Re: [asterisk-users] Stress Testing IVR

[Rajkumar S]

On Wed, Feb 18, 2009 at 3:51 AM, David Backeberg  wrote:
> As for actually putting delays and pressing the right buttons, you're
> on your own. You would need to write a custom AGI script specific to
> your IVR, and call it from your call file, which you then put in a
> bash loop. In that case, DTMF is your friend.

Thanks for the tip, I will work in this direction and post any results
to the list.

raj

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Re: [asterisk-users] What is the purpose of membermacro in queues.conf

[Rajkumar S]

On Tue, 17 Feb 2009, Mark Michelson wrote:

> The purpose of exposing these values is to allow for an administrator to 
> use these for any purpose he may desire.

An example would be really great :)

I am confused because these values are exported just before the call is 
connected and I am wondering how can I intervene at this point and do some 
thing?

> Finally, you asked about membermacro. This allows for a macro to execute 
> on a queue member's channel when he answers the call. This is very 
> similar to the 'M' option for the dial application.

Thanks, Does this support some thing similar to MACRO_RESULT ?

with regards,

raj

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[asterisk-users] Distributed presence in 1.6

[Rajkumar S]

Hi,

Russell's blog[1] is down and there are not much information about
this any where else. Any one with more information about res_ais and
how it is used?

raj

[1] 
http://www.russellbryant.net/blog/index.php/2008/06/10/asterisk-16-now-with-distributed-presence/

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Re: [asterisk-users] Credit Card processing machines

[bilal ghayyad]

Really once I read credit card, I got to become interested to know whatis 
exactly happenning.

I am looking to have the possibility to pay to the bank using the VoIP adaptor 
or IP Telephony, by entering the credit card digits and the password and the 
amound.

I do not know if u can help me in this point, and if I am far from your subject 
or not.

Is there a bank or credit card processor provider that can help to acheive such 
kind of service? (credit card/visa electron payment through VoIP gateway or 
Telephone lines)?

Regards
Bilal



> The ADT alarm going thru VoIP will create a life safety
> issue.  Hope you planned for that..
> --Don
> 
> 
> 
> On 2/17/09 6:31 AM, "Jeff LaCoursiere"
>  wrote:
> 
> 
> 
> 
> On Tue, 17 Feb 2009, Andrew Joakimsen wrote:
> 
> > On Fri, Feb 6, 2009 at 17:11, Jeff LaCoursiere
>  wrote:
> >>
> >> Anyone have much luck with these on ATA's?  I
> have a few sites that use
> >> them succesfully with multi-port Audiocodes boxes,
> but just connected ten
> >> machines to Linksys 2102s and they are very flaky.
>  Using u-law on a 100Mb
> >> switched network that is barely utilized, then out
> a T1 on a Sangoma card.
> >>
> >> Perhaps there is some tuning on the Linksys or the
> credit card machine
> >> itself?  Going to look into reducing the baud rate
> on the machines, but
> >> sadly the bank has them password protected and
> wants to charge a
> >> "reprogramming fee" :(
> >
> > They make credit card terminals with Ethernet -- use
> that instead.
> >
> 
> The client's processor charges 7c/transaction over IP
> (plus normal
> charges), so they are quite keen to keep it working the way
> it was before
> I replaced their PBX ;)
> 
> As a followup, *99 prepended on any Linksys ATA does indeed
> make a
> difference in modem reliability.  Both their CCs and their
> ADT alarm
> devices now function reliably.  I also reduced the CC baud
> rate to 300
> baud (!), and it is rock solid now!
> 
> j



  

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Re: [asterisk-users] Network architecture

[michel freiha]

Dear Alex,

Thanks for the reply..Can you please list some of these solutions that you
talked about on your reply?
Even I would like to ask if you had a bad experience with asterisk regarding
simultaneous calls limitation and If I'll send 1k calls to an asterisk
machine with the appropriate hardware what will happen?
Kindly note that no trans coding is done, just pass thru codec

Regards

On Tue, Feb 17, 2009 at 5:34 PM, Alex Balashov wrote:

> No, asterisk on conventional hardware can handle at most a few hundred
> calls.
>
> I would strongly discourage the use of Asterisk purely as a transit
> element for billing. Just because a2billing is available does not mean
> you should. Far more scalable solutions are easily available.
>
> --
> Sent from mobile device
>
> On Feb 17, 2009, at 10:19 AM, michel freiha  wrote:
>
> > Hi all,
> >
> > I'm planning to build a VOIP solution for handling SIP calls coming
> > from endpoints registered on a specific SIP proxy...I made some
> > research regarding network architecture and found out that the best
> > solution is to use OpenSips as SIP proxy for registration and local
> > calls between registered endpoints and use asterisk server with
> > a2billing for PSTN calls, rating, routing and all other stuff plus a
> > MySQL database...
> >
> > This architecture convinced me, but I have some questions regarding
> > asterisk and I need asterisk expert answers in order to take
> > decision...
> >
> > 1- Is there any Software limitation on asterisk regarding number of
> > simulltaneous calls?
> > 2- Can 1 asterisk server handle 5000 simuitaneous calls if I have
> > the appropriate hardware?
> > 3- It's etter to have one asterisk server for hadling 5k
> > simultaneous calls or divide the load on different servers?
> >
> >
> > Waiting your reply
> >
> > Regards
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> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Network architecture

[michel freiha]

Dear Helm,

Kindly confirm why you do not recommend the VMs solution and if you had bad
experience for it and what did you get?

Regards

On Tue, Feb 17, 2009 at 9:24 PM, Wilton Helm  wrote:

> >You may be able to split up some of the servers into multiple VMs -- maybe
> five >servers with five VMs each.
>
> I'm not sure I see the merit in this.  VMs seem to be regarded as a magic
> bullet (i.e. free lunch).  I don't know of any case where 5 VMs can
> accomplish more work on one processor than simply letting the processor
> manage it all (except if the OS and or application can't efficiently split
> the task into the necessary multiple threads, which I don't think is an
> issue here).  By definition, the total accomplished must be less with VMs,
> because the hypervisor will take some CPU cycles.
>
> Wilton
>
>
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[asterisk-users] Setting SIP header on agent calls made by a queue

[Lenz Emilitri]

Hello list,

I am trying to set a custom SIP header on all calls that are made by the app
queue because I want to track a certain state at the SIP level.

If I use the following code:

exten => s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID})
exten => s,n,Queue(myQueue)

this works fine for the FIRST call made from the queue to an agent; but if
that call does not go through, it's not repeated on subsequent calls.

I know I could use Local channels as members of the queue, but I was
wondering if ther was something more general and that worked whatever your
channel configuration might be.

I also tried exporting the variable by setting the URL field of the queue()
call, but it is not shown in the resulting SIP dialog.

 Any suggestion but patching the source?

l.




-- 
Loway - home of QueueMetrics - http://queuemetrics.com
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Re: [asterisk-users] Credit Card processing machines

[Tim Panton]

On 17 Feb 2009, at 19:20, David Gibbons wrote:



We will be testing the ADT connection heavily this week.  The modem
connections to my understanding are 2400 baud.  Over G.711U and a T1 I
don't see why this wouldn't be as solid as a POTS line, but our  
tests will

tell!


We do *fax* in this way and it works like a charm. We can hit much  
more than 2400 baud I think too.


--Dave



Our creditcard company's small print _insists_ on a direct analog  
exchange line

with no other devices in between.

Tim.

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk





smime.p7s
Description: S/MIME cryptographic signature
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Re: [asterisk-users] Credit Card processing machines

[Tzafrir Cohen]

On Wed, Feb 18, 2009 at 10:02:28AM +, Tim Panton wrote:

> Our creditcard company's small print _insists_ on a direct analog  
> exchange line with no other devices in between.

Wow. You have a direct copper wire to their credit card processing
system?  :-)

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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[asterisk-users] AGI pdf book

[michel freiha]

Dear Sir,

Can someone help me please to find a free ebook talking about AGI scripting
through asterisk?

Regards
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Re: [asterisk-users] Asterisk supports SIP-T?

[Daviramos Roussenq Fortunato]

How to convert SIP-T to SIP for Asterisk?

2009/2/17 Raj Jain 

> On Tue, Feb 17, 2009 at 1:06 PM, Daviramos Roussenq Fortunato
>  wrote:
> > Asterisk supports SIP-T?
>
> Nope. Here is some old discussion on this topic:
> http://lists.digium.com/pipermail/asterisk-biz/2008-May/026690.html
>
> --
> Raj Jain
>
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Re: [asterisk-users] Asterisk supports SIP-T?

[Raj Jain]

On Wed, Feb 18, 2009 at 6:55 AM, Daviramos Roussenq Fortunato
 wrote:
> How to convert SIP-T to SIP for Asterisk?

You'll need to strip out ISUP MIME body in your SIP messaging with Asterisk.

--
Raj Jain

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Re: [asterisk-users] AGI pdf book

[David fire]

http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.com/books/9780596510480.pdf
asterisk->support books section Asterisk: The Future of
Telephony
is greate.
David

2009/2/18 michel freiha 

> Dear Sir,
>
> Can someone help me please to find a free ebook talking about AGI scripting
> through asterisk?
>
> Regards
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
(\__/)
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(")_(")signature to help him gain world domination.
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Re: [asterisk-users] ztdummy compile under 2.6.28 ?

[Gordon Henderson]

On Tue, 17 Feb 2009, Gordon Henderson wrote:

> It looks like something has changed in the HPET kernel code in 2.6.28
> (maybe .27 too) that's stopped ztdummy.c compiling (in 1.2 and 1.4
> versions of zapata) A kernel structure member has been renamed with some
> crypic comments in the lkml about it.

OK, a few checks in the dahdi code (look, I spelt it right! :) and the 
changes are minimal and would apply to both the 1.2 and 1.4 versions of 
ztdummy. What's the recomended way to publish patches - if anyone wants 
them?

Thanks,

Gordon


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Re: [asterisk-users] AGI pdf book

[michel freiha]

Dear Sir,

the asterisk book from oreilly does not make full description to the AGI
scripting...I suggest please if someone advice to me a free PDF book just
dedicated for AGI and nothing else

Regards

On Wed, Feb 18, 2009 at 2:09 PM, David fire  wrote:

>
> http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.com/books/9780596510480.pdf
> asterisk->support books section Asterisk: The Future of 
> Telephony
> is greate.
> David
>
> 2009/2/18 michel freiha 
>
>> Dear Sir,
>>
>> Can someone help me please to find a free ebook talking about AGI
>> scripting through asterisk?
>>
>> Regards
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> (\__/)
> (='.'=)This is Bunny. Copy and paste bunny into your
> (")_(")signature to help him gain world domination.
>
>
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[asterisk-users] Detecting which party initiates a hangup

[Darren Murphy]

Hi,

I would like to know if it is possible to detect which party initiates a
hangup - and if so, how this is done.

In my asterisk log, I see something like the following:

Feb 18 04:14:13 VERBOSE[17488] logger.c: -- Executing
Hangup("IAX2/ToHK1-16", "") in new stack
Feb 18 04:14:13 VERBOSE[17488] logger.c: -- Hungup 'IAX2/ToHK1-16'

This tells me when the call was terminated, but doesn't tell me which party
actually hung up first.

Is this possible to detect?

thanks,
Darren
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[asterisk-users] Ditech API

[Dean Collins]

http://www.speechtechmag.com/Articles/News/News-Feature/Ditech-to-Delive
r-Voice-Based-Web-Interaction-during-Mobile-Calls--52606.aspx

 

thought this Ditech API might interest a few people.

 

Regards,

Dean Collins
Cognation Inc
d...@cognation.net
 +1-212-203-4357   New York
+61-2-9016-5642   (Sydney in-dial).
+44-20-3129-6001 (London in-dial).

 

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[asterisk-users] Accumulated call time

[Geoff Lane]

Hi All,

Asterisk 1.4.12 CentOS 5

My ISP account includes nearly 500 minutes of VOIP calls per month but
the service is expensive for unbundled minutes. So I'm trying to find
a way to keep an accumulated total of calls made through that trunk so
that I can automatically switch to a lower-cost provider when my
bundled minutes are used. The plan is to store the accumulated time in
AstDB and reset this with a cron job at the beginning of each period.

I understand that the Dial() application sets two variables -
DIALEDTIME and ANSWEREDTIME - to the total time the Dial() application
ran and the time since the call was answered respectively. However, I
can't find a way to access these. I've tried the following:

exten => s,1,Dial(${rgMain}/${EXTEN},${RINGTIME},t)
exten => s,n,Log(NOTICE, Call to ${EXTEN} lasted ${DIALEDTIME})

However the expected notice does not appear in
/var/log/asterisk/messages, which is where other notices generated
with the Log() application do.

Can someone point the way?

TIA,

-- 
Geoff


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Re: [asterisk-users] Network architecture

[Benny Amorsen]

"Wilton Helm"  writes:

> I'm not sure I see the merit in this.  VMs seem to be regarded as a magic
> bullet (i.e. free lunch).  I don't know of any case where 5 VMs can
> accomplish more work on one processor than simply letting the processor
> manage it all

Modern machines have more than one processor, and Asterisk doesn't
scale linearly with the number of processors. Multiple Asterisk
instances aren't a magic bullet, but they can help sometimes.


/Benny


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Re: [asterisk-users] Credit Card processing machines

[Jeff LaCoursiere]

You shoudl start with your bank.  They can probably provide the equipment.

j


On Wed, 18 Feb 2009, bilal ghayyad wrote:

> Really once I read credit card, I got to become interested to know whatis 
> exactly happenning.
>
> I am looking to have the possibility to pay to the bank using the VoIP 
> adaptor or IP Telephony, by entering the credit card digits and the password 
> and the amound.
>
> I do not know if u can help me in this point, and if I am far from your 
> subject or not.
>
> Is there a bank or credit card processor provider that can help to acheive 
> such kind of service? (credit card/visa electron payment through VoIP gateway 
> or Telephone lines)?
>
> Regards
> Bilal
>
> 
>
>> The ADT alarm going thru VoIP will create a life safety
>> issue.  Hope you planned for that..
>> --Don
>>
>>
>>
>> On 2/17/09 6:31 AM, "Jeff LaCoursiere"
>>  wrote:
>>
>>
>>
>>
>> On Tue, 17 Feb 2009, Andrew Joakimsen wrote:
>>
>>> On Fri, Feb 6, 2009 at 17:11, Jeff LaCoursiere
>>  wrote:

 Anyone have much luck with these on ATA's?  I
>> have a few sites that use
 them succesfully with multi-port Audiocodes boxes,
>> but just connected ten
 machines to Linksys 2102s and they are very flaky.
>>  Using u-law on a 100Mb
 switched network that is barely utilized, then out
>> a T1 on a Sangoma card.

 Perhaps there is some tuning on the Linksys or the
>> credit card machine
 itself?  Going to look into reducing the baud rate
>> on the machines, but
 sadly the bank has them password protected and
>> wants to charge a
 "reprogramming fee" :(
>>>
>>> They make credit card terminals with Ethernet -- use
>> that instead.
>>>
>>
>> The client's processor charges 7c/transaction over IP
>> (plus normal
>> charges), so they are quite keen to keep it working the way
>> it was before
>> I replaced their PBX ;)
>>
>> As a followup, *99 prepended on any Linksys ATA does indeed
>> make a
>> difference in modem reliability.  Both their CCs and their
>> ADT alarm
>> devices now function reliably.  I also reduced the CC baud
>> rate to 300
>> baud (!), and it is rock solid now!
>>
>> j
>
>
>
>
>

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[asterisk-users] US DID

[Nhadie]

Hi,

Anyone knows a DID provider that can do both outbound and inbound?

Regards
Nhadie

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Re: [asterisk-users] Network architecture

[Steve Totaro]

Check out FreeSwitch to replace Asterisk in your core.

On Wed, Feb 18, 2009 at 3:42 AM, michel freiha  wrote:

> Dear Alex,
>
> Thanks for the reply..Can you please list some of these solutions that you
> talked about on your reply?
> Even I would like to ask if you had a bad experience with asterisk
> regarding simultaneous calls limitation and If I'll send 1k calls to an
> asterisk machine with the appropriate hardware what will happen?
> Kindly note that no trans coding is done, just pass thru codec
>
> Regards
>
>
> On Tue, Feb 17, 2009 at 5:34 PM, Alex Balashov 
> wrote:
>
>> No, asterisk on conventional hardware can handle at most a few hundred
>> calls.
>>
>> I would strongly discourage the use of Asterisk purely as a transit
>> element for billing. Just because a2billing is available does not mean
>> you should. Far more scalable solutions are easily available.
>>
>> --
>> Sent from mobile device
>>
>> On Feb 17, 2009, at 10:19 AM, michel freiha  wrote:
>>
>> > Hi all,
>> >
>> > I'm planning to build a VOIP solution for handling SIP calls coming
>> > from endpoints registered on a specific SIP proxy...I made some
>> > research regarding network architecture and found out that the best
>> > solution is to use OpenSips as SIP proxy for registration and local
>> > calls between registered endpoints and use asterisk server with
>> > a2billing for PSTN calls, rating, routing and all other stuff plus a
>> > MySQL database...
>> >
>> > This architecture convinced me, but I have some questions regarding
>> > asterisk and I need asterisk expert answers in order to take
>> > decision...
>> >
>> > 1- Is there any Software limitation on asterisk regarding number of
>> > simulltaneous calls?
>> > 2- Can 1 asterisk server handle 5000 simuitaneous calls if I have
>> > the appropriate hardware?
>> > 3- It's etter to have one asterisk server for hadling 5k
>> > simultaneous calls or divide the load on different servers?
>> >
>> >
>> > Waiting your reply
>> >
>> > Regards
>> > ___
>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> >
>> > asterisk-users mailing list
>> > To UNSUBSCRIBE or update options visit:
>> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> ___
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] asterisk-users Digest, Vol 55, Issue 52

[bilal ghayyad]

But clients can call to your IVR and do the payment from their home or the 
mobile, correct?

If that is possible, how they pay $50.53? I mean, how they enter the 0.53$? 
They use the * to express the (.)?

>From the other side, they asking for analog telephone line or they need a 
>leased line between your site and their machine?

About the IVR, are u using Asterisk?
Regards
Bilal

> --
> 
> Message: 17
> Date: Wed, 18 Feb 2009 12:23:41 +0200
> From: Tzafrir Cohen 
> Subject: Re: [asterisk-users] Credit Card processing
> machines
> To: asterisk-users@lists.digium.com
> Message-ID: <20090218102341.gd21...@xorcom.com>
> Content-Type: text/plain; charset=us-ascii
> 
> On Wed, Feb 18, 2009 at 10:02:28AM +, Tim Panton wrote:
> 
> > Our creditcard company's small print _insists_ on
> a direct analog  
> > exchange line with no other devices in between.
> 
> Wow. You have a direct copper wire to their credit card
> processing
> system?  :-)
> 
> -- 
>Tzafrir Cohen
> icq#16849755  jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir



  

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Re: [asterisk-users] Accumulated call time

[David fire]

use the h exten.
when someone hangup dial go to exten h.
or  put the option in the dial command to go to the next priority on hangup
but there is a problem if during the call they transfer it to other exten
you dont have the next priority.
David

2009/2/18 Geoff Lane 

> Hi All,
>
> Asterisk 1.4.12 CentOS 5
>
> My ISP account includes nearly 500 minutes of VOIP calls per month but
> the service is expensive for unbundled minutes. So I'm trying to find
> a way to keep an accumulated total of calls made through that trunk so
> that I can automatically switch to a lower-cost provider when my
> bundled minutes are used. The plan is to store the accumulated time in
> AstDB and reset this with a cron job at the beginning of each period.
>
> I understand that the Dial() application sets two variables -
> DIALEDTIME and ANSWEREDTIME - to the total time the Dial() application
> ran and the time since the call was answered respectively. However, I
> can't find a way to access these. I've tried the following:
>
> exten => s,1,Dial(${rgMain}/${EXTEN},${RINGTIME},t)
> exten => s,n,Log(NOTICE, Call to ${EXTEN} lasted ${DIALEDTIME})
>
> However the expected notice does not appear in
> /var/log/asterisk/messages, which is where other notices generated
> with the Log() application do.
>
> Can someone point the way?
>
> TIA,
>
> --
> Geoff
>
>
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Re: [asterisk-users] AGI pdf book

[Danny Nicholas]

Try www.cpan.org   --> modules --> agi.  This will
help you from a PERL perspective.  The AGI is also applicable for (at least)
PHP and C+.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, February 18, 2009 6:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AGI pdf book

 

Dear Sir,

the asterisk book from oreilly does not make full description to the AGI
scripting...I suggest please if someone advice to me a free PDF book just
dedicated for AGI and nothing else

Regards

On Wed, Feb 18, 2009 at 2:09 PM, David fire  wrote:

http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.com/b
ooks/9780596510480.pdf
asterisk->support books section Asterisk: The Future of Telephony
 
is greate.
David

2009/2/18 michel freiha 

Dear Sir,

Can someone help me please to find a free ebook talking about AGI scripting
through asterisk?

Regards

 

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(")_(")signature to help him gain world domination. 


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Re: [asterisk-users] Setting SIP header on agent calls made by a queue

[Danny Nicholas]

Put this snippet in a macro and call the macro.   That way the data lives
for the duration of the incoming call.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Wednesday, February 18, 2009 3:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Setting SIP header on agent calls made by a queue

 

Hello list,

I am trying to set a custom SIP header on all calls that are made by the app
queue because I want to track a certain state at the SIP level. 


If I use the following code:

exten => s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID})
exten => s,n,Queue(myQueue)

this works fine for the FIRST call made from the queue to an agent; but if
that call does not go through, it's not repeated on subsequent calls. 

I know I could use Local channels as members of the queue, but I was
wondering if ther was something more general and that worked whatever your
channel configuration might be. 

I also tried exporting the variable by setting the URL field of the queue()
call, but it is not shown in the resulting SIP dialog.

 Any suggestion but patching the source?

l.

 


-- 
Loway - home of QueueMetrics - http://queuemetrics.com

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Re: [asterisk-users] only ring phones that are not on a call

[Benny Amorsen]

"Danny Nicholas"  writes:

> You could set up a hint for the extension and check the hint for "inuse"
> before executing the Dial in your dialplan
>
> Exten => 801,hint,SIP/100
>
> Exten => XXX,1,System("/usr/sbin/asterisk -rx "core show hints"|/bin/grep
> SIP/100|/bin/grep InUse

I think we have a winner for "most creative use of System()".

If the calls go through a queue, a much easier solution is to simply
set ringinuse=so. Job done.

If the calls are done with Dial(), the best way is to use groups. See
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+group


/Benny


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[asterisk-users] Need help on Forwarding

[Max Alex]

Hi All,
I am using asterisk 1.4.19,
I have setup the dialplans to get the incoming call and that will be sent to
another context by local channel,
In another context i have setup the ring group, that portion is working
fine.
I have noticed that when i have set one of the extension in call forwarding
in phone (linksys)
then it says to me 302 Moved Temporarily and call is forwarded to that
number.
In this i need to disable the forwarding from dialplan or any configuration
method, so when the ring group is in process then no call will be forwarded.
Please provide help regarding this!!
Thanks in Advance!!
Thanks,
Max Alex
Voip Developer
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Re: [asterisk-users] Setting SIP header on agent calls made by a queue

[Benny Amorsen]

Lenz Emilitri  writes:

> If I use the following code:
>
> exten => s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID})
> exten => s,n,Queue(myQueue)
>
> this works fine for the FIRST call made from the queue to an agent; but if
> that call does not go through, it's not repeated on subsequent calls. 

That sounds like a bug to me.


/Benny


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Re: [asterisk-users] US DID

[Paul Chambers]

Nhadie wrote:
> Hi,
> 
> Anyone knows a DID provider that can do both outbound and inbound?
> 
> Regards
> Nhadie
> 

I've been happy with FlowRoute.

I have a number port pending with them that's taking forever (requested 
in Oct), but I doubt it's their fault. In every other respect, I've been 
happy with their service.

I should note I'm a low-volume customer, residential and home office use.

Paul

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[asterisk-users] directrtpsetup=yes does not work in 1.4.23.1

[Arturo Díaz Almagro]

Hello,

I have a working system based on asterisk 1.4.23.1 and I want RTP going
end-to-end but not using canreinvite because it creates problems in my
configuration. I have tested directrtpsetup=yes and canreinvite=no but media
goes through asterisk. I know this feature is experimental, but so
experimental that does not work anyway? Could somebody help me?

Regards

-- 
Arturo Díaz
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Re: [asterisk-users] Credit Card processing machines

[Edwin Quijada]

> 
> Our creditcard company's small print _insists_ on a direct analog 
> exchange line
> with no other devices in between.
> 
> Tim.
> 
> Tim Panton - Web/VoIP consultant and implementor
> www.westhawk.co.uk
> 


You can do it an interface using AGI to comunicate with equipment or verifone.  
I did it once 
*---* 
*-Edwin Quijada 
*-Developer DataBase 
*-JQ Microsistemas 
*-809-849-8087
* " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo 
comun" 
*---*




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Re: [asterisk-users] Setting SIP header on agent calls made by a queue

[Lenz Emilitri]

I think this is by design - each time the Dial() is performed, SIP headers
are reset.
l.
2009/2/18 Benny Amorsen 
>

> Lenz Emilitri  writes:
>
> > If I use the following code:
> >
> > exten => s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID})
> > exten => s,n,Queue(myQueue)
> >
> > this works fine for the FIRST call made from the queue to an agent; but
> if
> > that call does not go through, it's not repeated on subsequent calls.Â
>
> That sounds like a bug to me.
>
>
> /Benny
>
>


-- 
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Re: [asterisk-users] Setting SIP header on agent calls made by a queue

[Lenz Emilitri]

Interestiong - how would you do this? I thought macros on the queue command
were only for 1.6.

l.


2009/2/18 Danny Nicholas 

>  Put this snippet in a macro and call the macro.   That way the data lives
> for the duration of the incoming call.
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri
> *Sent:* Wednesday, February 18, 2009 3:05 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Setting SIP header on agent calls made by a
> queue
>
>
>
> Hello list,
>
> I am trying to set a custom SIP header on all calls that are made by the
> app queue because I want to track a certain state at the SIP level.
>
> If I use the following code:
>
> exten => s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID})
> exten => s,n,Queue(myQueue)
>
> this works fine for the FIRST call made from the queue to an agent; but if
> that call does not go through, it's not repeated on subsequent calls.
>
> I know I could use Local channels as members of the queue, but I was
> wondering if ther was something more general and that worked whatever your
> channel configuration might be.
>
> I also tried exporting the variable by setting the URL field of the queue()
> call, but it is not shown in the resulting SIP dialog.
>
>  Any suggestion but patching the source?
>
> l.
>
>
>
>
> --
> Loway - home of QueueMetrics - http://queuemetrics.com
>
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>
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> To UNSUBSCRIBE or update options visit:
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>



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Re: [asterisk-users] US DID

[Fred Posner]

On Feb 18, 2009, at 9:57 AM, Paul Chambers wrote:

> Nhadie wrote:
>> Hi,
>>
>> Anyone knows a DID provider that can do both outbound and inbound?
>>
>> Regards
>> Nhadie
>>
>
> I've been happy with FlowRoute.
>
> I have a number port pending with them that's taking forever  
> (requested
> in Oct), but I doubt it's their fault. In every other respect, I've  
> been
> happy with their service.
>
> I should note I'm a low-volume customer, residential and home office  
> use.
>
> Paul
>

Also very happy with flowroute.


Fred Posner
f...@teamforrest.com

Main:   +1 (212) 937-7844
Direct: +1 (503) 914-0999

www.teamforrest.com








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Re: [asterisk-users] asterisk-users Digest, Vol 55, Issue 52

[Bill Michaelson]

> > What "economic downturn"?
> >
> > I'm sick and tired of hearing this mantra.


I wish you the best of luck in maintaining your immunity.



>
> Same here (in the UK).
>
> As long as people need to make phone calls ...
>
> Gordon



   The economy (and indeed humanity as a whole) needs periods of
   removing sludge,
   deadwood, and general stupidity.

Profound. In a way, I agree.

   Having said that I would think that many of the list contributors
   are based on
   the USA, so no surprises they would feel that way.

Because the US is economically isolated, I suppose.

   I thought our media left a bit to be desired until Youtube came
   along and I
   could see the propaganda and mindnumbing dross trotted out in the
   corporate
   controlled media there.

My mind is being numbed.

   So people should decide for themselves whether to think positive or
   negative
   thoughts.

And they should whistle a happy tune, too!

   I for one intend to take full advantage of the opportunities
   presented by this period of transition in human affairs.

Good to see the entrepreneurial spirit alive in this, uh, period of 
transition in human affairs. John Todd would like to hear how you are 
doing this.









smime.p7s
Description: S/MIME Cryptographic Signature
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[asterisk-users] connection to siemens hipath

[Jerry Geis]

I am connecting 1.4.22 and dahdi 2.1.0.3+2.1.0.2 to a siemens hipath 300 
and siemens hipath 4000. (2 channels to each switch)
with a TE210p card setup as T1 with em_w.

When the call is initiated to either switch the phone rings, when its 
answered then nothing...
I hear no audio etc... After the timeout period the call is hung up.
The phone switch 300 needs the T1 reset as the channel is not freed up.

Anyone ever ran into this? What might be going on? Its installed on a 
new dell poweredge 860.

Incoming calls I never see the "starting simple switch"...

--
loadzone=us
defaultzone=us
span=1,1,6,esf,b8zs
e&m=1-2
span=2,1,6,esf,b8zs
e&m=25-26
--- the card is on its own interrupt.
   CPU0   CPU1  
  0:   92977641  0IO-APIC-edge  timer
  1:  9  0IO-APIC-edge  i8042
  8:  1  0IO-APIC-edge  rtc
  9:  0  0   IO-APIC-level  acpi
 12:133  0IO-APIC-edge  i8042
 14: 39  0IO-APIC-edge  ide0
 50:   76292403996   IO-APIC-level  uhci_hcd:usb1, 
ehci_hcd:usb4, libata
 66:   9326   92941584   IO-APIC-level  wct2xxp
169:2941012573   IO-APIC-level  eth0
177:  0  0   IO-APIC-level  uhci_hcd:usb2
185:  0  0   IO-APIC-level  uhci_hcd:usb3
NMI:   3066   4650
LOC:   91316250   91316178
ERR:  0
MIS:  0


- chan_dahdi.c
[channels]


switchtype=national
signalling=em_w
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
context=smvoice-incoming
group=1
channel => 1-2


switchtype=national
signalling=em_w
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
context=smvoice-incoming
group=1
channel => 25-26

yes they are just using channels 1,2 on both T1's

Any thoughts?

Jerry

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Re: [asterisk-users] Accumulated call time

[Geoff Lane]

On Wednesday, February 18, 2009, David fire wrote:

> use the h exten. when someone hangup dial go to exten h. or  put the
> option in the dial command to go to the next priority on hangup but
> there is a problem if during the call they transfer it to other
> exten you dont have the next priority.

Thanks. FWIW, I was wondering why my system kept giving warnings:
  chan_sip.c: No such host: h
Until your post, I didn't know that special extension existed.

I guess I need to play around with that and also the (just discovered)
"g" option for the Dial() application.

Thanks again,

-- 
Geoff


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Re: [asterisk-users] Setting SIP header on agent calls made by aqueue

[Danny Nicholas]

Lets say your dialplan looks like this:

exten => s,1,Answer()

exten => s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID})
exten => s,n,Queue(myQueue)

exten => s,n,blah

exten => s,n,Hangup()

 

you would make a macro like this

 

[macro-siphead]

exten => s,n,SIPAddHeader(X-Unique-ID: ${ARG1})
exten => s,n,Queue(myQueue)

 

and change your dialplan to do this

exten => s,1,Answer()

exten => s,n(redial),Noop()

exten => s,n,Macro(siphead, ${UNIQUEID})

exten => s,n,GOTOIF(blah),?redial
exten => s,n,blah

exten => s,n,Hangup()

 

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Wednesday, February 18, 2009 9:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Setting SIP header on agent calls made by
aqueue

 

Interestiong - how would you do this? I thought macros on the queue command
were only for 1.6.

l.

 

2009/2/18 Danny Nicholas 

Put this snippet in a macro and call the macro.   That way the data lives
for the duration of the incoming call.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Wednesday, February 18, 2009 3:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Setting SIP header on agent calls made by a queue

 

Hello list,

I am trying to set a custom SIP header on all calls that are made by the app
queue because I want to track a certain state at the SIP level. 


If I use the following code:

exten => s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID})
exten => s,n,Queue(myQueue)

this works fine for the FIRST call made from the queue to an agent; but if
that call does not go through, it's not repeated on subsequent calls. 

I know I could use Local channels as members of the queue, but I was
wondering if ther was something more general and that worked whatever your
channel configuration might be. 

I also tried exporting the variable by setting the URL field of the queue()
call, but it is not shown in the resulting SIP dialog.

 Any suggestion but patching the source?

l.

 


-- 
Loway - home of QueueMetrics - http://queuemetrics.com


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-- 
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Re: [asterisk-users] AGI pdf book

[Jared Smith]

On Wed, 2009-02-18 at 14:36 +0200, michel freiha wrote:
> the asterisk book from oreilly does not make full description to the
> AGI scripting...

You're right.. the O'Reilly book doesn't make a full and complete
description of AGI programming.  (It was better than anything else
written at the time, but it's nowhere near perfect.)

If you have specific suggestions on what more you'd like to see covered
in the AGI chapter, I'm certainly open to feedback.

> I suggest please if someone advice to me a free PDF book just
> dedicated for AGI and nothing else

The only book I'm aware of that covers AGI and only AGI is the AGI book
written by Nir Simionovich.  It's not free, but I hear that it's the
best book in the world on the subject of AGI programming, and I'm
looking forward to reading it myself.  More info at
http://www.packtpub.com/asterisk-gateway-interface-programming/book



-- 
Jared Smith
Digium, Inc. | Training Manager 




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Re: [asterisk-users] Network architecture

[Jared Smith]

On Tue, 2009-02-17 at 12:24 -0700, Wilton Helm wrote:
>  I'm not sure I see the merit in this.  VMs seem to be regarded as a
> magic bullet (i.e. free lunch).  I don't know of any case where 5 VMs
> can accomplish more work on one processor than simply letting the
> processor manage it all (except if the OS and or application can't
> efficiently split the task into the necessary multiple threads, which
> I don't think is an issue here).  By definition, the total
> accomplished must be less with VMs, because the hypervisor will take
> some CPU cycles.

While this would appear to be the case at first glance, there's
something more subtle going on here.  In this particular case, there are
data structures inside of Asterisk that get less efficient as you put
more and more calls through the system.  Let's take a linked list of
channels for example... when you have ten calls on the system, it's
fairly simple to walk down the list of channels and find the channel
you're looking for.  When you have a thousand calls on the system,
that's certainly less efficient.  

This should make it apparent why the resource requirements for Asterisk
don't scale linearly with the call volume.  Or, to put it another way,
you you can think of splitting the calls across two VMs as a crude way
of bringing some efficiency back into those structures.

Now, in the interest of full disclosure, I used the idea of a
linked-list above only as an example.  Many of the changes in Asterisk
between 1.4 and 1.6.0 have been to re-plumb a lot of the internal
structures to behave better under higher call volumes (things like
replacing linked lists with hash tables, etc.).  The Asterisk developers
are continuing to work on the efficiency of the code data structures
within Asterisk, but in the meantime, I hope I've given you some insight
into why splitting calls across virtual machines on the same box can
offer improvements, despite the overhead of running a hypervisor.


-- 
Jared Smith
Digium, Inc. | Training Manager 




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Re: [asterisk-users] Open Source in an Economic Downturn: Asterisk stories needed

[John Novack]

Michael wrote:

On Wed, 18 Feb 2009 13:37:57 John Todd wrote:
  

I'm giving a talk at SCALE 2009 (Southern CAlifornia Linux Expo) on
Sunday in Los Angeles, and the topic of my talk is "Open Source in an
Economic Downturn".  I've got lots of talking points for this talk,
but it would be interesting to hear some short anecdotes about how you
in the Asterisk community are thriving, or at least surviving, by
virtue of the benefits of Open Source.  I find that real-world
examples are worth more than all of the bullet points in the world,
and timely stories from the community would be more interesting than
hearing me prattle on.



What "economic downturn"?

I'm sick and tired of hearing this mantra.

Michael
  
Perhaps there isn't a downturn in a country with more sheep than people, 
but MOST of the rest of the world is headed downhill rapidly.
This could easily turn into a decade long period, with few having any 
real answers.
If people are out of a job, they can't buy much of anything, and 
telephone and computer services certainly go by the wayside long before 
food and shelter


John Novack

--
Dog is my co-pilot

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Re: [asterisk-users] Credit Card processing machines

[bilal ghayyad]

And is there a bank accept to give such kind of communication?

The user was able to dial his card number and the amount from his phone (or IP 
Phone registered with Asterisk), and Asterisk communicate with the bank or 
company credit card provider?

How the user will enter $50.25?
What about expiration date of the credit card?

Regards
Bilal

 
> > Our creditcard company's small print _insists_ on
> a direct analog 
> > exchange line
> > with no other devices in between.
> > 
> > Tim.
> > 
> > Tim Panton - Web/VoIP consultant and implementor
> > www.westhawk.co.uk
> > 
> 
> 
> You can do it an interface using AGI to comunicate with
> equipment or verifone.  I did it once 



  

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Re: [asterisk-users] licensed g729

[Thomas Kenyon]

Michael Graves wrote:
> On Sun, 15 Feb 2009 10:23:50 +0800, Nhadie wrote:
> 
>> Hi All,
>>
>> If i buy 20 g729 and install to my asterisk, if 20 calls are already 
>> engaged using g729. would the next call then revert to using the other 
>> codec, in this case ulau and alaw?
> 
> Yes, if you set the codec preferences this way. Allow both but prefer
> G.729. And presuming that the end-points do likewise.
> 
Since when? It certainly used to be that if you'd used up all the 
licenses, next call would still negotiate g.729 and they'd be no audio 
and lots of errors in the console. (albeit this was a long time ago).

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Re: [asterisk-users] Please help test the gender detection moduleat 575-613-4392

[Asterisk Asterisk]

Thanks for the feedback. I did some research and it looks like you were calling 
over international lines. It also appears that there was high than average 
static on the line, which is not normal for my system. It's true that I threw 
my recordings together quickly and the beep was supposed to be funny - it was 
actually me saying "beep". However, the static and noise you received was 
probably not from my system.

Nonetheless, I am working on improving the results of detection and will have a 
new release today or tomorrow. I'll post it up on the test systems for people 
to test and build additional data for refinement. Most importantly, I'll be 
adding a background noise filter and fine tuning the male/female results. After 
I get the gender detection done, I'll also be adding age range detection.

Justin

--Original Message--
From: Anselm Martin Hoffmeister
To: nt_jnew...@yahoo.com
Sent: Feb 18, 2009 4:09 AM
Subject: Re: [asterisk-users] Please help test the gender detection moduleat 
575-613-4392

Am Montag, den 16.02.2009, 11:45 -0800 schrieb Asterisk Asterisk:

> Let me know how it works when you try the test number at 575-613-4392.

Hi Justin,

I tried your module half an hour ago, and first of all, the bad sound
quality immediately came to my attention. The prompts where noisy as if
I had called from a mobile (which I did not) - intercontinental lines
may be the cause here.

The pause after the voice recording is quite long, giving the impression
that you somehow missed the (short) voice recording and still wait for
the caller's input: After the beep, I spoke as told, then had to wait
five or so seconds.

Besides, I have been detected as "female", three times in a row. I
should probably go to the bathroom and check, but I think you are in
error there.

BR
Anselm


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[asterisk-users] OT: Re: Credit Card processing machines

[Richard Lyman]

bilal ghayyad wrote:
> And is there a bank accept to give such kind of communication?
>
> The user was able to dial his card number and the amount from his phone (or 
> IP Phone registered with Asterisk), and Asterisk communicate with the bank or 
> company credit card provider?
>
> How the user will enter $50.25?
> What about expiration date of the credit card?
>
> Regards
> Bilal
>   

There are already touch tone merchant services available, google 'touchpay'.

To answer your question, yes, dial a number for the processor, enter 
merchant number, enter transaction type, enter amount  (dollars and 
cents, $10 would be 1000), enter exp date (also 4 digits), wait for read 
back of auth number or denial response.

Obviously this is a subject you need to discuss with your 
merchant/processor provider.



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Re: [asterisk-users] Credit Card processing machines

[Kinjal Dixit]

Ideally the person needs to enter the credit card number, expiration date in
mmyy format (which is the format in which the expiration date is shown on
the card), and the ccv number.  The amount would probably be calculated on
the basis of the outstanding amounts, or the products selected.  Think of
trying to buy a plane ticket or pay a bill.  You are unlikely to want the
caller to enter the amount.  The thing is to structure the IVR in such a way
that the caller is informed of the amount and does not have to enter it.  If
you take a far out case of a donation help line, you can simply go "for $5
press 1, for $10 press 2, for $20 press 3".  If someone wanted to donate
$15, too bad for us.  If it turns out a lot of people want to donate $15,
you can simply adjust the IVR (and of course the other logic).

This is a simple enough task.  The big deal is supposed to be in ensuring
that the date and the ccv number DTMF do not show up in any log files or
trace files, and surely do not get logged by the application.  You can
simply turn off all DTMF logging, but you dont want to do that.  Only the
place where you accept the secure information, the logging should be
absolutely turned off.

Getting the issue?


On Wed, Feb 18, 2009 at 11:20 PM, bilal ghayyad  wrote:

> And is there a bank accept to give such kind of communication?
>
> The user was able to dial his card number and the amount from his phone (or
> IP Phone registered with Asterisk), and Asterisk communicate with the bank
> or company credit card provider?
>
> How the user will enter $50.25?
> What about expiration date of the credit card?
>
> Regards
> Bilal
>
> 
> > > Our creditcard company's small print _insists_ on
> > a direct analog
> > > exchange line
> > > with no other devices in between.
> > >
> > > Tim.
> > >
> > > Tim Panton - Web/VoIP consultant and implementor
> > > www.westhawk.co.uk
> > >
> >
> >
> > You can do it an interface using AGI to comunicate with
> > equipment or verifone.  I did it once
>
>
>
>
>
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-- 
http://www.linkedin.com/in/kinjaldixit

open networker
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[asterisk-users] trunk to trunk

[Leonja Cerebro]

Hi,
Sorry, I'm a newbee in Asterisk, and I want to call from one SIP trunk of
Asterisk B (registered in Asterisk A as extension)
to incoming call across another trunk of Asterisk B to extension of Asterisk
C
What the dial plan should be?

Thanks
-- 
We never did too much talking anyway
So don't think twice, it's all right
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Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Steve Totaro]

On Mon, Feb 16, 2009 at 2:45 PM, Asterisk Asterisk wrote:

> This module detects gender and approximate age range. I'm working on
> getting it's accuracy to 80%+ on a consistent basis, after implementing
> filters to remove background noise and other artifacts.
>
> It's designed for a number of things. To start, I have several clients
> (primarily mobile content and servers providers) that want to profile and
> generate demographics of their users for selling advertising. They also want
> to understand their user base. Plus, some customers have found that male and
> female users tend to respond differently to different prompts, flows, etc.
> This helps in designing a system that meets needs of many different types of
> users.
>
> Of course, there are many other uses and I'm sure people can generate some
> cool ideas.
>
> Let me know how it works when you try the test number at 575-613-4392.
> Also, let me know if you have any interest in the module.
>
> Justin
> nt_jnewman at yahoo.com
>
> --
> *From:* Ron Joffe 
> *To:* asterisk-users@lists.digium.com
> *Cc:* Asterisk Asterisk 
> *Sent:* Monday, February 16, 2009 11:05:24 AM
> *Subject:* Re: [asterisk-users] Please help test the gender detection
> module at 575-613-4392
>
> That's an interesting module.
>
> Care to elaborate on what you designed it for ?
>
> Thanks,
>
> Ron
>
>
>
>
> On Monday 16 February 2009 13:29, Asterisk Asterisk wrote:
> > I need your help: please help test the gender detection module at
> > 575-613-4392.
> >
> > I wrote a gender detection module and thought I'd try it out. It only
> takes
> > a second. I've been showing 90%+ accuracy and I want to make sure it's
> > working correctly. Rain and significant background noise seems to throw
> it
> > off, so I still have a bit of work to do.
> >
> > Have your friends and significant others call too. Also, let me know if
> you
> > have any need for the module.
> >
> > Justin Newman
> > nt_jnewman at yahoo.com
>

Tried to test and got "call could not be completed as dialed".

This sounds very interesting Justin.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] OT: Re: Credit Card processing machines

[Jeff LaCoursiere]

On Wed, 18 Feb 2009, Richard Lyman wrote:

> bilal ghayyad wrote:
>> And is there a bank accept to give such kind of communication?
>>
>> The user was able to dial his card number and the amount from his phone (or 
>> IP Phone registered with Asterisk), and Asterisk communicate with the bank 
>> or company credit card provider?
>>
>> How the user will enter $50.25?
>> What about expiration date of the credit card?
>>
>> Regards
>> Bilal
>>
>
> There are already touch tone merchant services available, google 'touchpay'.
>
> To answer your question, yes, dial a number for the processor, enter
> merchant number, enter transaction type, enter amount  (dollars and
> cents, $10 would be 1000), enter exp date (also 4 digits), wait for read
> back of auth number or denial response.
>
> Obviously this is a subject you need to discuss with your
> merchant/processor provider.
>

I think there is a deeper meaning in the question.  Bilal, you are missing 
a piece of software in the middle.  Yes, you can call your processor 
directly and enter your merchant code, the CC, the amount, and get an 
approval (this is what a cashier will do if their CC machine is down, for 
example).  I don't think this is what you are looking for.

Allow me to assume that you have a potential product that you would like 
an IVR to accept payment for.  You need to write an AGI script or a very 
clever dialplan that will prompt the customer for the appropriate 
information, then wrap it all up nicely and send it to your processor. 
Your AGI or dialplan script will then be able to tell the customer that 
the payment was accepted and instructions for delivering the product, or 
that it was denied and hang up on them :)

I'm sure someone has written this AGI already.  There are probably pay-for 
IVRs you could buy and integrate.  You could write it yourself, or pay 
someone here on the list to write it for you.  Its actually not that 
difficult.

In all cases, however, you will need to make an arrangement with your bakn 
to accept credit card payments and get the details for how to send 
transactions.  There are MANY processors that will accept transactions 
over the net using various protocols.  I've used IC Verify in the past, 
for instance.

Hope this helps,

j

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Re: [asterisk-users] trunk to trunk

[Robert Broyles]

Hi,

You might want to check out this tutorial:
http://hostseries.com/connecting-to-asterisk-servers-via-sip/

It's a good place to start.

--
Regards,
Robert Broyles




Leonja Cerebro wrote:

Hi,
Sorry, I'm a newbee in Asterisk, and I want to call from one SIP trunk 
of Asterisk B (registered in Asterisk A as extension)
to incoming call across another trunk of Asterisk B to extension of 
Asterisk C

What the dial plan should be?

Thanks
--
We never did too much talking anyway
So don't think twice, it's all right


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Re: [asterisk-users] OT: Re: Credit Card processing machines

[Jon Pounder]

Jeff LaCoursiere wrote:
> On Wed, 18 Feb 2009, Richard Lyman wrote:
>
>   
>> bilal ghayyad wrote:
>> 
>>> And is there a bank accept to give such kind of communication?
>>>
>>> The user was able to dial his card number and the amount from his phone (or 
>>> IP Phone registered with Asterisk), and Asterisk communicate with the bank 
>>> or company credit card provider?
>>>
>>> How the user will enter $50.25?
>>> What about expiration date of the credit card?
>>>
>>> Regards
>>> Bilal
>>>
>>>   
>> There are already touch tone merchant services available, google 'touchpay'.
>>
>> To answer your question, yes, dial a number for the processor, enter
>> merchant number, enter transaction type, enter amount  (dollars and
>> cents, $10 would be 1000), enter exp date (also 4 digits), wait for read
>> back of auth number or denial response.
>>
>> Obviously this is a subject you need to discuss with your
>> merchant/processor provider.
>>
>> 
>
> I think there is a deeper meaning in the question.  Bilal, you are missing 
> a piece of software in the middle.  Yes, you can call your processor 
> directly and enter your merchant code, the CC, the amount, and get an 
> approval (this is what a cashier will do if their CC machine is down, for 
> example).  I don't think this is what you are looking for.
>
> Allow me to assume that you have a potential product that you would like 
> an IVR to accept payment for.  You need to write an AGI script or a very 
> clever dialplan that will prompt the customer for the appropriate 
> information, then wrap it all up nicely and send it to your processor. 
> Your AGI or dialplan script will then be able to tell the customer that 
> the payment was accepted and instructions for delivering the product, or 
> that it was denied and hang up on them :)
>
> I'm sure someone has written this AGI already.  There are probably pay-for 
> IVRs you could buy and integrate.  You could write it yourself, or pay 
> someone here on the list to write it for you.  Its actually not that 
> difficult.
>
> In all cases, however, you will need to make an arrangement with your bakn 
> to accept credit card payments and get the details for how to send 
> transactions.  There are MANY processors that will accept transactions 
> over the net using various protocols.  I've used IC Verify in the past, 
> for instance.
>   

check out www.opayc.com for a generic api that works with many many 
processors, and looks like an odbc database driver to asterisk/agi etc., 
also works with gateways where you need a browser on the net as it just 
emulates that part of things.



> Hope this helps,
>
> j
>
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>   


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Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Steve Totaro]

On Wed, Feb 18, 2009 at 1:28 PM, Steve Totaro <
stot...@totarotechnologies.com> wrote:

>
>
> On Mon, Feb 16, 2009 at 2:45 PM, Asterisk Asterisk 
> wrote:
>
>> This module detects gender and approximate age range. I'm working on
>> getting it's accuracy to 80%+ on a consistent basis, after implementing
>> filters to remove background noise and other artifacts.
>>
>> It's designed for a number of things. To start, I have several clients
>> (primarily mobile content and servers providers) that want to profile and
>> generate demographics of their users for selling advertising. They also want
>> to understand their user base. Plus, some customers have found that male and
>> female users tend to respond differently to different prompts, flows, etc.
>> This helps in designing a system that meets needs of many different types of
>> users.
>>
>> Of course, there are many other uses and I'm sure people can generate some
>> cool ideas.
>>
>> Let me know how it works when you try the test number at 575-613-4392.
>> Also, let me know if you have any interest in the module.
>>
>> Justin
>> nt_jnewman at yahoo.com
>>
>> --
>> *From:* Ron Joffe 
>> *To:* asterisk-users@lists.digium.com
>> *Cc:* Asterisk Asterisk 
>> *Sent:* Monday, February 16, 2009 11:05:24 AM
>> *Subject:* Re: [asterisk-users] Please help test the gender detection
>> module at 575-613-4392
>>
>> That's an interesting module.
>>
>> Care to elaborate on what you designed it for ?
>>
>> Thanks,
>>
>> Ron
>>
>>
>>
>>
>> On Monday 16 February 2009 13:29, Asterisk Asterisk wrote:
>> > I need your help: please help test the gender detection module at
>> > 575-613-4392.
>> >
>> > I wrote a gender detection module and thought I'd try it out. It only
>> takes
>> > a second. I've been showing 90%+ accuracy and I want to make sure it's
>> > working correctly. Rain and significant background noise seems to throw
>> it
>> > off, so I still have a bit of work to do.
>> >
>> > Have your friends and significant others call too. Also, let me know if
>> you
>> > have any need for the module.
>> >
>> > Justin Newman
>> > nt_jnewman at yahoo.com
>>
>
> Tried to test and got "call could not be completed as dialed".
>
> This sounds very interesting Justin.
>
> --
> Thanks,
> Steve Totaro
>

Justin, how about building some additional functionality.

How about a moving stress variable that could be used as a lie detector of
sorts or just to measure how certain parts of a script, or certain questions
may prove to be more stressful where simply rewording them may have a less
stressful response?

I guess to get a baseline, you would have to ask a few inert questions.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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[asterisk-users] No Audio PlayBack Asterisk 1.6 Dahdi 2.1.0.3

[Daviramos Roussenq Fortunato]

Hi List.


I'm having problems with Asterisk 1.6 + DAHDI 2.1.0.3

PlayBack does not ring, is still in command, and not later in the following
context.

Disabling the dahdi operates normally.

I'm using dahdi_dummy.
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Re: [asterisk-users] Credit Card processing machines

[Andrew Joakimsen]

On Wed, Feb 18, 2009 at 12:50, bilal ghayyad  wrote:
> And is there a bank accept to give such kind of communication?
>
> The user was able to dial his card number and the amount from his phone (or 
> IP Phone registered with Asterisk), and Asterisk communicate with the bank or 
> company credit card provider?
>
> How the user will enter $50.25?
> What about expiration date of the credit card?
>

Where there are two solutions:

1) The bank provides the service... you do nothing but call the number
they provide.

2) The bank provides some sort of API (authorize.net is common in the
United States of America) and you write code (an AGI script) that a)
accepts the input via the phone b) communicates with the bank using
the API, probably via the internet using some sort of encryption
(HTTPS is pretty common)

Answers to your questions:

1) Probably just by entering 5025
2) Probably just by entering MMYY (month, month, year, year. e.g.:
1210 = December of 2010)

This is rather simple since the format is known. Currency usually has
two decimal places and years are again a standard format. If using
option 2) above it would be wise to provide a confirmation (user dials
5025 and then a prompt would say "You entered fifty dollars
twenty-five cents. Is that correct?", etc.)

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Re: [asterisk-users] Open Source in an Economic Downturn: Asterisk stories needed

[Chris Bagnall]

> Perhaps there isn't a downturn in a country with more sheep than people

That's a little harsh, New Zealand is one of those places that really appeals 
as a decent place to live.

> This could easily turn into a decade long period, with few having any real
> answers.

I think that's somewhat alarmist. Even the most pessimistic assessments over 
here in the UK seem to indicate around 18 months or so before things start to 
improve.

At the end of the day, we got ourselves into this mess (cheap credit, lenders 
not checking people could actually repay their debts, etc.), we have no-one to 
blame but ourselves.

As others have said, there are opportunities in the the current financial 
climate that wouldn't otherwise be available.

Regards,

Chris



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Re: [asterisk-users] Open Source in an Economic Downturn: Asterisk stories needed

[Michael]

On Thu, 19 Feb 2009 08:33:56 Chris Bagnall wrote:
> > Perhaps there isn't a downturn in a country with more sheep than people
>
> That's a little harsh, New Zealand is one of those places that really
> appeals as a decent place to live.

It is. And when it isn't so hot (mid summer) Australia is also quite decent.

> At the end of the day, we got ourselves into this mess (cheap credit,
> lenders not checking people could actually repay their debts, etc.), we
> have no-one to blame but ourselves.

Add to this (from the perspective of anyone living there) - a stupid war and 
throwing handouts at car makings who make products that people don't want to 
buy - watch 'who killed the electric car' on youtube.

The poster is based in the USA, so it could well be that bad for him.

As for his opinions about the rest of the world, well Fox and CNN are always 
reliable and educational.

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[asterisk-users] benchg729 - no valid g729 license

[Adam Robins]

I have five Asterisk servers running 1.2.14, and am planning to upgrade
to 1.4 this weekend.  In preparation, to use the most efficient g729
codec, I am running the new benchg729 program.  It works great on two
systems, but on the other three it says it cannot locate a valid g729
license.  I have valid licenses on all systems, which show just fine
when typing "show g729" from CLI.

Any ideas are appreciated.

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Re: [asterisk-users] Open Source in an Economic Downturn: Asterisk stories

[JR Richardson]

>
> >> I'm giving a talk at SCALE 2009 (Southern CAlifornia Linux Expo) on
> >> Sunday in Los Angeles, and the topic of my talk is "Open Source in an
> >> Economic Downturn".  I've got lots of talking points for this talk,
> >> but it would be interesting to hear some short anecdotes about how you
> >> in the Asterisk community are thriving, or at least surviving, by
> >> virtue of the benefits of Open Source.  I find that real-world
> >> examples are worth more than all of the bullet points in the world,
> >> and timely stories from the community would be more interesting than
> >> hearing me prattle on.
> >>
>
I'm a Texas based service provider, VoIP and Internet for business
customers.  The last 2 month are actually picking up a bit.  We are finding
that business folks are really wanting less expensive alternatives for voice
and data services, not so much for the new VoIP technology we offer.  A year
ago, we really had to effectively sell the new technological advantages and
promote business enhancing solutions based on voice and data convergence.
Now it is all about the bottom line cost.  I'm not real concerned for the
reason businesses are buying our service, but I'm glad they are.

Competitively speaking, we are seeing the LEC (ATT) and other CLEC providers
dropping their prices to capture market share.  They are getting real
aggressive, but it make a good statement about why their prices where
high for the past few years.  The technology and how it is delivered has not
changed much over the past year so the "Economic Downturn" has affected them
enough to reposition their margin strategies.

JR
-- 
JR Richardson
Engineering for the Masses
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[asterisk-users] call file FXO channel problem

[Ray Chen]

I have problem of using call file to make auto outbound dial through FXO
channel. I put "Channel: DAHDI/1/xx" (xx is the
destination PSTN number to dial). For some reason asterisk did not dial
the number but the control came to the context that I defined in the call
file as if the peer had answed the call. It works if I change the channel
from DAHDI to a SIP channel like SIP/4567 or I dial DAHDI/1/xx
from a SIP channel. I am using asterisk1.4.23.1. Is it a bug in this
release? Thanks Ray

-- 
Be Yourself @ mail.com!
Choose From 200+ Email Addresses
Get a Free Account at www.mail.com

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Re: [asterisk-users] benchg729 - no valid g729 license

[Kevin P. Fleming]

Adam Robins wrote:
> I have five Asterisk servers running 1.2.14, and am planning to upgrade
> to 1.4 this weekend.  In preparation, to use the most efficient g729
> codec, I am running the new benchg729 program.  It works great on two
> systems, but on the other three it says it cannot locate a valid g729
> license.  I have valid licenses on all systems, which show just fine
> when typing "show g729" from CLI.

How recently have you re-run the 'register' tool for those licenses?
It's possible the license files are in an old format that the new
programs don't expect.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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[asterisk-users] Asterisk on the Cloud With a Click - pre-built Asterisk Amazon EC2 instance

[Eric Chamberlain]

Asterisk-users,

Our two-part tutorial explaining how to use VoIP and Asterisk in  
Amazon’s Elastic Compute Cloud (EC2) has garnered quite a bit of  
attention. But due to the time required to complete the many steps  
needed to get up and running, some of you have asked if it is possible  
to create a much simpler to install “pre-built” Asterisk EC2 “instance.”

In short, yes it is. And we’ve done just that for you.

With the power of the cloud, it’s not necessary have to wait days or  
hours for servers to be rebuilt. We don’t even need to start with a  
server that has nothing more than an operating system on it. Someone  
(Voxillans) can do all the grunt work: building, compiling, installing  
software; then share the complete server with others (you). Amazon  
calls this sharing Amazon Machine Images (AMIs).

Now you have two choices, you can either build the Asterisk server  
yourself, or you can use Voxilla’s pre-built image to eliminate a lot  
of the heavy lifting.

Learn more at 
http://voxilla.com/2009/02/18/asterisk-on-the-cloud-with-a-click-1405


--
Eric Chamberlain, Founder
RF.com - http://RF.com/








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Re: [asterisk-users] benchg729 - no valid g729 license

[Adam Robins]

That did it.  Thanks.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Wednesday, February 18, 2009 4:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] benchg729 - no valid g729 license

Adam Robins wrote:
> I have five Asterisk servers running 1.2.14, and am planning to
upgrade
> to 1.4 this weekend.  In preparation, to use the most efficient g729
> codec, I am running the new benchg729 program.  It works great on two
> systems, but on the other three it says it cannot locate a valid g729
> license.  I have valid licenses on all systems, which show just fine
> when typing "show g729" from CLI.

How recently have you re-run the 'register' tool for those licenses?
It's possible the license files are in an old format that the new
programs don't expect.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] Asterisk on the Cloud With a Click - pre-built Asterisk Amazon EC2 instance

[John Todd]

On Feb 18, 2009, at 1:18 PM, Eric Chamberlain wrote:

> Asterisk-users,
>
> Our two-part tutorial explaining how to use VoIP and Asterisk in
> Amazon’s Elastic Compute Cloud (EC2) has garnered quite a bit of
> attention. But due to the time required to complete the many steps
> needed to get up and running, some of you have asked if it is possible
> to create a much simpler to install “pre-built” Asterisk EC2  
> “instance.”
>
> In short, yes it is. And we’ve done just that for you.
>
> With the power of the cloud, it’s not necessary have to wait days or
> hours for servers to be rebuilt. We don’t even need to start with a
> server that has nothing more than an operating system on it. Someone
> (Voxillans) can do all the grunt work: building, compiling, installing
> software; then share the complete server with others (you). Amazon
> calls this sharing Amazon Machine Images (AMIs).
>
> Now you have two choices, you can either build the Asterisk server
> yourself, or you can use Voxilla’s pre-built image to eliminate a lot
> of the heavy lifting.
>
> Learn more at 
> http://voxilla.com/2009/02/18/asterisk-on-the-cloud-with-a-click-1405
>
> --
> Eric Chamberlain, Founder
> RF.com - http://RF.com/



I'd like to add that this is a particularly useful image since it  
includes both a fairly recent version of Asterisk (1.6.0.5) and also  
includes functional DAHDI timers to allow MeetMe and other timing- 
specific apps to work correctly.

I'd very much like to hear of anyone doing load testing on the small,  
medium, and large instances of EC2 with this image.  I think it's a  
great way to get people working with Asterisk very quickly, and  
without having to compile or manage operating-system level issues to  
start with.  It's always easier to learn to be a mechanic by starting  
with an assembled car and working backwards than with a pile of parts,  
even if ultimately you need to be able to take the parts and build the  
car yourself.

http://developer.amazonwebservices.com/connect/entry.jspa?externalID=2086

There are several Asterisk images on EC2 that are already available,  
but this one seems to have all the parts for a baseline Asterisk  
system without too many frills.

JT

---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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[asterisk-users] Understand SIP REFER

[Klaus Darilion]

Hi!

I have some problems understanding the concept of REFER in Asterisk 1.4.23.

I have the following scenario:

Incoming SIP call (incoming leg) from a SIP trunk into Asterisk (handled 
in context fromTrunk), forwarded to the SIP Client (outgoing leg).

Now, the SIP Client sends a REFER request (unattended transfer) to 
another extension. This terminates the outgoing leg and the incoming leg 
continues dialplan processing in the context of the SIP client 
(fromSipClient).

Processing in the client's context is IMO fine, but the problem is that 
the channel is the incoming channel from the trunk. So the fromSipClient 
is processed by the trunk channel. This in my case does not work as it 
expects to have certain variables set (setvar in sip.conf) - but these 
variables are not present as the new extension is executed by the 
trunk's channel.

I it possible to execute the second call setup completely in the SIP 
clients settings (e.g. loading the clients setvar options)?

regards
klaus

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Re: [asterisk-users] call file FXO channel problem

[Eric Wieling, Asteria Solutions Group]

Ray Chen wrote:
> I have problem of using call file to make auto outbound dial through FXO 
> channel. I put "Channel: DAHDI/1/xx" (xx is the destination 
> PSTN number to dial). For some reason asterisk did not dial the number but 
> the control came to the context that I defined in the call file as if the 
> peer had answed the call. It works if I change the channel from DAHDI to a 
> SIP channel like SIP/4567 or I dial DAHDI/1/xx from a SIP channel. I 
> am using asterisk1.4.23.1. Is it a bug in this release?
>   
Analog FXO ports are considered "answered" as soon as dialing is 
finished.  The telco does not provide a signal to the calling device to 
indicate the far end answered the phone.  This does not apply to PRI or 
FXS.  Virtually all SIP service providers use PRIs.  If the service 
provider used analog you would also experience this when dialing using SIP.

-- 
Eric Wieling * Asteria Solutions Group * Huntsville, AL
Call centers * IVRs * Enterprise PBXs * Conferencing applications
256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com


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Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Asterisk Asterisk]

Steve,

>Tried to test and got "call could not be completed as dialed".

Were you able to connect? If not, please try again. Call volume has been 
growing.

>How about a moving stress variable that could be used as a lie detector
of sorts or 
>just to measure how certain parts of a script, or certain
questions may

This is possible. Do you want to call or e-mail to discuss?


>I guess to get a baseline, you would have to ask a few inert questions. 

Yes, I definitely need to do this and will probably add this in for the next 
release.

Justin Newman
nt_jnewman at yahoo.com




From: Steve Totaro 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Wednesday, February 18, 2009 10:57:47 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at 
575-613-4392




On Wed, Feb 18, 2009 at 1:28 PM, Steve Totaro  
wrote:




On Mon, Feb 16, 2009 at 2:45 PM, Asterisk Asterisk  
wrote:

This module detects gender and approximate age range. I'm working on getting 
it's accuracy to 80%+ on a consistent basis, after implementing filters to 
remove background noise and other artifacts.

It's designed for a number of things. To start, I have several clients 
(primarily mobile content and servers providers) that want to profile and 
generate demographics of their users for selling advertising. They also want to 
understand their user base. Plus, some customers have found that male and 
female users tend to respond differently to different prompts, flows, etc. This 
helps in designing a system that meets needs of many different types of users.

Of course, there are many other uses and I'm sure people can generate some cool 
ideas.

Let me know how it works when you try the test number at 575-613-4392. Also, 
let me know if you have any interest in the module.

Justin

nt_jnewman at yahoo.com





From: Ron Joffe 
To: asterisk-users@lists.digium.com
Cc: Asterisk Asterisk 
Sent: Monday, February 16, 2009 11:05:24 AM
Subject: Re: [asterisk-users] Please help test the gender detection module at 
575-613-4392

That's an interesting module.

Care to elaborate on what you designed it for ?

Thanks,

Ron




On Monday 16 February 2009 13:29, Asterisk Asterisk wrote:
> I need your help: please help test the gender detection module at
> 575-613-4392.
>
> I wrote a gender detection module and thought I'd try it out. It only takes
> a second. I've been showing 90%+ accuracy and I want to make sure it's
> working correctly. Rain and significant background noise seems to throw it
> off, so I still have a bit of work to do.
>
> Have your friends and significant others call too. Also, let me know if you
> have any need for the module.
>
> Justin Newman
> nt_jnewman at yahoo.com


Tried to test and got "call could not be completed as dialed".

This sounds very interesting Justin.   

-- 
Thanks,
Steve Totaro 


Justin, how about building some additional functionality.  

How about a moving stress variable that could be used as a lie detector of 
sorts or just to measure how certain parts of a script, or certain questions 
may prove to be more stressful where simply rewording them may have a less 
stressful response?

I guess to get a baseline, you would have to ask a few inert questions. 

-- 
Thanks,
Steve Totaro 
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)



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Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Jeff LaCoursiere]

Hi Justin,

How far is your work from being able to do speaker verification?  Not 
*identification* mind you, but being able to tell that a captured voice is 
the same as another that is stored...

Cheers,

j

On Wed, 18 Feb 2009, Asterisk Asterisk wrote:

> Steve,
>
>> Tried to test and got "call could not be completed as dialed".
>
> Were you able to connect? If not, please try again. Call volume has been 
> growing.
>
>> How about a moving stress variable that could be used as a lie detector
> of sorts or
>> just to measure how certain parts of a script, or certain
> questions may
>
> This is possible. Do you want to call or e-mail to discuss?
>
>
>> I guess to get a baseline, you would have to ask a few inert questions.
>
> Yes, I definitely need to do this and will probably add this in for the next 
> release.
>
> Justin Newman
> nt_jnewman at yahoo.com
>
>
>
> 
> From: Steve Totaro 
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Sent: Wednesday, February 18, 2009 10:57:47 AM
> Subject: Re: [asterisk-users] Please help test the gender detection module at 
> 575-613-4392
>
>
>
>
> On Wed, Feb 18, 2009 at 1:28 PM, Steve Totaro 
>  wrote:
>
>
>
>
> On Mon, Feb 16, 2009 at 2:45 PM, Asterisk Asterisk  
> wrote:
>
> This module detects gender and approximate age range. I'm working on getting 
> it's accuracy to 80%+ on a consistent basis, after implementing filters to 
> remove background noise and other artifacts.
>
> It's designed for a number of things. To start, I have several clients 
> (primarily mobile content and servers providers) that want to profile and 
> generate demographics of their users for selling advertising. They also want 
> to understand their user base. Plus, some customers have found that male and 
> female users tend to respond differently to different prompts, flows, etc. 
> This helps in designing a system that meets needs of many different types of 
> users.
>
> Of course, there are many other uses and I'm sure people can generate some 
> cool ideas.
>
> Let me know how it works when you try the test number at 575-613-4392. Also, 
> let me know if you have any interest in the module.
>
> Justin
>
> nt_jnewman at yahoo.com
>
>
>
>
> 
> From: Ron Joffe 
> To: asterisk-users@lists.digium.com
> Cc: Asterisk Asterisk 
> Sent: Monday, February 16, 2009 11:05:24 AM
> Subject: Re: [asterisk-users] Please help test the gender detection module at 
> 575-613-4392
>
> That's an interesting module.
>
> Care to elaborate on what you designed it for ?
>
> Thanks,
>
> Ron
>
>
>
>
> On Monday 16 February 2009 13:29, Asterisk Asterisk wrote:
>> I need your help: please help test the gender detection module at
>> 575-613-4392.
>>
>> I wrote a gender detection module and thought I'd try it out. It only takes
>> a second. I've been showing 90%+ accuracy and I want to make sure it's
>> working correctly. Rain and significant background noise seems to throw it
>> off, so I still have a bit of work to do.
>>
>> Have your friends and significant others call too. Also, let me know if you
>> have any need for the module.
>>
>> Justin Newman
>> nt_jnewman at yahoo.com
>
>
> Tried to test and got "call could not be completed as dialed".
>
> This sounds very interesting Justin.
>
> -- 
> Thanks,
> Steve Totaro
>
>
> Justin, how about building some additional functionality.
>
> How about a moving stress variable that could be used as a lie detector of 
> sorts or just to measure how certain parts of a script, or certain questions 
> may prove to be more stressful where simply rewording them may have a less 
> stressful response?
>
> I guess to get a baseline, you would have to ask a few inert questions.
>
> -- 
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>
>
>
>

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Re: [asterisk-users] Please help test the gender detection moduleat 575-613-4392

[Asterisk Asterisk]

I hadn't even thought of that, but that's a great idea. I wrote some code that 
does speech recognition based on generated tokens and no learning needed. We 
could certainly apply the gender detection and that sr to a project like this. 
I would store only the token in my current model, but we could store voice too.

I'll send you another email with my contact info. Maybe we can talk offline and 
put something together this week? This would be really cool...voice auth and 
ID. Probably not too much work either.

Justin Newman
nt_jnewman at yahoo.com

-Original Message-
From: Jeff LaCoursiere 
Date: Thu, 19 Feb 2009 00:01:15 
To: Asterisk Users Mailing List - Non-Commercial 
Discussion
Cc: 
Subject: Re: [asterisk-users] Please help test the gender detection module
at 575-613-4392



Hi Justin,

How far is your work from being able to do speaker verification?  Not 
*identification* mind you, but being able to tell that a captured voice is 
the same as another that is stored...

Cheers,

j

On Wed, 18 Feb 2009, Asterisk Asterisk wrote:

> Steve,
>
>> Tried to test and got "call could not be completed as dialed".
>
> Were you able to connect? If not, please try again. Call volume has been 
> growing.
>
>> How about a moving stress variable that could be used as a lie detector
> of sorts or
>> just to measure how certain parts of a script, or certain
> questions may
>
> This is possible. Do you want to call or e-mail to discuss?
>
>
>> I guess to get a baseline, you would have to ask a few inert questions.
>
> Yes, I definitely need to do this and will probably add this in for the next 
> release.
>
> Justin Newman
> nt_jnewman at yahoo.com
>
>
>
> 
> From: Steve Totaro 
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Sent: Wednesday, February 18, 2009 10:57:47 AM
> Subject: Re: [asterisk-users] Please help test the gender detection module at 
> 575-613-4392
>
>
>
>
> On Wed, Feb 18, 2009 at 1:28 PM, Steve Totaro 
>  wrote:
>
>
>
>
> On Mon, Feb 16, 2009 at 2:45 PM, Asterisk Asterisk  
> wrote:
>
> This module detects gender and approximate age range. I'm working on getting 
> it's accuracy to 80%+ on a consistent basis, after implementing filters to 
> remove background noise and other artifacts.
>
> It's designed for a number of things. To start, I have several clients 
> (primarily mobile content and servers providers) that want to profile and 
> generate demographics of their users for selling advertising. They also want 
> to understand their user base. Plus, some customers have found that male and 
> female users tend to respond differently to different prompts, flows, etc. 
> This helps in designing a system that meets needs of many different types of 
> users.
>
> Of course, there are many other uses and I'm sure people can generate some 
> cool ideas.
>
> Let me know how it works when you try the test number at 575-613-4392. Also, 
> let me know if you have any interest in the module.
>
> Justin
>
> nt_jnewman at yahoo.com
>
>
>
>
> 
> From: Ron Joffe 
> To: asterisk-users@lists.digium.com
> Cc: Asterisk Asterisk 
> Sent: Monday, February 16, 2009 11:05:24 AM
> Subject: Re: [asterisk-users] Please help test the gender detection module at 
> 575-613-4392
>
> That's an interesting module.
>
> Care to elaborate on what you designed it for ?
>
> Thanks,
>
> Ron
>
>
>
>
> On Monday 16 February 2009 13:29, Asterisk Asterisk wrote:
>> I need your help: please help test the gender detection module at
>> 575-613-4392.
>>
>> I wrote a gender detection module and thought I'd try it out. It only takes
>> a second. I've been showing 90%+ accuracy and I want to make sure it's
>> working correctly. Rain and significant background noise seems to throw it
>> off, so I still have a bit of work to do.
>>
>> Have your friends and significant others call too. Also, let me know if you
>> have any need for the module.
>>
>> Justin Newman
>> nt_jnewman at yahoo.com
>
>
> Tried to test and got "call could not be completed as dialed".
>
> This sounds very interesting Justin.
>
> -- 
> Thanks,
> Steve Totaro
>
>
> Justin, how about building some additional functionality.
>
> How about a moving stress variable that could be used as a lie detector of 
> sorts or just to measure how certain parts of a script, or certain questions 
> may prove to be more stressful where simply rewording them may have a less 
> stressful response?
>
> I guess to get a baseline, you would have to ask a few inert questions.
>
> -- 
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>
>
>
>



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Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

[Darren Wiebe]

Pretty cool.  I'm almost offended though as I'm not usually guessed as a 
female of the species. :)

Darren Wiebe
dar...@aleph-com.net

Asterisk Asterisk wrote:
> Steve,
>
> >Tried to test and got "call could not be completed as dialed".
>
> Were you able to connect? If not, please try again. Call volume has 
> been growing.
>
> >How about a moving stress variable that could be used as a lie 
> detector of sorts or
> >just to measure how certain parts of a script, or certain questions may
>
> This is possible. Do you want to call or e-mail to discuss?
>
> >I guess to get a baseline, you would have to ask a few inert questions.
>
> Yes, I definitely need to do this and will probably add this in for 
> the next release.
>
> Justin Newman
> nt_jnewman at yahoo.com
>
> 
> *From:* Steve Totaro 
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> *Sent:* Wednesday, February 18, 2009 10:57:47 AM
> *Subject:* Re: [asterisk-users] Please help test the gender detection 
> module at 575-613-4392
>
>
>
> On Wed, Feb 18, 2009 at 1:28 PM, Steve Totaro 
>  > wrote:
>
>
>
> On Mon, Feb 16, 2009 at 2:45 PM, Asterisk Asterisk
> mailto:nt_aster...@yahoo.com>> wrote:
>
> This module detects gender and approximate age range. I'm
> working on getting it's accuracy to 80%+ on a consistent
> basis, after implementing filters to remove background noise
> and other artifacts.
>
> It's designed for a number of things. To start, I have several
> clients (primarily mobile content and servers providers) that
> want to profile and generate demographics of their users for
> selling advertising. They also want to understand their user
> base. Plus, some customers have found that male and female
> users tend to respond differently to different prompts, flows,
> etc. This helps in designing a system that meets needs of many
> different types of users.
>
> Of course, there are many other uses and I'm sure people can
> generate some cool ideas.
>
> Let me know how it works when you try the test number at
> 575-613-4392. Also, let me know if you have any interest in
> the module.
>
> Justin
>
> nt_jnewman at yahoo.com 
>
> 
> 
> *From:* Ron Joffe  >
> *To:* asterisk-users@lists.digium.com
> 
> *Cc:* Asterisk Asterisk  >
> *Sent:* Monday, February 16, 2009 11:05:24 AM
> *Subject:* Re: [asterisk-users] Please help test the gender
> detection module at 575-613-4392
>
> That's an interesting module.
>
> Care to elaborate on what you designed it for ?
>
> Thanks,
>
> Ron
>
>
>
>
> On Monday 16 February 2009 13:29, Asterisk Asterisk wrote:
> > I need your help: please help test the gender detection
> module at
> > 575-613-4392.
> >
> > I wrote a gender detection module and thought I'd try it
> out. It only takes
> > a second. I've been showing 90%+ accuracy and I want to make
> sure it's
> > working correctly. Rain and significant background noise
> seems to throw it
> > off, so I still have a bit of work to do.
> >
> > Have your friends and significant others call too. Also, let
> me know if you
> > have any need for the module.
> >
> > Justin Newman
> > nt_jnewman at yahoo.com 
>
>
> Tried to test and got "call could not be completed as dialed".
>
> This sounds very interesting Justin.  
>
> -- 
> Thanks,
> Steve Totaro
>
>
> Justin, how about building some additional functionality. 
>
> How about a moving stress variable that could be used as a lie 
> detector of sorts or just to measure how certain parts of a script, or 
> certain questions may prove to be more stressful where simply 
> rewording them may have a less stressful response?
>
> I guess to get a baseline, you would have to ask a few inert questions.
>
> -- 
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>
> 
>
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-- 
Darren Wiebe
dar...@aleph-com.net

Aleph Communications
www.aleph-com.net


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Re: [asterisk-users] AGI pdf book

[Steve Edwards]

On Wed, 18 Feb 2009, Danny Nicholas wrote:

> The AGI is also applicable for (at least) PHP and C+.

An AGI (or more accurately, an executable conforming to the AGI 
specification) can be written in any language -- Fortran, assembly, shell 
script, BLISS (if you wanted to "fastagi" over to a VAX running VMS) -- I 
can't think of a "language" you couldn't use :)

My personal preference is C because of speed, flexibility, speed, small 
footprint, and speed. Oh, and I know it "best."

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] AGI pdf book

[Steve Edwards]

On Wed, 18 Feb 2009, michel freiha wrote:

> I suggest please if someone advice to me a free PDF book just dedicated 
> for AGI and nothing else

It takes a rare individual to put the effort required to write a book and 
then distribute it for free.

I would like to write it, but my kids have grown accustomed to eating :)

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] AGI pdf book

[Emmanuel Bruno]

I think PHP is the best language to write AGIs, there is a library available
(PHPAGI) and it is easier to work with as compared to the complexity of the
C language.  That book written by Nir Simionovich (Asterisk AGI 1.4 and 1.6
Programming)  clearly describe the pros and cons of writing AGI scripts in
various languages (Java/C etc...) and also his book cover the PHPAGI
library.




On Wed, Feb 18, 2009 at 7:35 PM, Steve Edwards wrote:

> On Wed, 18 Feb 2009, michel freiha wrote:
>
> > I suggest please if someone advice to me a free PDF book just dedicated
> > for AGI and nothing else
>
> It takes a rare individual to put the effort required to write a book and
> then distribute it for free.
>
> I would like to write it, but my kids have grown accustomed to eating :)
>
> 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] AGI pdf book

[Emmanuel Bruno]

There's a cheaper pdf version of his book as well.  You can get more info
at:
http://www.packtpub.com/asterisk-gateway-interface-programming/book




On Wed, Feb 18, 2009 at 7:59 PM, Emmanuel Bruno  wrote:

> I think PHP is the best language to write AGIs, there is a library
> available (PHPAGI) and it is easier to work with as compared to the
> complexity of the C language.  That book written by Nir Simionovich
> (Asterisk Gateway Interface 1.4 and 1.6 Programming)  clearly describe the
> pros and cons of writing AGI scripts in various languages (Java/C etc...)
> and also his book cover the PHPAGI library.
>
>
>
>
> On Wed, Feb 18, 2009 at 7:35 PM, Steve Edwards 
> wrote:
>
>> On Wed, 18 Feb 2009, michel freiha wrote:
>>
>> > I suggest please if someone advice to me a free PDF book just dedicated
>> > for AGI and nothing else
>>
>> It takes a rare individual to put the effort required to write a book and
>> then distribute it for free.
>>
>> I would like to write it, but my kids have grown accustomed to eating :)
>>
>> Thanks in advance,
>> 
>> Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
>> Newline Fax: +1-760-731-3000
>>
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] AGI pdf book

[Jose P. Espinal]

Just for the records,

Those of you who likes PHP simplicity (I'm one of them :) ) , but would 
like C small footprint, and almost *incomparable speed*; there's a PHP 
compiler that takes PHP code to C, and produces an optimized executable.

You can see it here:
http://www.phpcompiler.org/index.html

Of course, Open Source.

Take a look at it, seems interesting.




Emmanuel Bruno wrote:
> I think PHP is the best language to write AGIs, there is a library 
> available (PHPAGI) and it is easier to work with as compared to the 
> complexity of the C language.  That book written by Nir Simionovich 
> (Asterisk AGI 1.4 and 1.6 Programming)  clearly describe the pros and 
> cons of writing AGI scripts in various languages (Java/C etc...) and 
> also his book cover the PHPAGI library.
> 
> 
> 
> 
> On Wed, Feb 18, 2009 at 7:35 PM, Steve Edwards  @sedwards.com > wrote:
> 
> On Wed, 18 Feb 2009, michel freiha wrote:
> 
>  > I suggest please if someone advice to me a free PDF book just
> dedicated
>  > for AGI and nothing else
> 
> It takes a rare individual to put the effort required to write a
> book and
> then distribute it for free.
> 
> I would like to write it, but my kids have grown accustomed to eating :)
> 
> Thanks in advance,
> 
> Steve Edwards  sedwa...@sedwards.com
>   Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
> 
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> 
> 
> 
> 
> 
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-- 
Jose P. Espinal
http://www.eSlackware.com
IRC: [OFTC|FreeNode]
Khratos @ #slackware | #asterisk/-doc/-bugs


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Re: [asterisk-users] AGI pdf book

[Michael]

On Thu, 19 Feb 2009 13:35:25 Steve Edwards wrote:
> On Wed, 18 Feb 2009, michel freiha wrote:
> > I suggest please if someone advice to me a free PDF book just dedicated
> > for AGI and nothing else
>
> It takes a rare individual to put the effort required to write a book and
> then distribute it for free.
>
> I would like to write it, but my kids have grown accustomed to eating :)

This is everything that is wrong with Open Source - no body wants to pay for 
anything

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Re: [asterisk-users] AGI pdf book

[Roderick A. Anderson]

Michael wrote:
> On Thu, 19 Feb 2009 13:35:25 Steve Edwards wrote:
>> On Wed, 18 Feb 2009, michel freiha wrote:
>>> I suggest please if someone advice to me a free PDF book just dedicated
>>> for AGI and nothing else
>> It takes a rare individual to put the effort required to write a book and
>> then distribute it for free.
>>
>> I would like to write it, but my kids have grown accustomed to eating :)
> 
> This is everything that is wrong with Open Source - no body wants to pay for 
> anything

Hi my name is nobody.  I like to pay for many (FOSS) things.  Not as 
much as I'd like but when the right Powerball ticket makes it into my 
hand I'll do more.  :-)


Rod
-- 
> 
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[asterisk-users] SLA and Flashing BLF

[Muiz Motani]

Kevin P. Fleming wrote:
> Muiz Motani wrote:
> > I understand that the Asterisk SLA implementation is somewhat different
> > from most key systems and PBX systems. I also understand that in
> > Asterisk, one does not put an SLA line on hold since it is just a MeetMe
> > conference. However, is there any way to make the BLF flash when the
> > answering party on the Asterisk system presses the hold key on their set
> > and leaves the calling party alone in the MeetMe? The current behaviour
> > is to leave the line BLF solid, not flashing.
> 
> Actually, the code does set the device state of the MeetMe to 'hold',
> but not all SIP phones can display a 'hold' state on a line key using
> the method we use for signaling to them. What brand of phones are you
> using, and are you using up-to-date firmware for them?

They are Aastra 9331i phones. I couldn't tell you what version of firmware they
are running.

-- 
Muiz Motani 
Askari Technologies


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Re: [asterisk-users] AGI pdf book

[Michael]

> >> I would like to write it, but my kids have grown accustomed to eating :)
> >
> > This is everything that is wrong with Open Source - no body wants to pay
> > for anything
>
> Hi my name is nobody.  I like to pay for many (FOSS) things.  Not as
> much as I'd like but when the right Powerball ticket makes it into my
> hand I'll do more.  :-)

And you're going to stay this way because the ahem... 'customer' knows it's 
free and pays you accordingly.

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[asterisk-users] DeadAgi Application in asterisk 1.6

[Max Alex]

Hi All,
I have configured the phpagi application for counting the duration of call,
The call is originated from the script and after hangup the call the
duration and status will be stored.
This functionality and php script is working fine with deadagi application
with asterisk 1.4.
I have a problem with asterisk 1.6 deadagi application, when the call is
hangup at that time the script is exited and no duration and status will be
counted, So please provide help regarding this deadagi application in
asterisk 1.6 branch,
Please help me regarding this!!
Thanks in Advance!!!

Thanks,
Max Alex
Voip Developer
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Re: [asterisk-users] AOC-E pass through

[Jean-Denis Girard]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Olivier a écrit :
>> just for curiosity, is AOC-E messages sending included in telco basic
>> subscription or is an option needed for that ?
>> cheers

It's included on PRI (even partial) subscriptions, optional on BRI.


Thanks,
- --
Jean-Denis Girard

SysNux  Systèmes  Linux  en Polynésie française
http://www.sysnux.pf/   Tél: +689 50 10 40 / GSM: +689 79 75 27
-BEGIN PGP SIGNATURE-

iEYEARECAAYFAkmc89wACgkQuu7Rv+oOo/jc6ACgopAymdTqm6cI8vnTre0uZr5T
+F0AoJGzU1sgpbDiWdK8ZLVckLMmJUba
=0Mgk
-END PGP SIGNATURE-

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[asterisk-users] Managing SIP hardphones call history

[Olivier]

Hi,

I've been asked sometimes to tailor call history features embeded in SIP
hardphones.
For example, a cutomer wanted internal call to be taken out.
Another wanted calls to sorted according specific criteria.

1. Have you identified a phone offering the possibility to display as Call
History, an XML list produced on a distant web server ?
With this feature, you would simply have to tell the hardphone which query
to send and then, you would get a customized Call History.

2. Is there a standard a B2BUA could implement to tell a hardphone "Please,
don't add this call data into your call history" ?
I was thinking of some sort of "SIP Alert-Info" data or "200 Reason: Call
completed elsewhere".

Regards
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Re: [asterisk-users] AOC-E pass through

[Olivier]

2009/2/19 Jean-Denis Girard 

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Olivier a écrit :
> >> just for curiosity, is AOC-E messages sending included in telco basic
> >> subscription or is an option needed for that ?
> >> cheers
>
> It's included on PRI (even partial) subscriptions, optional on BRI.


Thanks for replying to my late question : I

I've got a customer (a University) who should be interested in advanced call
cost control techniques.
Beside limiting dialing, reading AOC-D messages would be help to keep those
costs down.

Are these AOC-D data reliable for any route (fixed to mobile, fixed to
international, ...) and up-to-date (price variation) ?
Is it easy to add this data in CDR ?

Thanks


>
>
> Thanks,
> - --
> Jean-Denis Girard
>
> SysNux  Systèmes  Linux  en Polynésie française
> http://www.sysnux.pf/   Tél: +689 50 10 40 / GSM: +689 79 75 27
> -BEGIN PGP SIGNATURE-
>
> iEYEARECAAYFAkmc89wACgkQuu7Rv+oOo/jc6ACgopAymdTqm6cI8vnTre0uZr5T
> +F0AoJGzU1sgpbDiWdK8ZLVckLMmJUba
> =0Mgk
> -END PGP SIGNATURE-
>
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Re: [asterisk-users] Asterisk supports SIP-T?

[Johansson Olle E]

18 feb 2009 kl. 13.07 skrev Raj Jain:

> On Wed, Feb 18, 2009 at 6:55 AM, Daviramos Roussenq Fortunato
>  wrote:
>> How to convert SIP-T to SIP for Asterisk?
>
> You'll need to strip out ISUP MIME body in your SIP messaging with  
> Asterisk.

I don't think you need to strip it out actually, Raj. Have you tested?

The SIP code ignores unknown bodyparts in multipart-mime attachments.
It could be a problem though, if there's something significant you  
want to
be handled in the ISUP part.

/O

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Re: [asterisk-users] Understand SIP REFER

[Johansson Olle E]

19 feb 2009 kl. 00.08 skrev Klaus Darilion:

> Hi!
>
> I have some problems understanding the concept of REFER in Asterisk  
> 1.4.23.
>
> I have the following scenario:
>
> Incoming SIP call (incoming leg) from a SIP trunk into Asterisk  
> (handled
> in context fromTrunk), forwarded to the SIP Client (outgoing leg).
>
> Now, the SIP Client sends a REFER request (unattended transfer) to
> another extension. This terminates the outgoing leg and the incoming  
> leg
> continues dialplan processing in the context of the SIP client
> (fromSipClient).
>
> Processing in the client's context is IMO fine, but the problem is  
> that
> the channel is the incoming channel from the trunk. So the  
> fromSipClient
> is processed by the trunk channel. This in my case does not work as it
> expects to have certain variables set (setvar in sip.conf) - but these
> variables are not present as the new extension is executed by the
> trunk's channel.
>
> I it possible to execute the second call setup completely in the SIP
> clients settings (e.g. loading the clients setvar options)?

I would suggest you do some processing in the TRANSFER_CONTEXT
and load the variables you need. The SIPPEER dialplan function
might be useful.

/O

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Re: [asterisk-users] Managing SIP hardphones call history

[Johansson Olle E]

19 feb 2009 kl. 07.47 skrev Olivier:

> Hi,
>
> I've been asked sometimes to tailor call history features embeded in  
> SIP hardphones.
> For example, a cutomer wanted internal call to be taken out.
> Another wanted calls to sorted according specific criteria.
>
> 1. Have you identified a phone offering the possibility to display  
> as Call History, an XML list produced on a distant web server ?
> With this feature, you would simply have to tell the hardphone which  
> query to send and then, you would get a customized Call History.
>
> 2. Is there a standard a B2BUA could implement to tell a hardphone  
> "Please, don't add this call data into your call history" ?
> I was thinking of some sort of "SIP Alert-Info" data or "200 Reason:  
> Call completed elsewhere".

That's an interesting privacy issue. No, I haven't seen that, but  
maybe people who are working more with the end-devices than
the core servers know...

/O

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