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Brian Henning

2011-Apr-07 21:53 UTC

[asterisk-users] Occasional call from "asterisk"

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 
------------------------------------------------------

Cary Fitch

2011-Apr-07 23:16 UTC

head link

[asterisk-users] Occasional call from "asterisk"

We were getting "a lot" of those. We installed IPTables with blocking
of
everything outside of North America and they all but vanished.

No direct evidence, but a pretty good empirical guess that they were related
to hackers trying to get paths to the US.

CF

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Brian Henning
Sent: Thursday, April 07, 2011 4:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Occasional call from "asterisk"

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 
------------------------------------------------------ 



--
_____________________________________________________________________
-- 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
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   http://lists.digium.com/mailman/listinfo/asterisk-users

Warren Selby

2011-Apr-07 23:16 UTC

head link

[asterisk-users] Occasional call from "asterisk"

On Thu, Apr 7, 2011 at 4:53 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:
>
<snip>

I've seen this on cases where a "phantom" call comes in on a DAHDI
channel -
these calls were the results of faulty wiring on the part of the telco.
Check your logs for any errors on your DAHDI channels around the time of the
"ghost" calls.

It could also be a case of someone calls in and then hangs up before the
call is actually passed to asterisk, and the telco is just slow to hangup
the call.

> 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.
>
Your Set() syntax is wrong.  Try this:

exten => s,n,Set(CALLERID(num)=9${CALLERID(num)})

-- 
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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Bruce B

2011-May-07 02:54 UTC

head link

[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
> ------------------------------------------------------
>
>
>
> --
> _____________________________________________________________________
> -- 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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Bruce B

2011-May-09 15:32 UTC

head link

[asterisk-users] Occasional call from "asterisk"

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
> ------------------------------------------------------
>
>
>
> --
> _____________________________________________________________________
> -- 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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