Re: [asterisk-users] ring group failure with "ExtensionState: 4"

2013-04-04 Thread C Goodwin 
J (the original poster, my Linux/networking/hardware guru) tried a Polycom
on this server earlier today - exact same result on inbound calls to the
Ring Groups, "Extension State 4 (UNKNOWN)", call goes to Unavailable
voicemail. Direct-dialing the extensions still works normally.

So, the issue looks to be with this new 3.0.0 release, not the Digium
phones themselves.

Can't anyone offer some feedback? We *were* excited by the integration and
feature set of the new Digium phone - this customer bought 76 of them. With
the failure of this new release (or an update package associated with it?),
the excitement quickly fades...

Any thoughts are appreciated!


C
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Re: [asterisk-users] fax - sound/tone - dealing with SPAM

2013-04-04 Thread Doug Lytle 

Joseph wrote:
Can you please share more details, is it done via dial plan? 


We use extension 2000.


You'll need to added the 'check_blacklist' sub-routine to the inbound 
parts of your dial plan.




[tele_torture]

exten => 2000,1,GotoIf($["${CALLERID(number)}" = "0"]?7:2)
exten => 2000,n,Gosub(todays_date,s,1)
exten => 2000,n,Gosub(tele_query,s,1)
exten => 2000,n,NoOP(${torture.calls})
exten => 2000,n,GosubIf($["${torture.calls}" = "1"]?tele_insert,s,1)
exten => 2000,n,GosubIf($["${torture.calls}" = "2"]?tele_blacklist,s,1)


;*Death to telemarketers
exten => 2000,n,Answer()
exten => 2000,n,Set(CDR(userfield)=Torture)
exten => 2000,n,SetMusicOnHold(conference)
exten => 2000,n,WaitMusicOnHold(30)
exten => 2000,n,Wait(1.5)
exten => 2000,n,Playback(pls-hold-while-try)
exten => 2000,n,Ringing()
exten => 2000,n,Wait(25)
exten => 2000,n,Goto,10

[tele_query]

exten => s,1,Set(torture.calls=0)
exten => s,n,MYSQL(Connect connid localhost username 'password' torture)
exten => s,n,GosubIf($["${MYSQL_STATUS}" = "-1"]?mysql_failed,s,6)
exten => s,n,MYSQL(Query resultid ${connid} SELECT phone \, calls \, name FROM 
Facility WHERE phone = "${CALLERID(number)}")
exten => s,n,MYSQL(Fetch fetchid ${resultid} torture.phone torture.calls 
torture.name)
exten => s,n,MYSQL(Disconnect ${connid})
exten => s,n,MYSQL(Clear ${resultid})
exten => s,n,Set(torture.calls=${IF($[ ${torture.calls} = 1]?2:1)})
exten => s,n,Return()

[tele_insert]

exten => s,1,MYSQL(Connect connid 192.168.104.122 username 'password' torture)
exten => s,n,GosubIf($["${MYSQL_STATUS}" = "-1"]?mysql_failed,s,6)
exten => s,n,MYSQL(Query resultid ${connid} INSERT INTO Facility set 
phone="${CALLERID(number)}" \, calls="${torture.calls}" \, 
name="${CALLERID(number)}" \, time_date="${TODAY}")
exten => s,n,MYSQL(Disconnect ${connid})

exten => s,n,System(/usr/local/bin/scripts/add_watchlist_email.sh 
opera...@mydomain.com ${CALLERID(number)} ${TODAY})
exten => s,n,Return()

[tele_blacklist]

exten => s,1,MYSQL(Connect connid 192.168.104.122 username 'password' 
blacklisted)
exten => s,n,GosubIf($["${MYSQL_STATUS}" = "-1"]?mysql_failed,s,6)
exten => s,n,MYSQL(Query resultid ${connid} INSERT INTO Facility set 
phone="${CALLERID(number)}" \, flag="YES" \, note="Blacklisted by Tele-Torture 
- ${TODAY}")
exten => s,n,MYSQL(Disconnect ${connid})

exten => s,n,System(/usr/local/bin/scripts/add_blacklist_email.sh 
opera...@mydomain.com ${CALLERID(number)} ${TODAY})
exten => s,n,Return()

[check_blacklist]

exten => s,1,GotoIf($["${CALLERID(num)}" = "" ]?2:3)
exten => s,n,Set(CALLERID(all)=Restricted <0>)
exten => s,n,MYSQL(Connect connid localhost username 'password' blacklisted)
exten => s,n,GosubIf($["${MYSQL_STATUS}" = "-1"]?mysql_failed,s,6)
exten => s,n,MYSQL(Query resultid ${connid} SELECT flag \, note FROM Facility 
WHERE phone = ${CALLERID(num)})
exten => s,n,MYSQL(Fetch fetchid ${resultid} results note)
exten => s,n,MYSQL(Disconnect ${connid})
exten => s,n,MYSQL(Clear ${resultid})
exten => s,n,Set(BLACKLISTED=${results})
exten => s,n,GotoIf($["${BLACKLISTED}" = "YES"]?blacklisted,s,1)
exten => s,n,NoOP(Caller not blacklisted)
exten => s,n,Return

[blacklisted]

exten => s,1,NoOP(Caller: ${CALLERID(num)} is on the black list)
exten => s,n,NoOP(NOTE: ${note})
exten => s,n,Set(CDR(userfield)=Blacklisted)
exten => s,n,Zapateller(answer)
exten => s,n,Hangup(2)
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Re: [asterisk-users] fax - sound/tone - dealing with SPAM

2013-04-04 Thread Joseph 

On 04/05/13 00:58, Patrick Lists wrote:

On 04/04/2013 09:54 PM, Joseph wrote:

+1.7044972383


If that number is his actual number, maybe create a script that calls
him 10 times an hour, every hour between 00:00 - 07:00am and plays
screaming monkeys every time he picks up (or his voicemail kicks in).

Regards,
Patrick


I'm not sure fighting evil with evil will do any good; but he has a 
1-800-number on his web-page so he has to pay for the call.
Utilizing asterisk to call his number 7/24 might teach him a lesson :-/

http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out+deliver+message
Though, I'm not sure this would be the correct way to go.

--
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Re: [asterisk-users] fax - sound/tone - dealing with SPAM

2013-04-04 Thread Patrick Lists 

On 04/04/2013 09:54 PM, Joseph wrote:

+1.7044972383


If that number is his actual number, maybe create a script that calls 
him 10 times an hour, every hour between 00:00 - 07:00am and plays 
screaming monkeys every time he picks up (or his voicemail kicks in).


Regards,
Patrick




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Re: [asterisk-users] fax - sound/tone - dealing with SPAM

2013-04-04 Thread Steve Edwards 

On Thu, 4 Apr 2013, Joseph wrote:


I receive several calls from this scamer:...

Does asterisk have a "fax" sound tone? If I block their number and play 
"fax" tone/sound maybe they will remove me from their calling list.


The fax CNG may work. How about SIT?

http://en.wikipedia.org/wiki/Special_information_tones

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] fax - sound/tone - dealing with SPAM

2013-04-04 Thread Doug Lytle 
>> Can you please share more details, is it done via dial plan?

I'll post a portion of my dial plan when I get home from work.  It's currently 
for Asterisk 1.4 using the mysql command, I'm in the process of moving it to 
Asterisk 11 and func_odbc.

Doug

-- 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."

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Re: [asterisk-users] fax - sound/tone - dealing with SPAM

2013-04-04 Thread Joseph 

On 04/04/13 16:21, Doug Lytle wrote:

It is an automated call and they keep rotating their caller ID so it is harder 
to block them.


Automate it.

We have an extension that the operators forward calls to that add the number to 
the black list database.

Doug


Can you please share more details, is it done via dial plan?

--
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Re: [asterisk-users] fax - sound/tone - dealing with SPAM

2013-04-04 Thread Doug Lytle 
>> It is an automated call and they keep rotating their caller ID so it is 
>> harder to block them.

Automate it.

We have an extension that the operators forward calls to that add the number to 
the black list database.  

Doug

-- 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."

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Re: [asterisk-users] fax - sound/tone - dealing with SPAM

2013-04-04 Thread Joseph 

On 04/05/13 01:29, Mitul Limbani wrote:

  Why dont u run a reverse dialer on the admin contacts phone number.
  Leave him clueless as well.

  Mitul


Reverse dialer on 7044972383 ? 
What is going to do?


--
Joseph




  On Apr 5, 2013 1:25 AM, "Joseph" <[1]syscon...@gmail.com> wrote:

I receive several calls from this scamer: Senior SafeAlert
It is an automated call and they keep rotating their caller ID so it
is harder to block them.
Does asterisk have a "fax" sound tone? If I block their number and
play "fax" tone/sound maybe they will remove me from their calling
list.
I've tried to call them but nothing helps.
Any better ideas? They keep calling sometimes few times a day.
Their web site is [2]http://www.seniorsafealert.com/
Registered through: GoDaddy.com, LLC (Domain Names | The World's
Largest Domain Name Registrar - Go Daddy)
Domain Name: [3]SENIORSAFEALERT.COM
Created on: 28-Aug-12
Expires on: 28-Aug-13
Last Updated on: 28-Aug-12
Registrant:
Sam Alure
8832 Thornbury Ln
Huntersville, North Carolina 28078
United States
Administrative Contact:
Alure, Sam [4]seniorsafeal...@gmail.com
8832 Thornbury Ln
Huntersville, North Carolina 28078
United States
[5]+1.7044972383



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Re: [asterisk-users] fax - sound/tone - dealing with SPAM

2013-04-04 Thread Mitul Limbani 
Why dont u run a reverse dialer on the admin contacts phone number. Leave
him clueless as well.

Mitul
On Apr 5, 2013 1:25 AM, "Joseph"  wrote:

> I receive several calls from this scamer: Senior SafeAlert
> It is an automated call and they keep rotating their caller ID so it is
> harder to block them.
>
> Does asterisk have a "fax" sound tone? If I block their number and play
> "fax" tone/sound maybe they will remove me from their calling list.
> I've tried to call them but nothing helps.
> Any better ideas? They keep calling sometimes few times a day.
>
>
> Their web site is 
> http://www.seniorsafealert.**com/
>
> Registered through: GoDaddy.com, LLC (Domain Names | The World's Largest
> Domain Name Registrar - Go Daddy)
> Domain Name: SENIORSAFEALERT.COM
> Created on: 28-Aug-12
> Expires on: 28-Aug-13
> Last Updated on: 28-Aug-12
>
> Registrant:
> Sam Alure
> 8832 Thornbury Ln
> Huntersville, North Carolina 28078
> United States
>
> Administrative Contact:
> Alure, Sam seniorsafeal...@gmail.com
> 8832 Thornbury Ln
> Huntersville, North Carolina 28078
> United States
> +1.7044972383
>
> --
> Joseph
>
> --
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>   
> http://lists.digium.com/**mailman/listinfo/asterisk-**users
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[asterisk-users] fax - sound/tone - dealing with SPAM

2013-04-04 Thread Joseph 

I receive several calls from this scamer: Senior SafeAlert
It is an automated call and they keep rotating their caller ID so it is harder 
to block them.

Does asterisk have a "fax" sound tone? If I block their number and play "fax" 
tone/sound maybe they will remove me from their calling list.
I've tried to call them but nothing helps. 


Any better ideas? They keep calling sometimes few times a day.


Their web site is http://www.seniorsafealert.com/

Registered through: GoDaddy.com, LLC (Domain Names | The World's Largest Domain 
Name Registrar - Go Daddy)
Domain Name: SENIORSAFEALERT.COM
Created on: 28-Aug-12
Expires on: 28-Aug-13
Last Updated on: 28-Aug-12

Registrant:
Sam Alure
8832 Thornbury Ln
Huntersville, North Carolina 28078
United States

Administrative Contact:
Alure, Sam seniorsafeal...@gmail.com
8832 Thornbury Ln
Huntersville, North Carolina 28078
United States
+1.7044972383

--
Joseph

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Re: [asterisk-users] Asterisk SIP deadlocks - update_provisional_keepalive

2013-04-04 Thread Duane Larson 
Thanks Jim.  Searched through the change log for "deadlock" but nothing
really stuck out.  I'll upgrade to 11.3 and see if that makes a difference.


On Thu, Apr 4, 2013 at 10:59 AM, Jim Lucas  wrote:

> On 04/03/2013 08:15 PM, Duane Larson wrote:
>
>> So it just happened again on both machines at the same time and I was
>> running debug on both servers.  I am running OpenSIPS and load balancing
>> between both servers so I am guessing when the invite was sent to the
>> first
>> server it was frozen for some reason and then OpenSIPS sent the invite to
>> the second server and that server was also frozen/deadlocked because of
>> the
>> SIP message.  I noticed on both servers the last log that was posted with
>> Asterisk deadlocked was the following
>>
>>
>> Asterisk version 11.0.1
>> [Apr  3 21:39:42] DEBUG[12984] res_timing_timerfd.c: Expected to
>> acknowledge 1 ticks but got 11805 instead
>>
>> Asterisk version 11.2.1
>> [Apr  3 21:39:50] DEBUG[1854] res_timing_timerfd.c: Expected to
>> acknowledge
>> 1 ticks but got 12423 instead
>>
>>
>> In my last email I posted the debug from the Asterisk server with 11.0.1
>> version of code.  Here is a post of the debug for the Asterisk server with
>> version 11.2.1
>>
>> http://pastebin.com/mbjSSAWM
>>
>>
>> This has to be a bug right?  I am thinking of opening an issue on the
>> Asterisk JIRA system
>>
>>
> A number of deadlocks were fixed in the current release of 11.3.  Please
> read the change log to see if any fit your issue.
>
> http://downloads.asterisk.org/**pub/telephony/asterisk/**
> ChangeLog-11-current
>
>
>
>>
>> On Wed, Apr 3, 2013 at 4:45 PM, Duane Larson 
>> wrote:
>>
>>  It just happened again on the 11.0.1 box and I was able to grab a debug.
>>>   I am hoping someone can tell me if this is a bug or something wrong
>>> with
>>> my config.
>>>
>>> gdb asterisk-bin/sbin/asterisk 29048
>>>
>>> Go here for the debug output
>>> http://pastebin.com/DGXx0BSk
>>>
>>>
>>> On Tue, Apr 2, 2013 at 7:42 PM, Duane Larson >> >wrote:
>>>
>>>  I am currently running two different versions of Asterisk

 11.0.1
 11.2.1

 I have noticed the bug occur on both servers.

 The issue is that when I try to dial a phone number sometimes the call
 will never go out.  I will check the Asterisk server with NGREP and see
 that the SIP messages are making it to Asterisk but Asterisk isn't
 responding.

 I do the following command "netstat -nap |grep 5060" and see that
 Asterisk has a lot under the "Recv-Q" column.

 It usually takes about 10 minutes before Asterisk becomes responsive
 again or else before 10 minutes is up I could restart Asterisk and
 everything will be back to normal.

 I see in the message logs the following errors

 On the 11.0.1 Asterisk server
 WARNING[23723][C-0010] chan_sip.c: Unable to cancel schedule ID
 11473.  This is probably a bug (chan_sip.c:
 update_provisional_keepalive,
 line 4406).

 On the 11.2.1 Asterisk server
 WARNING[3493][C-001f] chan_sip.c: Unable to cancel schedule ID
 30810.
   This is probably a bug (chan_sip.c: update_provisional_keepalive, line
 4683).


 When I look in chan_sip.c on both servers I see that they are the same
 line of code

 AST_SCHED_DEL_UNREF(sched, pvt->provisional_keepalive_**sched_id,
 dialog_unref(pvt, "when you delete the provisional_keepalive_sched_**id,
 you
 should dec the refcount for the stored dialog ptr"));



 What could be causing this because it seems to happen at least once a
 day.


>>>
>>>
>>> --
>>> --
>>> *--*--*--*--*--*
>>> Duane
>>> *--*--*--*--*--*
>>> --
>>>
>>>
>>
>>
>>
>>
>> --
>> __**__**_
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>> 
>> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>>
>>
>
> --
> Jim Lucas
>
> http://www.cmsws.com/
> http://www.cmsws.com/examples/
>



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Duane
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Re: [asterisk-users] Asterisk SIP deadlocks - update_provisional_keepalive

2013-04-04 Thread Jim Lucas 

On 04/03/2013 08:15 PM, Duane Larson wrote:

So it just happened again on both machines at the same time and I was
running debug on both servers.  I am running OpenSIPS and load balancing
between both servers so I am guessing when the invite was sent to the first
server it was frozen for some reason and then OpenSIPS sent the invite to
the second server and that server was also frozen/deadlocked because of the
SIP message.  I noticed on both servers the last log that was posted with
Asterisk deadlocked was the following


Asterisk version 11.0.1
[Apr  3 21:39:42] DEBUG[12984] res_timing_timerfd.c: Expected to
acknowledge 1 ticks but got 11805 instead

Asterisk version 11.2.1
[Apr  3 21:39:50] DEBUG[1854] res_timing_timerfd.c: Expected to acknowledge
1 ticks but got 12423 instead


In my last email I posted the debug from the Asterisk server with 11.0.1
version of code.  Here is a post of the debug for the Asterisk server with
version 11.2.1

http://pastebin.com/mbjSSAWM


This has to be a bug right?  I am thinking of opening an issue on the
Asterisk JIRA system



A number of deadlocks were fixed in the current release of 11.3.  Please 
read the change log to see if any fit your issue.


http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11-current





On Wed, Apr 3, 2013 at 4:45 PM, Duane Larson  wrote:


It just happened again on the 11.0.1 box and I was able to grab a debug.
  I am hoping someone can tell me if this is a bug or something wrong with
my config.

gdb asterisk-bin/sbin/asterisk 29048

Go here for the debug output
http://pastebin.com/DGXx0BSk


On Tue, Apr 2, 2013 at 7:42 PM, Duane Larson wrote:


I am currently running two different versions of Asterisk

11.0.1
11.2.1

I have noticed the bug occur on both servers.

The issue is that when I try to dial a phone number sometimes the call
will never go out.  I will check the Asterisk server with NGREP and see
that the SIP messages are making it to Asterisk but Asterisk isn't
responding.

I do the following command "netstat -nap |grep 5060" and see that
Asterisk has a lot under the "Recv-Q" column.

It usually takes about 10 minutes before Asterisk becomes responsive
again or else before 10 minutes is up I could restart Asterisk and
everything will be back to normal.

I see in the message logs the following errors

On the 11.0.1 Asterisk server
WARNING[23723][C-0010] chan_sip.c: Unable to cancel schedule ID
11473.  This is probably a bug (chan_sip.c: update_provisional_keepalive,
line 4406).

On the 11.2.1 Asterisk server
WARNING[3493][C-001f] chan_sip.c: Unable to cancel schedule ID 30810.
  This is probably a bug (chan_sip.c: update_provisional_keepalive, line
4683).


When I look in chan_sip.c on both servers I see that they are the same
line of code

AST_SCHED_DEL_UNREF(sched, pvt->provisional_keepalive_sched_id,
dialog_unref(pvt, "when you delete the provisional_keepalive_sched_id, you
should dec the refcount for the stored dialog ptr"));



What could be causing this because it seems to happen at least once a day.





--
--
*--*--*--*--*--*
Duane
*--*--*--*--*--*
--







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--
Jim Lucas

http://www.cmsws.com/
http://www.cmsws.com/examples/

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Re: [asterisk-users] Fundemental changes to CDR within single asterisk family

2013-04-04 Thread Ishfaq Malik 
On Tue, 2013-03-26 at 07:26 -0500, Matthew Jordan wrote:
> On 03/26/2013 05:22 AM, Ishfaq Malik wrote:
> > Hi
> > 
> > In asterisk 1.8.7.0, an inbound call that was transferred to another
> > peer would have 2 cdr entries.
> > 
> > In asterisk 1.8.18.0 this same activity has a single cdr entry.
> > 
> > This is a rather large and fundamental change to be enacting halfway
> > through a single family branch, was there any reason why this happened?
> > It means we can't upgrade without doing significant extra development
> > and testing.
> > 
> 
> This was most likely an unintended consequence of some other change
> (most likely dealing with masquerades). Is 1.8.18.0 the exact version
> when the behaviour changed?
> 
> Just so I'm clear on the scenario, what are the channel technologies
> involved? Is the transfer initiated via a protocol message or via a DTMF
> feature?
> 
> Thanks,
> 
> Matt
> 

Hi Matt

Did you ever spot/recreate the change I was referring to?

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Ishfaq Malik 
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Company: Packnet Limited
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