[Asterisk-Users] Shady dial anyone??
Nauman Farooq
nauman.farooq at 2bt.com.pk
Thu Jul 8 07:04:18 MST 2004
wondering if anybody knows this......does shady dial work only with a zap
interface or can it be configured to be used with SIP or IAX.
Nauman
-----Original Message-----
From: asterisk-users-admin at lists.digium.com
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Sent: Thursday, July 08, 2004 5:48 PM
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Subject: Asterisk-Users digest, Vol 1 #4448 - 10 msgs
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Today's Topics:
1. Re: VoIP hackers gut Caller ID (tpanton at attglobal.net)
2. Cisco 7960 NAT question (Ben Merrills)
3. Re: Small Linux Distro (matt.riddell at sineapps.com)
4. RE: Cisco 7960 NAT question (Hall, Eric M.)
5. Minimum install required for Asterisk + voicemail & SIP friends from
mysql (=?iso-8859-1?q?Umar=20Sear?=)
6. Re: ISDN, AVM C4, HFC-cards and echo (Junaid Saeed Uppal)
7. RE: ISDN, AVM C4, HFC-cards and echo (Robinson Tim-W10277)
8. Question about Cisco IP Phone 7960 (Hall, Eric M.)
9. Re: VoIP hackers gut Caller ID (Brian Cuthie)
10. Re: VoIP hackers gut Caller ID (Stuart Baggs)
--__--__--
Message: 1
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID
From: <tpanton at attglobal.net>
Date: Thu, 8 Jul 2004 11:59:00 0100
Reply-To: asterisk-users at lists.digium.com
Because if p2p voip means i get the
same volume of junk phonecalls as i
currently do spam emails
i am not even going to _think_ about
adopting it.
We _need_ authentification.
"Steve Totaro" <asterisk at totarotechnologies.com> wrote:
__________
>why regulate? nobody regulates the return address on a letter sent via
>USPS.
>
>
>----- Original Message -----
>From: "Kevin Walsh" <kevin at cursor.biz>
>To: <asterisk-users at lists.digium.com>
>Sent: Wednesday, July 07, 2004 10:00 AM
>Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID
>
>
>> Adam Hart [adam at teragen.com.au] wrote:
>> > Chris Foster wrote:
>> > > The Register is carrying a article written by Kevin Poulsen of
>> > > Securtiy Focus, calling asterisk "..the most powerful tool for
>> > > manipulating and accessing CPN data.."
>> > >
>> > > I hope NuFone doesn't drop asterisk-set-able callerid's after
>> > > this article; i've been wanting that feature from voicepluse for
>> > > a long time.
>> > >
>> > These kind of things will be reason (excuse) for Voip to be
>> > regulated
>> >
>> Perhaps service providers who allow the Caller*ID to be set should
>> insist that customers provide evidence that they own the phone
>> numbers that they want to publish, and then limit the customers'
>> choices to only the numbers in their approved list. Calling the
>> customer on the provided number(s) would be an easy way to check, and
>> a setup fee could be levied to cover the provider's time and expenses, if
required.
>>
>> Being able to discover a "blocked" Caller*ID is another matter. Both
>> are good areas for regulation.
>>
>> --
>> _/ _/ _/_/_/_/ _/ _/ _/_/_/ _/ _/
>> _/_/_/ _/_/ _/ _/ _/ _/_/ _/ K e v i n W a l s h
>> _/ _/ _/ _/ _/ _/ _/ _/_/ kevin at cursor.biz
>> _/ _/ _/_/_/_/ _/ _/_/_/ _/ _/
>>
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
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> http://lists.digium.com/mailman/listinfo/asterisk-users
--__--__--
Message: 2
Date: Thu, 8 Jul 2004 12:00:55 +0100
From: "Ben Merrills" <ben at griffin.com>
To: <asterisk-users at lists.digium.com>
Subject: [Asterisk-Users] Cisco 7960 NAT question
Reply-To: asterisk-users at lists.digium.com
I've got 4 Cisco 7960's and they're behind a firewall (sonic wall). The
asterisk box is on a WAN connection on the other end of a DS3, the phones
connect fine to the Asterisk server as you can see from the output of show
sip peers below.
tp3/tp3 <firewall-ip> D N 255.255.255.255 60665
Unmonitored
tp2/tp2 <firewall-ip> D N 255.255.255.255 60646
Unmonitored
tp1/tp1 <firewall-ip> D N 255.255.255.255 60649
Unmonitored
Now, the Cisco phones are set to use nat (nat =3D 1) and in the SIP
configuration, the phones are also configured for SIP.
[tp1]
type=3Dfriend
secret=3Dtp1
host=3Ddynamic
nat=3Dyes
callerid=3D"Test Phone 1"
I can make calls out over the phones, but can't get anything back in. If I
call voicemail say, then that's fine. But if I try and call another phone
behind the firewall, it just sits there :/
IS there a specific port range I need to open? Should I be using a different
sip config?
Cheers for any help,
Ben
www.griffin.com
--__--__--
Message: 3
From: matt.riddell at sineapps.com
To: asterisk-users at lists.digium.com
Date: Thu, 08 Jul 2004 23:21:25 +1200
Subject: Re: [Asterisk-Users] Small Linux Distro
Reply-To: asterisk-users at lists.digium.com
Thanks to everybody for your links and support. I think I have convinced
all of the shareholders (tonight) that GPL is the way for anything we
develop with regards to Asterisk, and they are going to put some more money
into building a couple of prototypes boxes...(fun!!! hacking time!!!)
So, once I have the hardware all setup and working properly, I'll post again
here and find out what kind of features people would like to see in the
LCD/button interface and what kind of statistics and options would be
beneficial.
Kind regards,
Matt Riddell
--__--__--
Message: 4
Subject: RE: [Asterisk-Users] Cisco 7960 NAT question
Date: Thu, 8 Jul 2004 07:54:49 -0400
From: "Hall, Eric M." <ehall at amaxx.com>
To: <asterisk-users at lists.digium.com>
Reply-To: asterisk-users at lists.digium.com
I had the same problem. What I found is I needed to set register with proxy
to yes in the sip config.=20
Hope this helps
-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Ben Merrills
Sent: Thursday, July 08, 2004 7:01 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Cisco 7960 NAT question
I've got 4 Cisco 7960's and they're behind a firewall (sonic wall). The
asterisk box is on a WAN connection on the other end of a DS3, the phones
connect fine to the Asterisk server as you can see from the output of show
sip peers below.
tp3/tp3 <firewall-ip> D N 255.255.255.255 60665
Unmonitored
tp2/tp2 <firewall-ip> D N 255.255.255.255 60646
Unmonitored
tp1/tp1 <firewall-ip> D N 255.255.255.255 60649
Unmonitored
Now, the Cisco phones are set to use nat (nat =3D 1) and in the SIP
configuration, the phones are also configured for SIP.
[tp1]
type=3Dfriend
secret=3Dtp1
host=3Ddynamic
nat=3Dyes
callerid=3D"Test Phone 1"
I can make calls out over the phones, but can't get anything back in. If I
call voicemail say, then that's fine. But if I try and call another phone
behind the firewall, it just sits there :/
IS there a specific port range I need to open? Should I be using a different
sip config?
Cheers for any help,
Ben
www.griffin.com
_______________________________________________
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--__--__--
Message: 5
Date: Thu, 8 Jul 2004 13:02:41 +0100 (BST)
From: =?iso-8859-1?q?Umar=20Sear?= <usedcanon at yahoo.co.uk>
To: asterisk users <asterisk-users at lists.digium.com>
Subject: [Asterisk-Users] Minimum install required for Asterisk + voicemail
& SIP friends from mysql
Reply-To: asterisk-users at lists.digium.com
I have been trying to install asterisk with MySQL for voicemail and SIP
friends.
Using redhat 9 I am installing all the base components required for
asterisk, mysql, mysql server and mysql-devel.
If I do a make clean, make install without enabling the mysql options in the
/apps/Makefile and /channels/Makefile all goes well and make completes.
If I enable the options Make fails, when it gets to
-L/usr/lib/mysql -lmysqlclient -lz
with an error /usr/bin/ld: cannot find -lz
Now If I do the same after installing Redhat 9 doing a complete install make
completes successfully.
Ideally I would (I am sure others would too) like to install the minimum
required.
Help/advice will be greatly appreciated.
Umar.
___________________________________________________________ALL-NEW Yahoo!
Messenger - sooooo many all-new ways to express yourself
http://uk.messenger.yahoo.com
--__--__--
Message: 6
Date: Thu, 8 Jul 2004 17:08:02 +0500
From: Junaid Saeed Uppal <junaid.uppal at gmail.com>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] ISDN, AVM C4, HFC-cards and echo
Reply-To: asterisk-users at lists.digium.com
I have a HFC-S Card with European Signaling , Can you tell me what
distro/kernel are you using and how did you get the card detected with linux
? a small procedure ?
regards
~uppal
On Thu, 8 Jul 2004 10:12:04 +0100, Robinson Tim-W10277
<tim.robinson at motorola.com> wrote:
>
> I have 2 HFC cards. There is no echo. If you use the bri-stuff
> drivers they use the native zaptel echo cancellers. And I have no echo.
None.
>
> Rgds
> Tim
>
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Peer
> Oliver Schmidt
> Sent: 08 July 2004 08:47
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] ISDN, AVM C4, HFC-cards and echo
>
> Hello,
>
> I've been using * for quite a while using AVM C4 card and a few
> Grandstream Budgetone 102 hard phones plus couple of HandyTone 286.
>
> Echo is a big problem. I am getting used to it, but some users complain.
>
> Anyone has experience w/ regards to echo comparing the AVM C4 with two
> HFC-cards? Before shelling out time and money, maybe someone else
> already has done so, and could tell me if it is wortwhile. I only need
> four B-channels, so two HFC cards should be all I need.
>
> Any help, pointers and tips are greatly appreciated.
>
> Thanks for your time.
>
> rgds
> pos
>
> PS: Current CAPI.CONF settings:
> [general]
> nationalprefix=0
> internationalprefix=00
> rxgain=0.8
> txgain=0.8
>
> [interfaces]
> msn=123456
> echosquelsh=1
> incomingmsn=*
> controller=1
> softdtmf=0
> context=default
> ;echocancel=yes
> ;echotail=64
> ;deflect=12345678
> devices=2
> callgroup=1
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--__--__--
Message: 7
Subject: RE: [Asterisk-Users] ISDN, AVM C4, HFC-cards and echo
Date: Thu, 8 Jul 2004 13:21:41 +0100
From: "Robinson Tim-W10277" <Tim.Robinson at motorola.com>
To: <asterisk-users at lists.digium.com>
Reply-To: asterisk-users at lists.digium.com
RH9 and the drivers from=20
http://capi4linux.thepenguin.de/download/asterisk/
You don't use CAPI or isdn4linux. =20
Rgds
Tim
-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Junaid Saeed
Uppal
Sent: 08 July 2004 13:08
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] ISDN, AVM C4, HFC-cards and echo
I have a HFC-S Card with European Signaling , Can you tell me what
distro/kernel are you using and how did you get the card detected with linux
? a small procedure ?
regards
~uppal
On Thu, 8 Jul 2004 10:12:04 +0100, Robinson Tim-W10277
<tim.robinson at motorola.com> wrote:
>=20
> I have 2 HFC cards. There is no echo. If you use the bri-stuff=20
>drivers they use the native zaptel echo cancellers. And I have no=20
>echo. None.
>=20
> Rgds
> Tim
>=20
>=20
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Peer=20
>Oliver Schmidt
> Sent: 08 July 2004 08:47
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] ISDN, AVM C4, HFC-cards and echo =20 Hello,
>=20 I've been using * for quite a while using AVM C4 card and a few=20
>Grandstream Budgetone 102 hard phones plus couple of HandyTone 286.
>=20
> Echo is a big problem. I am getting used to it, but some users=20
>complain.
>=20
> Anyone has experience w/ regards to echo comparing the AVM C4 with two
> HFC-cards? Before shelling out time and money, maybe someone else=20
> already has done so, and could tell me if it is wortwhile. I only need
> four B-channels, so two HFC cards should be all I need.
>=20
> Any help, pointers and tips are greatly appreciated.
>=20
> Thanks for your time.
>=20
> rgds
> pos
>=20
> PS: Current CAPI.CONF settings:
> [general]
> nationalprefix=3D0
> internationalprefix=3D00
> rxgain=3D0.8
> txgain=3D0.8
>=20
> [interfaces]
> msn=3D123456
> echosquelsh=3D1
> incomingmsn=3D*
> controller=3D1
> softdtmf=3D0
> context=3Ddefault
> ;echocancel=3Dyes
> ;echotail=3D64
> ;deflect=3D12345678
> devices=3D2
> callgroup=3D1
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com=20
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com=20
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
_______________________________________________
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Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--__--__--
Message: 8
Date: Thu, 8 Jul 2004 08:21:17 -0400
From: "Hall, Eric M." <ehall at amaxx.com>
To: <asterisk-users at lists.digium.com>
Subject: [Asterisk-Users] Question about Cisco IP Phone 7960
Reply-To: asterisk-users at lists.digium.com
I know this is a little off list but I can't think of a better place to ask
this question.
I upgrade the phone to 7.1 and it installed the Universal Application
Loader. Now I'm getting Protocol Application Invalid after it reads tftp
SIP(MAC).cnf
Any ideas?
Again sorry this is off topic
--__--__--
Message: 9
Date: Thu, 08 Jul 2004 08:28:47 -0400
From: Brian Cuthie <brian at systemix.com>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID
Reply-To: asterisk-users at lists.digium.com
The real problem here is that people shouldn't be using callerid as an
authentication scheme. Lots of people have had the ability to set arbitrary
clid for years and yet banks and other institutions have stupidly used it to
authenticate callers. Complaints should be directed to them and not the VoIP
industry.
-brian
Alex wrote:
>Here is what you can possibly do:
> - Steal calling cards if they are useing caller id authentication
>scheme
> - Get access to personal banking information (Citibank uses callerid
>as part of authentication process.)
> - Purchase goods and services backed up by calling verification.
>
>I can go on and on for hours. Main point of story that s@#t will hit
>the fan and VOIP will be regulated badly. Especially if some known
>terrorist will confess about using Vonage in Afaganistan.....or some of
>drug dealers/weapon traders will be cought .....
>
>Bug generraly author of that article is an idiot. He does not
>understand the difference beteween VOIP and ISDN PRI.
>
>
>-----Original Message-----
>From: asterisk-users-admin at lists.digium.com
>[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of listas
>iPfone
>Sent: Wednesday, July 07, 2004 6:26 PM
>To: asterisk-users at lists.digium.com
>Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID
>
>This is very interesting...
>
>Regulations..USA...
>
>But... what can i do faking a caller id? stolen what? what is the point?
>
>miklos
>
>----- Original Message -----
>From: "Steve Totaro" <asterisk at totarotechnologies.com>
>To: <asterisk-users at lists.digium.com>
>Sent: Wednesday, July 07, 2004 12:56 PM
>Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID
>
>
>
>
>>why regulate? nobody regulates the return address on a letter sent
>>via USPS.
>>
>>
>>----- Original Message -----
>>From: "Kevin Walsh" <kevin at cursor.biz>
>>To: <asterisk-users at lists.digium.com>
>>Sent: Wednesday, July 07, 2004 10:00 AM
>>Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID
>>
>>
>>
>>
>>>Adam Hart [adam at teragen.com.au] wrote:
>>>
>>>
>>>>Chris Foster wrote:
>>>>
>>>>
>>>>>The Register is carrying a article written by Kevin Poulsen of
>>>>>Securtiy Focus, calling asterisk "..the most powerful tool for
>>>>>manipulating and accessing CPN data.."
>>>>>
>>>>>I hope NuFone doesn't drop asterisk-set-able callerid's after this
>>>>>article; i've been wanting that feature from voicepluse for a long
>>>>>time.
>>>>>
>>>>>
>>>>>
>>>>These kind of things will be reason (excuse) for Voip to be
>>>>regulated
>>>>
>>>>
>>>>
>>>Perhaps service providers who allow the Caller*ID to be set should
>>>insist that customers provide evidence that they own the phone
>>>numbers that they want to publish, and then limit the customers'
>>>choices to only the numbers in their approved list. Calling the
>>>customer on the provided number(s) would be an easy way to check, and
>>>a setup fee could be levied to cover the provider's time and expenses, if
required.
>>>
>>>Being able to discover a "blocked" Caller*ID is another matter. Both
>>>are good areas for regulation.
>>>
>>>--
>>> _/ _/ _/_/_/_/ _/ _/ _/_/_/ _/ _/
>>> _/_/_/ _/_/ _/ _/ _/ _/_/ _/ K e v i n W a l s h
>>> _/ _/ _/ _/ _/ _/ _/ _/_/ kevin at cursor.biz
>>>_/ _/ _/_/_/_/ _/ _/_/_/ _/ _/
>>>
>>>_______________________________________________
>>>Asterisk-Users mailing list
>>>Asterisk-Users at lists.digium.com
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
--__--__--
Message: 10
From: "Stuart Baggs" <stuartbaggs at t-hosting.biz>
To: <asterisk-users at lists.digium.com>
Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID
Date: Thu, 8 Jul 2004 13:32:53 +0100
Reply-To: asterisk-users at lists.digium.com
It is imperative that the ability to set caller ID's is kept as we need this
in everyday business.
stuart
----- Original Message -----
From: "Brian Cuthie" <brian at systemix.com>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, July 08, 2004 1:28 PM
Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID
>
> The real problem here is that people shouldn't be using callerid as an
> authentication scheme. Lots of people have had the ability to set
> arbitrary clid for years and yet banks and other institutions have
> stupidly used it to authenticate callers. Complaints should be
> directed to them and not the VoIP industry.
>
> -brian
>
>
> Alex wrote:
>
> >Here is what you can possibly do:
> > - Steal calling cards if they are useing caller id authentication
> >scheme
> > - Get access to personal banking information (Citibank uses callerid
> >as part of authentication process.)
> > - Purchase goods and services backed up by calling verification.
> >
> >I can go on and on for hours. Main point of story that s@#t will hit
> >the
fan
> >and VOIP will be regulated badly. Especially if some known terrorist
> >will confess about using Vonage in Afaganistan.....or some of drug
dealers/weapon
> >traders will be cought .....
> >
> >Bug generraly author of that article is an idiot. He does not
> >understand
the
> >difference beteween VOIP and ISDN PRI.
> >
> >
> >-----Original Message-----
> >From: asterisk-users-admin at lists.digium.com
> >[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of listas
> >iPfone
> >Sent: Wednesday, July 07, 2004 6:26 PM
> >To: asterisk-users at lists.digium.com
> >Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID
> >
> >This is very interesting...
> >
> >Regulations..USA...
> >
> >But... what can i do faking a caller id? stolen what? what is the point?
> >
> >miklos
> >
> >----- Original Message -----
> >From: "Steve Totaro" <asterisk at totarotechnologies.com>
> >To: <asterisk-users at lists.digium.com>
> >Sent: Wednesday, July 07, 2004 12:56 PM
> >Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID
> >
> >
> >
> >
> >>why regulate? nobody regulates the return address on a letter sent
> >>via USPS.
> >>
> >>
> >>----- Original Message -----
> >>From: "Kevin Walsh" <kevin at cursor.biz>
> >>To: <asterisk-users at lists.digium.com>
> >>Sent: Wednesday, July 07, 2004 10:00 AM
> >>Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID
> >>
> >>
> >>
> >>
> >>>Adam Hart [adam at teragen.com.au] wrote:
> >>>
> >>>
> >>>>Chris Foster wrote:
> >>>>
> >>>>
> >>>>>The Register is carrying a article written by Kevin Poulsen of
> >>>>>Securtiy Focus, calling asterisk "..the most powerful tool for
> >>>>>manipulating and accessing CPN data.."
> >>>>>
> >>>>>I hope NuFone doesn't drop asterisk-set-able callerid's after
> >>>>>this article; i've been wanting that feature from voicepluse for
> >>>>>a long time.
> >>>>>
> >>>>>
> >>>>>
> >>>>These kind of things will be reason (excuse) for Voip to be
> >>>>regulated
> >>>>
> >>>>
> >>>>
> >>>Perhaps service providers who allow the Caller*ID to be set should
> >>>insist that customers provide evidence that they own the phone
> >>>numbers that they want to publish, and then limit the customers'
> >>>choices to only the numbers in their approved list. Calling the
> >>>customer on the provided number(s) would be an easy way to check,
> >>>and a setup fee could be levied to cover the provider's time and
expenses, if required.
> >>>
> >>>Being able to discover a "blocked" Caller*ID is another matter.
> >>>Both are good areas for regulation.
> >>>
> >>>--
> >>> _/ _/ _/_/_/_/ _/ _/ _/_/_/ _/ _/
> >>> _/_/_/ _/_/ _/ _/ _/ _/_/ _/ K e v i n W a l s h
> >>> _/ _/ _/ _/ _/ _/ _/ _/_/ kevin at cursor.biz
> >>>_/ _/ _/_/_/_/ _/ _/_/_/ _/ _/
> >>>
> >>>_______________________________________________
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> >>>
> >>>
> >>>
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> >>
> >>
> >>
> >>
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