "asterisk" as the CLID but it doesn't anymore so I am puzzled now how to even stop taking calls because my CLID is now blank and I can't refuse any call with no CLID.<div>
<br></div><div><b><i>WARNING[11002] chan_dahdi.c: CallerID returned with error on channel 'DAHDI/2-1'</i></b></div><div><br></div><div>Here are some out of place messages I am getting in my logs but nothing out of norm around the time I get Ghost calls though:</div>
<div><div><b><i>WARNING[11002] chan_dahdi.c: CallerID returned with error on channel 'DAHDI/2-1'</i></b></div><div><br></div><div><b><i>NOTICE[12524] chan_dahdi.c: Got event 17 (Polarity Reversal)...</i></b></div>
<div><b><i><br></i></b></div><div><b><div><i>DEBUG[12524] chan_dahdi.c: Ignore switch to REVERSED Polarity on channel 4, state 6</i></div><div><span class="Apple-style-span" style="font-weight: normal; "><b><div style="display: inline !important; ">
<i><br></i></div></b></span></div><div><span class="Apple-style-span" style="font-weight: normal; "><b><div style="display: inline !important; "><i>DEBUG[12524] chan_dahdi.c: Ignoring Polarity switch to IDLE on channel 4, state 6</i></div>
</b></span></div></b></div></div><div><br></div><div><br></div><div>Can someone shed light on these options as to what exactly they do:</div><div><span class="Apple-style-span" style="color: rgb(51, 51, 51); font-family: 'Lucida Grande', 'Lucida Sans Unicode', Helvetica, Verdana, Arial, Tahoma, sans-serif; font-size: 12px; line-height: 17px; ">hanguponpolarityswitch=yes<br>
answeronpolarityswitch=yes</span></div><div><font class="Apple-style-span" color="#333333" face="'Lucida Grande', 'Lucida Sans Unicode', Helvetica, Verdana, Arial, Tahoma, sans-serif"><span class="Apple-style-span" style="font-size: 12px; line-height: 17px;"><br>
</span></font></div><div>Hopefully some Asterisk guru can tell us more about what might be happening as I see this as a situation that can be avoided or at least there should be a workaround for this.</div><div><br></div>
<div>Regards,</div><div><br><div class="gmail_quote"><br></div><div class="gmail_quote"><br></div><div class="gmail_quote">On Mon, May 9, 2011 at 9:50 AM, Brian Henning <span dir="ltr"><<a href="mailto:bhenning@pineinst.com">bhenning@pineinst.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Hello Bruce,</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">I did not find a solution, only advice to lead me to think “huh, well that’s annoying but we can deal with it.” I understand from my users, though, that it’s <b>not</b> always the case that it’s a phantom call—sometimes there really is someone calling.</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Note that I haven’t tried what I’m about to suggest, but you might try examining the CALLERID data before dialing the SIP extensions and, if it is empty or contains “asterisk,” reset it to something like “not available.” </span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Cheers,</span></p><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">~Brian</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p><div style="border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in"><p class="MsoNormal"><b><span style="font-size:10.0pt">From:</span></b><span style="font-size:10.0pt"> Bruce B [mailto:<a href="mailto:bruceb444@gmail.com" target="_blank">bruceb444@gmail.com</a>] <br>
<b>Sent:</b> Friday, May 06, 2011 10:55 PM</span></p><div class="im"><br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br></div><b>Cc:</b> <a href="mailto:bhenning@pineinst.com" target="_blank">bhenning@pineinst.com</a><div class="im">
<br><b>Subject:</b> Re: [asterisk-users] Occasional call from "asterisk"</div><p></p></div><p class="MsoNormal"> </p><p class="MsoNormal">Hi Brian,</p><div><div></div><div class="h5"><div><p class="MsoNormal"> </p>
</div><div><p class="MsoNormal">Did you find a solution to your problem? or at least got a working dial-plan for it? I have the same problem again as well and want to know what to do with the dial-plan to off-set the effect at least since Telco says it's not their issue.</p>
</div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal">Regards,</p></div><div><p class="MsoNormal" style="margin-bottom:12.0pt">Bruce</p><div><p class="MsoNormal">On Thu, Apr 7, 2011 at 5:53 PM, Brian Henning <<a href="mailto:bhenning@pineinst.com" target="_blank">bhenning@pineinst.com</a>> wrote:</p>
<p class="MsoNormal">Hi,<br><br>Now and then our SIP phones ring with "asterisk" showing as the caller-ID.<br>Upon picking up the receiver, there is about five seconds of silence and<br>then the channel is closed (hangup). Can anyone offer some insight? Here's<br>
relevant snippets from my extensions.conf and Master.csv log:<br><br>This line shows up in Master.csv:<br><br>"","","1-NOANSWER","inbound","","DAHDI/1-1","SIP/505-00000150","Dial","SIP/5<br>
01&SIP/502&SIP/503&SIP/504&SIP/505&SIP/506,10,tTgr","2011-04-07<br>21:37:05","2011-04-07 21:37:16","2011-04-07<br>21:37:21",16,5,"ANSWERED","DOCUMENTATION","1302212225.444",""<br>
<br>Here's [inbound] from extensions.conf:<br>[inbound]<br>exten => s,1,Answer<br>exten => s,n,Ringing<br>exten => s,n,Set(CALLERID(num),9${CALLERID(num)})<br>exten => s,n,Dial(SIP/504&SIP/506,5,tTgr)<br>
exten => s,n,Goto(1-${DIALSTATUS},1)<br>exten => 1-ANSWER,1,Hangup<br>exten =><br>_1-.,1,Dial(SIP/501&SIP/502&SIP/503&SIP/504&SIP/505&SIP/506,10,tTgr)<br>exten => _1-.,n,Goto(2-${DIALSTATUS},1)<br>
exten => 2-ANSWER,1,Hangup<br>exten => _2-.,1,Voicemail(499@default,u)<br>exten => _2-.,2,Hangup<br><br>The idea is that first 504 and 506 ring, then if neither of them answer,<br>everyone rings. Works great most of the time.<br>
<br>I have a hunch that maybe this happens if the inbound caller hangs up while<br>the first Dial() is ringing, but I would've expected to see the first Dial<br>(to 504 and 506) show up in the Master.csv log, and it's not there. (The<br>
preceding line of the log is a call from almost an hour earlier). In that<br>case though I'd expect to see "1-CANCEL" in the log instead. Perhaps if the<br>caller happens to hang up right between the two Dial() commands?..<br>
<br>As an aside, the Set(CALLERID...) bit doesn't work. The idea was to prepend<br>a 9 so that a SIP user could use the "redial" feature of the phone's call<br>log to return a missed call (automatically including the 9 for outside<br>
line). Unfortunately the 9 does not get prepended.<br><br>Thanks in advance for any and all advice!<br>~Brian<br><br>------------------------------------------------------<br> Brian Henning, Software Engineer<br>
<br> /\ Pine Research Instrumentation<br> //\\ 5908 Triangle Drive<br> ///\\\ Raleigh, NC 27617<br> ////\\\\ USA<br> ||<br> || phone: <a href="tel:919.782.8320" target="_blank">919.782.8320</a><br> fax: <a href="tel:919.782.8323" target="_blank">919.782.8323</a><br>
email: <a href="mailto:bhenning@pineinst.com" target="_blank">bhenning@pineinst.com</a><br>------------------------------------------------------<br><br><br><br>--<br>_____________________________________________________________________<br>
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</div><p class="MsoNormal"> </p></div></div></div></div></div></blockquote></div><br></div>