[asterisk-users] Occasional call from "asterisk"

Steve Totaro stotaro at totarotechnologies.com
Sat May 7 00:27:44 CDT 2011


Telco always says it is not their issue.

This is all over google, did you even check?  Did you check your options in
chan_dahdi.conf?

hanguponpolarityswitch=yes

I am not sure if that is your problem but it would be helpful to list the
things you have found, tested, and ruled out.

As for prepending a 9 for redial, I would say doing it in the [outbound]
dial context would be best practice.

For my installations, I have eliminated the need to "Dial 9 for an outside
line"  That goes back to the key systems where 9 got you an outside line.

I have also eliminated the need to dial 1 as well.  A good dialplan makes
these legacy, I still leave them there to avoid confusion.

For some clients that use TDM and VoIP, I may make 8 + number go over VoIP
and 9 + whatever go over TDM.

Default without the 8 or 9 is to go out over TDM or whatever the customer
wants, or TDM if they seem lost.

I don't give them too many decisions to make, just educate them on the
options programmed into the system.

Last thing, your dialplan looks too over engineered.  How about this and
fixing your callerID syntax?

[inbound]
exten => s,1,Answer
exten => s,n,Set(CALLERID(num),9${CALLERID(num)})
exten => s,n,Dial(SIP/504&SIP/506,5,tTgr)
exten => s,n,Dial(SIP/501&SIP/502&SIP/503&SIP/504&SIP/505&SIP/506,10,tTgr)
exten => s,n,Voicemail(499 at default,u)
exten => s,n,Hangup

Thanks,
Steve Totaro


On Fri, May 6, 2011 at 10:54 PM, Bruce B <bruceb444 at gmail.com> wrote:

> 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
>>
>
>
> --
> _____________________________________________________________________
> -- 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110507/77dc474e/attachment.htm>


More information about the asterisk-users mailing list