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.
*WARNING[11002] chan_dahdi.c: CallerID returned with error on channel 'DAHDI/2-1'* 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: *WARNING[11002] chan_dahdi.c: CallerID returned with error on channel 'DAHDI/2-1'* *NOTICE[12524] chan_dahdi.c: Got event 17 (Polarity Reversal)...* * * * DEBUG[12524] chan_dahdi.c: Ignore switch to REVERSED Polarity on channel 4, state 6 DEBUG[12524] chan_dahdi.c: Ignoring Polarity switch to IDLE on channel 4, state 6 * Can someone shed light on these options as to what exactly they do: hanguponpolarityswitch=yes answeronpolarityswitch=yes 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. Regards, On Mon, May 9, 2011 at 9:50 AM, Brian Henning <bhenn...@pineinst.com> wrote: > Hello Bruce, > > > > 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 *not* always the case that it’s a phantom call—sometimes > there really is someone calling. > > > > 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.” > > > > Cheers, > > ~Brian > > > > *From:* Bruce B [mailto:bruceb...@gmail.com] > *Sent:* Friday, May 06, 2011 10:55 PM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Cc:* bhenn...@pineinst.com > > *Subject:* Re: [asterisk-users] Occasional call from "asterisk" > > > > Hi Brian, > > > > 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. > > > > Regards, > > Bruce > > On Thu, Apr 7, 2011 at 5:53 PM, Brian Henning <bhenn...@pineinst.com> > wrote: > > Hi, > > Now and then our SIP phones ring with "asterisk" showing as the caller-ID. > Upon picking up the receiver, there is about five seconds of silence and > then the channel is closed (hangup). Can anyone offer some insight? > Here's > relevant snippets from my extensions.conf and Master.csv log: > > This line shows up in Master.csv: > > > "","","1-NOANSWER","inbound","","DAHDI/1-1","SIP/505-00000150","Dial","SIP/5 > 01&SIP/502&SIP/503&SIP/504&SIP/505&SIP/506,10,tTgr","2011-04-07 > 21:37:05","2011-04-07 21:37:16","2011-04-07 > 21:37:21",16,5,"ANSWERED","DOCUMENTATION","1302212225.444","" > > Here's [inbound] from extensions.conf: > [inbound] > exten => s,1,Answer > exten => s,n,Ringing > exten => s,n,Set(CALLERID(num),9${CALLERID(num)}) > exten => s,n,Dial(SIP/504&SIP/506,5,tTgr) > exten => s,n,Goto(1-${DIALSTATUS},1) > exten => 1-ANSWER,1,Hangup > exten => > _1-.,1,Dial(SIP/501&SIP/502&SIP/503&SIP/504&SIP/505&SIP/506,10,tTgr) > exten => _1-.,n,Goto(2-${DIALSTATUS},1) > exten => 2-ANSWER,1,Hangup > exten => _2-.,1,Voicemail(499@default,u) > exten => _2-.,2,Hangup > > The idea is that first 504 and 506 ring, then if neither of them answer, > everyone rings. Works great most of the time. > > I have a hunch that maybe this happens if the inbound caller hangs up while > the first Dial() is ringing, but I would've expected to see the first Dial > (to 504 and 506) show up in the Master.csv log, and it's not there. (The > preceding line of the log is a call from almost an hour earlier). In that > case though I'd expect to see "1-CANCEL" in the log instead. Perhaps if > the > caller happens to hang up right between the two Dial() commands?.. > > As an aside, the Set(CALLERID...) bit doesn't work. The idea was to > prepend > a 9 so that a SIP user could use the "redial" feature of the phone's call > log to return a missed call (automatically including the 9 for outside > line). Unfortunately the 9 does not get prepended. > > Thanks in advance for any and all advice! > ~Brian > > ------------------------------------------------------ > Brian Henning, Software Engineer > > /\ Pine Research Instrumentation > //\\ 5908 Triangle Drive > ///\\\ Raleigh, NC 27617 > ////\\\\ USA > || > || phone: 919.782.8320 > fax: 919.782.8323 > email: bhenn...@pineinst.com > ------------------------------------------------------ > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >
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