Re: [Asterisk-Users] Limit DTMF tones
On November 4, 2004 10:56 am, Henry Devito wrote:
I have a question and I can't seem to find the answer anywhere. Is there a
way to limit the amount of digits dialed? For example I have a * box set
up for the department of corrections for prisoners to call home. It has
the Digium 2 FXO/ 2 FXS card in it. I have two Lines brought in to the fxo
ports and 2 standard 2500 analog sets for the prisoners to use to dial out.
Everything seems to work great. We use * to record all the calls. After
the prisoner dials the original number and their PIN, I do not want them
to be able to send anymore DTMF tones. The PIN number is not processed by
*. It is processed by the LEC's switch. Thanks in advance
I just did something similar:
[recordcall]
exten = s,1,SetVar(CALLFILENAME=/tmp/calls/${CALLERIDNUM}-${TIMESTAMP})
exten = s,2,Monitor(wav,${CALLFILENAME},m)
exten = s,3,Playback(agent-pass)
exten = s,4,DISA(/path/to/asterisk/passwd/file)
and then just make sure that the extension you dump them in to does not allow
them to dial anything ohter than what you want (i.e. 7-digit #s or whatnot)
I don't know of any way to suppress DTMF if that's what you're talking about.
-A.
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RE: [Asterisk-Users] Limit DTMF tones
The issue is the inmates have figured out a way to dial long distance
numbers by calling different private phone numbers and using that companies
DISA to complete calls. So in order to stop that I have to suppress dtmf
after so many digits are dialed. Any idea's?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Thursday, November 04, 2004 10:13 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Limit DTMF tones
On November 4, 2004 10:56 am, Henry Devito wrote:
I have a question and I can't seem to find the answer anywhere. Is there
a
way to limit the amount of digits dialed? For example I have a * box set
up for the department of corrections for prisoners to call home. It has
the Digium 2 FXO/ 2 FXS card in it. I have two Lines brought in to the
fxo
ports and 2 standard 2500 analog sets for the prisoners to use to dial
out.
Everything seems to work great. We use * to record all the calls. After
the prisoner dials the original number and their PIN, I do not want them
to be able to send anymore DTMF tones. The PIN number is not processed by
*. It is processed by the LEC's switch. Thanks in advance
I just did something similar:
[recordcall]
exten = s,1,SetVar(CALLFILENAME=/tmp/calls/${CALLERIDNUM}-${TIMESTAMP})
exten = s,2,Monitor(wav,${CALLFILENAME},m)
exten = s,3,Playback(agent-pass)
exten = s,4,DISA(/path/to/asterisk/passwd/file)
and then just make sure that the extension you dump them in to does not
allow
them to dial anything ohter than what you want (i.e. 7-digit #s or whatnot)
I don't know of any way to suppress DTMF if that's what you're talking
about.
-A.
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RE: [Asterisk-Users] Limit DTMF tones
On Thu, 2004-11-04 at 11:30 -0600, Henry Devito wrote:
The issue is the inmates have figured out a way to dial long distance
numbers by calling different private phone numbers and using that companies
DISA to complete calls. So in order to stop that I have to suppress dtmf
after so many digits are dialed. Any idea's?
look at the code for # transfer, Asterisk is finding the DTMF while a
call is in progress. You could probably do something along the way of
checking the call timer and once it exceeds a certain point, all DTMFs
are just ignored. It doesn't seem like it would be difficult, but I
haven't looked at the code, nor thought about how that could be merged
back into the codebase as a configurable option.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Thursday, November 04, 2004 10:13 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Limit DTMF tones
On November 4, 2004 10:56 am, Henry Devito wrote:
I have a question and I can't seem to find the answer anywhere. Is there
a
way to limit the amount of digits dialed? For example I have a * box set
up for the department of corrections for prisoners to call home. It has
the Digium 2 FXO/ 2 FXS card in it. I have two Lines brought in to the
fxo
ports and 2 standard 2500 analog sets for the prisoners to use to dial
out.
Everything seems to work great. We use * to record all the calls. After
the prisoner dials the original number and their PIN, I do not want them
to be able to send anymore DTMF tones. The PIN number is not processed by
*. It is processed by the LEC's switch. Thanks in advance
I just did something similar:
[recordcall]
exten = s,1,SetVar(CALLFILENAME=/tmp/calls/${CALLERIDNUM}-${TIMESTAMP})
exten = s,2,Monitor(wav,${CALLFILENAME},m)
exten = s,3,Playback(agent-pass)
exten = s,4,DISA(/path/to/asterisk/passwd/file)
and then just make sure that the extension you dump them in to does not
allow
them to dial anything ohter than what you want (i.e. 7-digit #s or whatnot)
I don't know of any way to suppress DTMF if that's what you're talking
about.
-A.
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--
Steven Critchfield [EMAIL PROTECTED]
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Re: [Asterisk-Users] Limit DTMF tones
On November 4, 2004 12:30 pm, Henry Devito wrote: The issue is the inmates have figured out a way to dial long distance numbers by calling different private phone numbers and using that companies DISA to complete calls. So in order to stop that I have to suppress dtmf after so many digits are dialed. Any idea's? Without hacking the source... no. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Limit DTMF tones
On 11/4/2004, Henry Devito [EMAIL PROTECTED] wrote:
The issue is the inmates have figured out a way to dial long distance
numbers by calling different private phone numbers and using that companies
DISA to complete calls. So in order to stop that I have to suppress dtmf
after so many digits are dialed. Any idea's?
My, prisoners are getting devious :)
Anyways, you'd only be able to do this by hacking the code, as others
have pointed out. What you want is in res_features.c (if I understand
the code correctly), in the function called ast_bridge_call. On *
release 1.0.1 it's somewhere on line 520:
if (f (f-frametype == AST_FRAME_DTMF)) {
if (who == peer)
ast_write(chan, f);
else
ast_write(peer, f);
}
So you'd have to hack it by disabling commenting out that section. I
think this bit of code is only executed once the two legs of the call
are bridged, so it probably wouldn't affect anything else.
I also think that if you were at some point required to be able to send
DTMF after the initial dial pattern, you could programmatically via the
dialplan use the D option in the Dial application to send dtmf
digits.
Hope you do test this out before putting it live ;) Free advice, so
don't knock me out if it breaks something else!!
Flynn
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Re: [Asterisk-Users] Limit DTMF tones
Andrew Kohlsmith wrote: On November 4, 2004 12:30 pm, Henry Devito wrote: The issue is the inmates have figured out a way to dial long distance numbers by calling different private phone numbers and using that companies DISA to complete calls. So in order to stop that I have to suppress dtmf after so many digits are dialed. Any idea's? If you configured a SIP phone to not transmit inband DTMF, would asterisk translate that to inband DTMF when bridged to an inband DTMF only connection, ie your POTS line? Note: Just talking out of my head here, I've not actually tested this... In any case, chan_sip would be much more likely to be hackable to make DTMF quit working. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limit DTMF tones
On Thu, 2004-11-04 at 13:59 -0500, Andrew Thompson wrote: Andrew Kohlsmith wrote: On November 4, 2004 12:30 pm, Henry Devito wrote: The issue is the inmates have figured out a way to dial long distance numbers by calling different private phone numbers and using that companies DISA to complete calls. So in order to stop that I have to suppress dtmf after so many digits are dialed. Any idea's? If you configured a SIP phone to not transmit inband DTMF, would asterisk translate that to inband DTMF when bridged to an inband DTMF only connection, ie your POTS line? Depends on the codec if it would be able to detect and therefore squelch. Note: Just talking out of my head here, I've not actually tested this... In any case, chan_sip would be much more likely to be hackable to make DTMF quit working. As long as asterisk is looking for DTMF, and it is connected, the best place would be in the bridging where it is looking at the frames. As has been posted before, when you are reading the frames as they come in, you could just look at the frame type and decide whether it needed to be sent or acted upon. In this case, acted upon could be dropping it to the floor and replacing it with a silence frame of the proper duration. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limit DTMF tones
On 11/4/2004, Andrew Thompson [EMAIL PROTECTED] wrote: In any case, chan_sip would be much more likely to be hackable to make DTMF quit working. Possibly, but his working configuration most likely doesn't use SIP (I would presume): It has the Digium 2 FXO/ 2 FXS card in it. I have two Lines brought in to the fxo ports and 2 standard 2500 analog sets for the prisoners to use to dial out. Flynn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limit DTMF tones
Flynn wrote: Possibly, but his working configuration most likely doesn't use SIP (I would presume): It has the Digium 2 FXO/ 2 FXS card in it. I have two Lines brought in to the fxo ports and 2 standard 2500 analog sets for the prisoners to use to dial out. Yeah, I saw that, but the replies I'd seen so far were not looking real promising, so I thought I'd throw out another idea. Even if the handsets were ruggedized, a Sipura could sit in between them and asterisk. Critchfield's response about the bridge code seems the place to look, but that's going to require coding and testing. If a SIP adapter could be dropped in and as a side effect of the configuration it broke sending DTMF out, only a few changes to the dialplan would be required to get things back in order. Anyway, it was just an idea, and he did say he was looking for ideas. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Limit DTMF tones
Hi Flynn,
Feel free to contact me offline if you feel this isn't suitable
conversation for online but I read an article about this a few weeks ago
about how you were freezing out other carriers for offering cheaper
calls to inmates than the inflated prices you charged. And I don't
understand how you are legally allowed to do this.
It must be profit driven because surely I could call one of my approved
numbers eg sister and then have her pass along the information to a
third party.
Surely if you were looking to solve this 'isolation' issue and were
serious about it you would be tackling this problem another way.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Flynn
Sent: Thursday, November 04, 2004 1:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Limit DTMF tones
On 11/4/2004, Henry Devito [EMAIL PROTECTED] wrote:
The issue is the inmates have figured out a way to dial long distance
numbers by calling different private phone numbers and using that
companies
DISA to complete calls. So in order to stop that I have to suppress
dtmf
after so many digits are dialed. Any idea's?
My, prisoners are getting devious :)
Anyways, you'd only be able to do this by hacking the code, as others
have pointed out. What you want is in res_features.c (if I understand
the code correctly), in the function called ast_bridge_call. On *
release 1.0.1 it's somewhere on line 520:
if (f (f-frametype == AST_FRAME_DTMF)) {
if (who == peer)
ast_write(chan, f);
else
ast_write(peer, f);
}
So you'd have to hack it by disabling commenting out that section. I
think this bit of code is only executed once the two legs of the call
are bridged, so it probably wouldn't affect anything else.
I also think that if you were at some point required to be able to send
DTMF after the initial dial pattern, you could programmatically via the
dialplan use the D option in the Dial application to send dtmf
digits.
Hope you do test this out before putting it live ;) Free advice, so
don't knock me out if it breaks something else!!
Flynn
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Re: [Asterisk-Users] Limit DTMF tones
It has the Digium 2 FXO/ 2 FXS card in it. I have two Lines brought in to the fxo ports and 2 standard 2500 analog sets for the prisoners to use to dial out. What about simply putting a lock over the buttons after they've dialled the phone number and entered the password? -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limit DTMF tones
On Thu, 2004-11-04 at 14:51 -0500, Andrew Kohlsmith wrote: It has the Digium 2 FXO/ 2 FXS card in it. I have two Lines brought in to the fxo ports and 2 standard 2500 analog sets for the prisoners to use to dial out. What about simply putting a lock over the buttons after they've dialled the phone number and entered the password? That doesn't stop pulse dialing, It also would require someone to do the work. Might as well create an operator who did all the dialing. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
