Re: [asterisk-users] log incoming calls without answering

[Frank Vanoni]

On Thu, 2017-04-20 at 17:26 -0300, Fabio Moretti wrote:

> Any idea?

I used to play with an analog telephone line and Asterisk by using a
Linksys SPA-3102 Voice Gateway.

I think it is no longer manufactured, but maybe you con buy a used one
on eBay or you can find an equivalent device from another manufacturer.




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Re: [asterisk-users] log incoming calls without answering

[Fabio Moretti]

Thank for all the replies, a lot of input and information!

Sorry for this useless mail, but I really wanted to say thank you.


Il 20/04/2017 17:26, Fabio Moretti ha scritto:
> Hi,
>
> I've some analogic lines and I'm asked if it's possible to program an 
> asterisk for "checking" the inbound calls without answering them, doing 
> something like this:
>
> analog line 1 -+-- asterisk
>|
>\__ analog phone
>
> when a call enter, asterisk sense it and store its values (callerid, date and 
> time, etc) somewhere, but nothing more, people will answer using the old 
> analog phone.
> The goal is to have a log of the inbound calls without touching the old 
> analog system (it's shared between different subjects).
>
> I'm pretty sure it's something possible, but how to tell asterisk: "ok, call 
> this AGI, and then don't answer and do nothing more".
>
> Any idea?
>
> Thanks
>
>
>
>   
>
>

-- 
Fabio Moretti
Gerente de Sistemas
www.tecytal.com 
0800 8780
(+598) 248 77921


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Re: [asterisk-users] log incoming calls without answering

[Tzafrir Cohen]

On Thu, Apr 20, 2017 at 05:51:59PM -0300, Fabio Moretti wrote:
> Il 20/04/2017 17:32, kevin.lar...@pioneerballoon.com ha scritto:
> >
> > This gets kinda Rube Golberg-ish, but convert the incoming analog line
> > to sip, route it through asterisk and have asterisk do its thing
> > before converting it back to analog to send to the phone. Only problem
> > is you get a lot of extra hardware involved in the mix to make it
> > work. It will be a lot of expense and trouble, so you need to make
> > sure that whatever part you want asterisk to play is worth that
> > effort. Also, I wouldn't touch a fax line in this manner.
> >
> > If you could give a bit more info on what you want asterisk to do, we
> > could maybe give better advice on how to solve your problem.
> 
> Hi Kevin,
> 
> I've already proposed your solution (is the most reasonable) but they
> have more than 60 analogs lines (no faxes) and some of them terminate in
> appliances like alarms, etc, so the solution must not touch in any way
> the connection between the line and his termination: doing a analog to
> digital conversion, passing it to asterisk and the convert it back to
> analog is prone to problems (what if asterisk crashes? or if a gateway
> fail?).
> I can split the existing lines (there are no complex things like adsl or
> digital signaling), convert the branches to digital and terminate then
> into an asterisk machine, so any failure will not affect the old
> circuit, but of course I've to configure asterisk to ONLY LOG calls and
> nothing more.
> 
> This is what they want:
> - line 1 ring
> - line 1 is splitted in two, the first branch (let's say the "analog"
> branch) go to an analog phone, that rings
> - the second branch go through a gateway and then to asterisk
> - asterisk log (with an AGI for example) "line 1 rings at  from "

Simple dialplan. Depending on the type of caller ID system, you may need
to wait a few seconds (in case the caller ID is sent after the first
ring).

Thus, assuming you have a DAHDI device, your dialplan is:

exten => s,1,Wait(5)   ; check how much and if waiting is needed
 same => s,n,NoOp(Caller ID is ${CALLERID(num)} on DAHDI channel 
${CHANNEL(dahdi_channel)})

And move on to report from there.

If you also need to report the total time of the call: that might be
possible if the remote side reverses polarity of the channels on call
start and end. Information about it is currently only reported in debug
messages by chan_dahdi. So it is possible (given polarity reversal), but
tricky.

-- 
   Tzafrir Cohen
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com

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Re: [asterisk-users] log incoming calls without answering

[J Montoya or A J Stiles]

On Thursday 20 Apr 2017, Fabio Moretti wrote:
> Hi,
> 
> I've some analogic lines and I'm asked if it's possible to program an
> asterisk for "checking" the inbound calls without answering them, doing
> something like this:
> 
> analog line 1 -+-- asterisk
> 
>\__ analog phone
> 
> when a call enter, asterisk sense it and store its values (callerid, date
> and time, etc) somewhere, but nothing more, people will answer using the
> old analog phone. The goal is to have a log of the inbound calls without
> touching the old analog system (it's shared between different subjects).
> 
> I'm pretty sure it's something possible, but how to tell asterisk: "ok,
> call this AGI, and then don't answer and do nothing more".
> 
> Any idea?

You can get FXO cards, such as the TDM410P/E; which connect to an analogue 
line just like a telephone and then allow the line to be treated as a DAHDI 
channel.  (You would need an FXS card to connect to an analogue telephone; 
this has an additional power supply to generate the 48V DC line voltage and 
80V, 25 Hz ringing voltage.  Usually these cards have 4 or 8 slots for either 
FXO or FXS modules; make sure you buy the right ones.  Explaining to the 
vendor what you want it for may be the best way.)

So you could have Asterisk as an extension to your analogue line, and it could 
monitor incoming calls.  (You would not be able to monitor outgoing calls, 
though; unless you connected each analogue telephone to an FXS port, and used 
Asterisk as a simple PABX.)  

You do not need to Answer() the line to get the caller ID; it comes down the 
line  (either as 300 baud modem tones representing ASCII, or DTMF digit tones, 
depending on your telephone company)  between the polarity reversal and the 
first ring.  And there is no need to have Answer()ed before running an AGI  (in 
fact, you might have good reasons why you would want to have a pre-answer 
script, such as deciding which extension to ring, if you want to direct 
inbound callers straight back to the agent who called them earlier).

All your AGI need do is fork(), and now -- as far as you are concerned -- 
there are two parallel universes, each containing a full copy of the original 
process with the same variables and file handles.  In one, which is the parent, 
fork() returned a non-zero value  (which is the PID of the child process, and 
you might have to clear it out of the process table if it exits before you 
do);  so that process can just exit and return control to Asterisk.  In the 
child process, where the fork() call returned zero  (you don't need to know 
your parent process's PID, since any still-living grandparent will just take 
care of you),  you need just to close STDIN, STDOUT and STDERR  (which are 
still connected to Asterisk)  so it will no longer be waiting for your process 
to do anything -- now you are free!  :)  You then connect to some sort of 
database and run a query like

INSERT INTO call_history(date,source,dest) 
VALUES(NOW(),"${CALLERID(num)}","${EXTEN}"); 

except that ${CALLERID(num)} and ${EXTEN} will have to be passed into the AGI 
script as parameters.  

Then. back in extensions.conf, Dial() a local channel that does nothing but 
Wait() for long enough for the ringing to stop; either because the analogue 
phone has been picked up, or because the caller has stopped trying.  Asterisk 
should get a Hangup() event when the far end hangs up, at any rate.


If you buy a single-channel one of the cheap Chinese clone cards  (from The 
Usual Place),  it ought to last long enough and work well enough for you to 
experiment with; but DO NOT rely on these cheap cards for anything even 
remotely mission-critical.

-- 
JM

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] log incoming calls without answering

[Brandon B.]

On 2017-04-20 02:33 PM, Fabio Moretti wrote:

Yes, I'll definitely do the test before set up the whole proyect, but
the point basically is: it is possibile for asterisk to log a call
without answering it? How to do it in the dialplan? Or I'm wasting time
because an analog line who enter asterisk is always answered?


Have you considered the legal implications of what you are trying to 
accomplish? You are describing a pen register 
. Asterisk is not suitable 
for this purpose, even if it technically can be used in a fashion. You 
might want look into high impedance telephone line equipment which solve 
the technical issues.


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Re: [asterisk-users] log incoming calls without answering

[Pete Mundy]

> On 21/04/2017, at 9:33 am, Fabio Moretti  wrote:
> 
> the point basically is: it is possibile for asterisk to log a call
> without answering it? How to do it in the dialplan? Or I'm wasting time
> because an analog line who enter asterisk is always answered?

Yes.

Something similar to this will work (untested, possibly with syntax errors, but 
gives the idea):

exten => s,1,NoOp(Inbound call from callerID $CALLERID(num))
exten => s,n,Hangup

The 'Hangup' will simply tell the FXS box (which you've used to terminate the 
line) that Asterisk isn't accepting the call.

One problem could be the FXS box re-inviting Asterisk over and over (causing 
duplicate NoOp logs), but you could add a Wait in there for 2-3 minutes before 
'Hangup'. Not many calls would ring for more than 2-3 minutes. In fact the PSTN 
may limit the number of rings (I have a feeling here in NZ it's about 250 rings 
or so).

exten => s,1,NoOp(Inbound call from callerID $CALLERID(num))
exten => s,n,Wait(180)
exten => s,n,Hangup

Pete



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Re: [asterisk-users] log incoming calls without answering

[Pete Mundy]

Fabio, this doesn't answer your question directly and it's not Asterisk related 
in any way, but it's another way to engineer a solution to the problem and I've 
seen it done before.

Many analog modems will decode the caller ID on the analog line and provide it 
as part of the 'RING' notification on their serial port. This can be used as 
input on a script on a computer to log the caller ID and datestamp.

For the Asterisk solution you propose to be able to work, the analog line has 
to support caller ID. If it does that then it will work with any of these 
modems too.

It's a purely passive connection to the line (just bridge it over the pair) and 
it may be cheaper/simpler/easier for you to build.

Food for thought?

Pete



> On 21/04/2017, at 8:26 am, Fabio Moretti  wrote:
> 
> Hi,
> 
> I've some analogic lines and I'm asked if it's possible to program an 
> asterisk for "checking" the inbound calls without answering them, doing 
> something like this:
> 
> analog line 1 -+-- asterisk
>   |
>   \__ analog phone
> 
> when a call enter, asterisk sense it and store its values (callerid, date and 
> time, etc) somewhere, but nothing more, people will answer using the old 
> analog phone.
> The goal is to have a log of the inbound calls without touching the old 
> analog system (it's shared between different subjects).
> 
> I'm pretty sure it's something possible, but how to tell asterisk: "ok, call 
> this AGI, and then don't answer and do nothing more".
> 
> Any idea?
> 
> Thanks


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Re: [asterisk-users] log incoming calls without answering

[James Cloos]

> "FM" == Fabio Moretti  writes:

FM> when a call enter, asterisk sense it and store its values (callerid,
FM> date and time, etc) somewhere, but nothing more, people will answer
FM> using the old analog phone.  The goal is to have a log of the inbound
FM> calls without touching the old analog system (it's shared between
FM> different subjects).

IIUC, the pots line has both some number of analog phones a/o fax
machines on it, plus a fxo->sip gateway, yes?

You can route the sip portion to asterisk and have the dialplan log
everything but never answer.

You may want to call the Ringing dialplan application, but even that
may not be required.  OTOH, calling Ringing should prevent the gateway
from assuming that the asterisk machine never saw the INVITE.

Eventually, when the other extension answers, the fxo->sip gateway will
cancel the sip call just like it would if the caller hung up.

(There is a possibility that any given gateway may not cancel the sip
call until the analog call is completed; you need to test.)

-JimC
-- 
James Cloos  OpenPGP: 0x997A9F17ED7DAEA6

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Re: [asterisk-users] log incoming calls without answering

[John Novack_PIV2]

Fabio Moretti wrote:

Il 20/04/2017 18:09, kevin.lar...@pioneerballoon.com ha scritto:

I honestly don't know if you can do what you want without some piece
of equipment picking up the line. What I would do is get an analog
line, an analog phone, an analog to sip device (there are many to
choose from) and a basic asterisk instance. I would then make a small
test setup where the analog line goes to a splitter. One side of the
splitter goes to your analog phone. One side goes to your analog to
SIP converter and then into your asterisk instance via your ethernet
network. Use your cell phone to call the number of your analog line
and see if it works. You would have to code a basic dialplan on the
asterisk side and set up the trunk to your converter, which I am
assuming you know how to do.

Yes, I'll definitely do the test before set up the whole proyect, but
the point basically is: it is possibile for asterisk to log a call
without answering it? How to do it in the dialplan? Or I'm wasting time
because an analog line who enter asterisk is always answered?


I used to do something similar, though not for 60 lines. I had ports off a 
channel bank bridged to a propitiatory ( think Vonage ) device, or even a 
copper PSTN line.
Not sure where you are in the world, but different analog lines deliver caller 
id differently, so timing can become an issue, and you will have no clue when 
the call is either answered or abandoned.

Here in the US it is between the first and second ring
You would need to work through the timing, as if the line rings several times 
you need to detect the ring, capture the CLID, then ignore any additional rings 
for x period of time as there will only be one CLID string sent per call.


Once I converted all lines to VOIP it all became much easier.
If your analog lines are NOT in the US or Canada, you may run into different 
issues, or it might even become easier.


John Novack

--
Dog is my co-pilot


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Re: [asterisk-users] log incoming calls without answering

[Fabio Moretti]

Il 20/04/2017 18:09, kevin.lar...@pioneerballoon.com ha scritto:
>
> I honestly don't know if you can do what you want without some piece
> of equipment picking up the line. What I would do is get an analog
> line, an analog phone, an analog to sip device (there are many to
> choose from) and a basic asterisk instance. I would then make a small
> test setup where the analog line goes to a splitter. One side of the
> splitter goes to your analog phone. One side goes to your analog to
> SIP converter and then into your asterisk instance via your ethernet
> network. Use your cell phone to call the number of your analog line
> and see if it works. You would have to code a basic dialplan on the
> asterisk side and set up the trunk to your converter, which I am
> assuming you know how to do.

Yes, I'll definitely do the test before set up the whole proyect, but
the point basically is: it is possibile for asterisk to log a call
without answering it? How to do it in the dialplan? Or I'm wasting time
because an analog line who enter asterisk is always answered?







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Re: [asterisk-users] log incoming calls without answering

[kevin . larsen]

> I've already proposed your solution (is the most reasonable) but they
> have more than 60 analogs lines (no faxes) and some of them terminate in
> appliances like alarms, etc, so the solution must not touch in any way
> the connection between the line and his termination: doing a analog to
> digital conversion, passing it to asterisk and the convert it back to
> analog is prone to problems (what if asterisk crashes? or if a gateway
> fail?).
> I can split the existing lines (there are no complex things like adsl or
> digital signaling), convert the branches to digital and terminate then
> into an asterisk machine, so any failure will not affect the old
> circuit, but of course I've to configure asterisk to ONLY LOG calls and
> nothing more.
> 
> This is what they want:
> - line 1 ring
> - line 1 is splitted in two, the first branch (let's say the "analog"
> branch) go to an analog phone, that rings
> - the second branch go through a gateway and then to asterisk
> - asterisk log (with an AGI for example) "line 1 rings at  from 
"
> no more is required from asterisk, if someone answer the analog phone or
> not is not my business.
> 
Ok, so I would agree with them that a conversion to digital and back again 
would tend to break things like fax lines and alarm lines. My analog lines 
in my facilities are there because a lot of alarm systems just don't work 
with SIP at all. It's something the alarm companies are going to have to 
figure out in the next decade or so as the Telcos are moving away from 
copper and switched networks and towards fiber and packet based networks.

I honestly don't know if you can do what you want without some piece of 
equipment picking up the line. What I would do is get an analog line, an 
analog phone, an analog to sip device (there are many to choose from) and 
a basic asterisk instance. I would then make a small test setup where the 
analog line goes to a splitter. One side of the splitter goes to your 
analog phone. One side goes to your analog to SIP converter and then into 
your asterisk instance via your ethernet network. Use your cell phone to 
call the number of your analog line and see if it works. You would have to 
code a basic dialplan on the asterisk side and set up the trunk to your 
converter, which I am assuming you know how to do.

This would at least give you a fairly low cost way to test to see if you 
can trigger what you want on the Asterisk side without also triggering the 
line itself to be answered. I would also note that you would only be able 
to log incoming calls this way. I can't see a way you would be able to 
detect an outgoing call from the analog extension.

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Re: [asterisk-users] log incoming calls without answering

[Fabio Moretti]

Il 20/04/2017 17:32, kevin.lar...@pioneerballoon.com ha scritto:
>
> This gets kinda Rube Golberg-ish, but convert the incoming analog line
> to sip, route it through asterisk and have asterisk do its thing
> before converting it back to analog to send to the phone. Only problem
> is you get a lot of extra hardware involved in the mix to make it
> work. It will be a lot of expense and trouble, so you need to make
> sure that whatever part you want asterisk to play is worth that
> effort. Also, I wouldn't touch a fax line in this manner.
>
> If you could give a bit more info on what you want asterisk to do, we
> could maybe give better advice on how to solve your problem.

Hi Kevin,

I've already proposed your solution (is the most reasonable) but they
have more than 60 analogs lines (no faxes) and some of them terminate in
appliances like alarms, etc, so the solution must not touch in any way
the connection between the line and his termination: doing a analog to
digital conversion, passing it to asterisk and the convert it back to
analog is prone to problems (what if asterisk crashes? or if a gateway
fail?).
I can split the existing lines (there are no complex things like adsl or
digital signaling), convert the branches to digital and terminate then
into an asterisk machine, so any failure will not affect the old
circuit, but of course I've to configure asterisk to ONLY LOG calls and
nothing more.

This is what they want:
- line 1 ring
- line 1 is splitted in two, the first branch (let's say the "analog"
branch) go to an analog phone, that rings
- the second branch go through a gateway and then to asterisk
- asterisk log (with an AGI for example) "line 1 rings at  from "
no more is required from asterisk, if someone answer the analog phone or
not is not my business.







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Re: [asterisk-users] log incoming calls without answering

[kevin . larsen]

> From: Fabio Moretti 
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Date: 04/20/2017 03:26 PM
> Subject: [asterisk-users] log incoming calls without answering
> Sent by: asterisk-users-boun...@lists.digium.com
> 
> Hi,
> 
> I've some analogic lines and I'm asked if it's possible to program 
> an asterisk for "checking" the inbound calls without answering them,
> doing something like this:
> 
> analog line 1 -+-- asterisk
>|
>\__ analog phone
> 
> when a call enter, asterisk sense it and store its values (callerid,
> date and time, etc) somewhere, but nothing more, people will answer 
> using the old analog phone.
> The goal is to have a log of the inbound calls without touching the 
> old analog system (it's shared between different subjects).
> 
> I'm pretty sure it's something possible, but how to tell asterisk: 
> "ok, call this AGI, and then don't answer and do nothing more".
> 
> Any idea?
> 
> Thanks

This gets kinda Rube Golberg-ish, but convert the incoming analog line to 
sip, route it through asterisk and have asterisk do its thing before 
converting it back to analog to send to the phone. Only problem is you get 
a lot of extra hardware involved in the mix to make it work. It will be a 
lot of expense and trouble, so you need to make sure that whatever part 
you want asterisk to play is worth that effort. Also, I wouldn't touch a 
fax line in this manner.

If you could give a bit more info on what you want asterisk to do, we 
could maybe give better advice on how to solve your problem.

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