[asterisk-users] Occasional call from "asterisk"

Mon May 9 10:32:13 CDT 2011

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 <bhenning at 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:bruceb444 at gmail.com]
> *Sent:* Friday, May 06, 2011 10:55 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Cc:* bhenning at 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 <bhenning at 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 at 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: bhenning at pineinst.com
> ------------------------------------------------------
>
>
>
> --
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