[opensuse] connecting my telephone to the internet
Hi all, I would like to connect my telephone to the internet, so as to be able to exploit telephone internet facilities (e.g. google talk). I guees a computer with modem as server should be set up, to link my plain old telephon line to my telephone. Any comments and suggestions? Thank you Paolo -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
On Thursday 16 August 2012 11:05:06 xpol wrote:
Hi all,
I would like to connect my telephone to the internet, so as to be able to exploit telephone internet facilities (e.g. google talk). I guees a computer with modem as server should be set up, to link my plain old telephon line to my telephone. Any comments and suggestions?
Thank you Paolo
Is your phone wifi compatible ? * no configuration on a Linux computer. * simple configuration on your wifi box (web interface) So, what is your phone model ? Dsant -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
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Hi all,
I would like to connect my telephone to the internet, so as to be able to exploit telephone internet facilities (e.g. google talk). I guees a computer with modem as server should be set up, to link my plain old telephon line to my telephone. Any comments and suggestions?
This is generally known as VoIp, voice over ip. No, you can not use a modem. There are hardware cards that serve to use a plain old telephone (POT) terminal with a VoIP network, and dedicated software (llike Asterisk) but it is probably cheaper to directly buy dedicated VoIp terminals (~150€). There are sold USB headphone-and-microphone gadgets for use with services such as Skype. Likewise, there are general purpose VoIp terminals that can be used without a computer to receive and start phone calls over internet. They need registering with a provider, that has the task of negotiating the connection and knowing the IP of each party - very rough explanation, not exact, but suffices. You can also do the same with your computer, audio card, a microphone and earphones. Examples: SFLphone, Ekiga, Linphone... - -- Cheers, Carlos E. R. (from 11.4 x86_64 "Celadon" at Telcontar) -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.16 (GNU/Linux) iEYEARECAAYFAlAtUZgACgkQtTMYHG2NR9XW9gCfYh4vYonVzWK8DN9jSSv4LoaF vLkAnjWN/oIIgZbhOsHW1KKceKW42+9b =BSzf -----END PGP SIGNATURE-----
Carlos E. R. wrote:
On Thursday, 2012-08-16 at 11:05 +0200, xpol wrote:
Hi all,
I would like to connect my telephone to the internet, so as to be able to exploit telephone internet facilities (e.g. google talk). I guees a computer with modem as server should be set up, to link my plain old telephon line to my telephone. Any comments and suggestions?
This is generally known as VoIp, voice over ip.
No, you can not use a modem. There are hardware cards that serve to use a plain old telephone (POT) terminal with a VoIP network, and dedicated software (llike Asterisk) but it is probably cheaper to directly buy dedicated VoIp terminals (~150€).
There are sold USB headphone-and-microphone gadgets for use with services such as Skype.
They also work quite well with VoIP softphones.
Likewise, there are general purpose VoIp terminals that can be used without a computer to receive and start phone calls over internet. They need registering with a provider, that has the task of negotiating the connection and knowing the IP of each party - very rough explanation, not exact, but suffices.
The provider could be just a local asterisk. -- Per Jessen, Zürich (21.5°C) -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 2012-08-22 08:00, Per Jessen wrote:
Carlos E. R. wrote:
Likewise, there are general purpose VoIp terminals that can be used without a computer to receive and start phone calls over internet. They need registering with a provider, that has the task of negotiating the connection and knowing the IP of each party - very rough explanation, not exact, but suffices.
The provider could be just a local asterisk.
Yes, but connected to the internet, and known to other servers outside. Not trivial. - -- Cheers / Saludos, Carlos E. R. (from 12.1 x86_64 "Asparagus" at Telcontar) -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.18 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAlA04c0ACgkQIvFNjefEBxpvRgCeMgULJYASMoMN2Vk4Y5OVzgkH lYcAoJZJY6KHvWCluj+KgEz5eRENKpHo =8Q6A -----END PGP SIGNATURE----- -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Carlos E. R. wrote:
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On 2012-08-22 08:00, Per Jessen wrote:
Carlos E. R. wrote:
Likewise, there are general purpose VoIp terminals that can be used without a computer to receive and start phone calls over internet. They need registering with a provider, that has the task of negotiating the connection and knowing the IP of each party - very rough explanation, not exact, but suffices.
The provider could be just a local asterisk.
Yes, but connected to the internet, and known to other servers outside. Not trivial.
To make calls on the internet, the asterisk server does need an internet connection, but it doesn't need to be known to other servers. It is actually fairly trivial :-) -- Per Jessen, Zürich (26.5°C) -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 2012-08-23 17:42, Per Jessen wrote:
Carlos E. R. wrote:
The provider could be just a local asterisk.
Yes, but connected to the internet, and known to other servers outside. Not trivial.
To make calls on the internet, the asterisk server does need an internet connection, but it doesn't need to be known to other servers. It is actually fairly trivial :-)
Making calls to somebody else registered on another server? Normally any sane asterisk will refuse an incoming call from another asterisk that is unknown. - -- Cheers / Saludos, Carlos E. R. (from 12.1 x86_64 "Asparagus" at Telcontar) -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.18 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAlA2a/sACgkQIvFNjefEBxqTzgCeN6XIccjWnL2QVWNMgHL0XNia efgAnjL/nHEgsIF1RByxVSGkADSpV0UB =AksB -----END PGP SIGNATURE----- -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Carlos E. R. wrote:
On 2012-08-23 17:42, Per Jessen wrote:
Carlos E. R. wrote:
The provider could be just a local asterisk.
Yes, but connected to the internet, and known to other servers outside. Not trivial.
To make calls on the internet, the asterisk server does need an internet connection, but it doesn't need to be known to other servers. It is actually fairly trivial :-)
Making calls to somebody else registered on another server? Normally any sane asterisk will refuse an incoming call from another asterisk that is unknown.
No it won't - it would never receive any external calls if it did. Any SIP-client can call me at sip://per@jessen.ch without my asterisk server knowing about where it comes from other than it's IP-address. -- Per Jessen, Zürich (23.4°C) -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
On Thu, 2012-08-23 at 20:45 +0200, Per Jessen wrote:
Carlos E. R. wrote:
On 2012-08-23 17:42, Per Jessen wrote:
Carlos E. R. wrote:
The provider could be just a local asterisk.
Yes, but connected to the internet, and known to other servers outside. Not trivial.
To make calls on the internet, the asterisk server does need an internet connection, but it doesn't need to be known to other servers. It is actually fairly trivial :-)
Making calls to somebody else registered on another server? Normally any sane asterisk will refuse an incoming call from another asterisk that is unknown.
No it won't - it would never receive any external calls if it did. Any SIP-client can call me at sip://per@jessen.ch without my asterisk server knowing about where it comes from other than it's IP-address.
afaicr, that is configurable in asterisk. You can set that * will refuse incoming call's from hosts that did not register before. Normally voip soft/hardphones should register on your asterisk, and your asterisk should register at your VOIP-provider. You don't have to do that, but it's a security issue. Like mom's tell their kids: "never listen to strangers" ;-) hw -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Hans Witvliet wrote:
On Thu, 2012-08-23 at 20:45 +0200, Per Jessen wrote:
Carlos E. R. wrote:
On 2012-08-23 17:42, Per Jessen wrote:
Carlos E. R. wrote:
The provider could be just a local asterisk.
Yes, but connected to the internet, and known to other servers outside. Not trivial.
To make calls on the internet, the asterisk server does need an internet connection, but it doesn't need to be known to other servers. It is actually fairly trivial :-)
Making calls to somebody else registered on another server? Normally any sane asterisk will refuse an incoming call from another asterisk that is unknown.
No it won't - it would never receive any external calls if it did. Any SIP-client can call me at sip://per@jessen.ch without my asterisk server knowing about where it comes from other than it's IP-address.
afaicr, that is configurable in asterisk.
Yes, it has to be configured to accept and route inbound calls.
You can set that * will refuse incoming call's from hosts that did not register before.
Yes, I'm sure that is possible, but why would I want to do that? It would be a bit like saying I only want POTS calls from people I know?
Normally voip soft/hardphones should register on your asterisk, and your asterisk should register at your VOIP-provider.
Well, we've been running Asterisk for five years as the local phone-system, without a VoIP provider. The external lines are still ISDN, but we accept inbound calls over the internet too. Although I don't think anyone has ever tried that :-) -- Per Jessen, Zürich (19.8°C) -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 2012-08-23 20:45, Per Jessen wrote:
Carlos E. R. wrote:
Making calls to somebody else registered on another server? Normally any sane asterisk will refuse an incoming call from another asterisk that is unknown.
No it won't - it would never receive any external calls if it did. Any SIP-client can call me at sip://per@jessen.ch without my asterisk server knowing about where it comes from other than it's IP-address.
I thought that was considered a security risk. Without direct and reverse DNS check, at least? - -- Cheers / Saludos, Carlos E. R. (from 12.1 x86_64 "Asparagus" at Telcontar) -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.18 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAlA4GeMACgkQIvFNjefEBxqQPQCfcMVW/rnfmhoFZ0mu3OLCo5GW KQcAn3pr98Mm3Lq6Z9JxNM5o9B25my5r =PcwO -----END PGP SIGNATURE----- -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Carlos E. R. wrote:
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On 2012-08-23 20:45, Per Jessen wrote:
Carlos E. R. wrote:
Making calls to somebody else registered on another server? Normally any sane asterisk will refuse an incoming call from another asterisk that is unknown.
No it won't - it would never receive any external calls if it did. Any SIP-client can call me at sip://per@jessen.ch without my asterisk server knowing about where it comes from other than it's IP-address.
I thought that was considered a security risk.
Without direct and reverse DNS check, at least?
Yes, I expect Asterisk does that, but mostly for display reasons. -- Per Jessen, Zürich (19.6°C) -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Per Jessen wrote:
Carlos E. R. wrote:
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On 2012-08-23 20:45, Per Jessen wrote:
Carlos E. R. wrote:
Making calls to somebody else registered on another server? Normally any sane asterisk will refuse an incoming call from another asterisk that is unknown.
No it won't - it would never receive any external calls if it did. Any SIP-client can call me at sip://per@jessen.ch without my asterisk server knowing about where it comes from other than it's IP-address.
I thought that was considered a security risk.
Without direct and reverse DNS check, at least?
Yes, I expect Asterisk does that, but mostly for display reasons.
Maybe we're are mixing things up - I don't quite see the security risk in receiving a VoIP call from someone@some.where on the internet? The caller will not be registering on my Asterisk server, it's only an inbound call that is routed to whoever the caller wants. In my case, my phone on my desk (a Linksys SPA) is registered with the Asterisk server as extension #123, and calls to sip://per@jessen.ch are routed to that. That's all. -- Per Jessen, Zürich (19.8°C) -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 2012-08-25 09:12, Per Jessen wrote:
Maybe we're are mixing things up - I don't quite see the security risk in receiving a VoIP call from someone@some.where on the internet? The caller will not be registering on my Asterisk server, it's only an inbound call that is routed to whoever the caller wants. In my case, my phone on my desk (a Linksys SPA) is registered with the Asterisk server as extension #123, and calls to sip://per@jessen.ch are routed to that. That's all.
I don't have field experience with asterisk, only some training. Reading the documentation I understood it was a risk, but I don't recall exactly why. On a bussiness you might get a call from a longtime and good client, dispatch a cargo to be charged 30 days later, and then learn it was a fired employee or someone from a rival company, faking the ID on the phone. Yes, it is social engineering, but trusting the number you see in your terminal is part of the issue. - -- Cheers / Saludos, Carlos E. R. (from 12.1 x86_64 "Asparagus" at Telcontar) -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.18 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAlA40MoACgkQIvFNjefEBxqPlQCeLMYpOkFbnPZKrhQRJF8P0nFZ d5QAoMgDtK6QS0Eh/2WJnJISZ6TwvcDs =hNne -----END PGP SIGNATURE----- -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Carlos E. R. wrote:
On 2012-08-25 09:12, Per Jessen wrote:
Maybe we're are mixing things up - I don't quite see the security risk in receiving a VoIP call from someone@some.where on the internet? The caller will not be registering on my Asterisk server, it's only an inbound call that is routed to whoever the caller wants. In my case, my phone on my desk (a Linksys SPA) is registered with the Asterisk server as extension #123, and calls to sip://per@jessen.ch are routed to that. That's all.
I don't have field experience with asterisk, only some training. Reading the documentation I understood it was a risk, but I don't recall exactly why. On a bussiness you might get a call from a longtime and good client, dispatch a cargo to be charged 30 days later, and then learn it was a fired employee or someone from a rival company, faking the ID on the phone. Yes, it is social engineering, but trusting the number you see in your terminal is part of the issue.
But that goes for POTS too, it isn't specific to Asterisk or VOIP. I don't think anyone has ever called me using purely VoIP, but running an Asterisk server that refuses inbound SIP calls seems like having a POTS PBX that doesn't accept external calls. One security risk with Asterisk is perhaps external SIP-clients. We have a number of people who primarily work from home. They're all have office phones at home, connected to the Asterisk box over VoIP over the internet. Two risks - 1) the SIP sign-on (userid+password) is, AFAIK, not encrypted, so it could be intercepted, giving someone access to use our internal system. 2) brute force attack trying to guess the password. It is easily countered, but we had a case last year where someone managed to guess a SIP userid+password. It meant a slightly higher phone-bill that month :-) -- Per Jessen, Zürich (24.9°C) -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 2012-08-25 16:06, Per Jessen wrote:
Carlos E. R. wrote:
I don't have field experience with asterisk, only some training. Reading the documentation I understood it was a risk, but I don't recall exactly why. On a bussiness you might get a call from a longtime and good client, dispatch a cargo to be charged 30 days later, and then learn it was a fired employee or someone from a rival company, faking the ID on the phone. Yes, it is social engineering, but trusting the number you see in your terminal is part of the issue.
But that goes for POTS too, it isn't specific to Asterisk or VOIP.
At least here the ID via POTs could be trusted, the network was closed.
I don't think anyone has ever called me using purely VoIP, but running an Asterisk server that refuses inbound SIP calls seems like having a POTS PBX that doesn't accept external calls.
No, you accept calls identified by the Telco.
One security risk with Asterisk is perhaps external SIP-clients. We have a number of people who primarily work from home. They're all have office phones at home, connected to the Asterisk box over VoIP over the internet.
Two risks -
1) the SIP sign-on (userid+password) is, AFAIK, not encrypted, so it could be intercepted, giving someone access to use our internal system. 2) brute force attack trying to guess the password. It is easily countered, but we had a case last year where someone managed to guess a SIP userid+password. It meant a slightly higher phone-bill that month :-)
You can encrypt both login data and conversations (two separate configs). We did that during my training. - -- Cheers / Saludos, Carlos E. R. (from 12.1 x86_64 "Asparagus" at Telcontar) -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.18 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAlA4544ACgkQIvFNjefEBxqaSQCdEmaCIIwmZa8sTFr4rT1FdTPK jfgAoIAjsfACiZ5GxBVPio6jn9Pl+xmX =c1ei -----END PGP SIGNATURE----- -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Carlos E. R. wrote:
On 2012-08-25 16:06, Per Jessen wrote:
Carlos E. R. wrote:
I don't have field experience with asterisk, only some training. Reading the documentation I understood it was a risk, but I don't recall exactly why. On a bussiness you might get a call from a longtime and good client, dispatch a cargo to be charged 30 days later, and then learn it was a fired employee or someone from a rival company, faking the ID on the phone. Yes, it is social engineering, but trusting the number you see in your terminal is part of the issue.
But that goes for POTS too, it isn't specific to Asterisk or VOIP.
At least here the ID via POTs could be trusted, the network was closed.
I don't think anyone has ever called me using purely VoIP, but running an Asterisk server that refuses inbound SIP calls seems like having a POTS PBX that doesn't accept external calls.
No, you accept calls identified by the Telco.
Which nowadays includes Skype calls with CLID=000. It also includes calls with suppressed or unavailable CLID. I guess calls with suppressed CLID could still be known by the telco. Carlos, I see no real difference: a) accepting calls identified by the Telco. b) accepting calls identified by the IP-address and CLID. They both include all kinds of unidentified/able calls. I would like to be able to ignore all calls with suppressed CLID, but unfortunately some banks practice that by default. (and apparently the employee cannot manually "un-suppress" it).
One security risk with Asterisk is perhaps external SIP-clients. We have a number of people who primarily work from home. They're all have office phones at home, connected to the Asterisk box over VoIP over the internet.
Two risks -
1) the SIP sign-on (userid+password) is, AFAIK, not encrypted, so it could be intercepted, giving someone access to use our internal system. 2) brute force attack trying to guess the password. It is easily countered, but we had a case last year where someone managed to guess a SIP userid+password. It meant a slightly higher phone-bill that month :-)
You can encrypt both login data and conversations (two separate configs). We did that during my training.
You're right, SIP can be done with TLS, but I don't think our Asterisk supports it (1.4.x, it's back-level). -- Per Jessen, Zürich (24.0°C) -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 2012-08-25 17:13, Per Jessen wrote:
Carlos E. R. wrote:
No, you accept calls identified by the Telco.
Which nowadays includes Skype calls with CLID=000. It also includes calls with suppressed or unavailable CLID. I guess calls with suppressed CLID could still be known by the telco.
Yes, they are known by the telco, and the suppression can be denied if you are some institution like the police or emergency service. Maybe some call centers like banks, not sure.
Carlos, I see no real difference:
a) accepting calls identified by the Telco. b) accepting calls identified by the IP-address and CLID.
Maybe I'm anchored in the past O:-)
They both include all kinds of unidentified/able calls. I would like to be able to ignore all calls with suppressed CLID, but unfortunately some banks practice that by default. (and apparently the employee cannot manually "un-suppress" it).
I believe the practice was ruled illegal here, in Spain. Any business call must have call ID. Nowdays when I see a call without ID I know it is a phony business and I tend to not lift the phone. If I do answer and they say they are my bank, I refuse to believe they, are and hang up; but typically they say they are some new phone company and want me as client, which I refuse as well, quite abruptly.
You can encrypt both login data and conversations (two separate configs). We did that during my training.
You're right, SIP can be done with TLS, but I don't think our Asterisk supports it (1.4.x, it's back-level).
Ah, that's possible. The asterisk documentation warns about many security risks and say that you must keep your installation updated :-) - -- Cheers / Saludos, Carlos E. R. (from 12.1 x86_64 "Asparagus" at Telcontar) -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.18 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAlA47sAACgkQIvFNjefEBxr8ywCfciphKKorNJ1nTPiCD5N3i1vI z44An2a1VRkUJyPNo2xXGR1DjKTKr+ec =544R -----END PGP SIGNATURE----- -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Carlos E. R. wrote:
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On 2012-08-25 17:13, Per Jessen wrote:
Carlos E. R. wrote:
No, you accept calls identified by the Telco.
Which nowadays includes Skype calls with CLID=000. It also includes calls with suppressed or unavailable CLID. I guess calls with suppressed CLID could still be known by the telco.
Yes, they are known by the telco, and the suppression can be denied if you are some institution like the police or emergency service. Maybe some call centers like banks, not sure.
Ah ok, I did have the feeling the telco sh/would know.
They both include all kinds of unidentified/able calls. I would like to be able to ignore all calls with suppressed CLID, but unfortunately some banks practice that by default. (and apparently the employee cannot manually "un-suppress" it).
I believe the practice was ruled illegal here, in Spain. Any business call must have call ID. Nowdays when I see a call without ID I know it is a phony business and I tend to not lift the phone.
Same here, but my wife works for a bank, and all outbound calls have suppressed CLID. I don't know the reasoning. It might be banking secrecy related.
You can encrypt both login data and conversations (two separate configs). We did that during my training.
You're right, SIP can be done with TLS, but I don't think our Asterisk supports it (1.4.x, it's back-level).
Ah, that's possible.
The asterisk documentation warns about many security risks and say that you must keep your installation updated :-)
Yup, we're on the most recent 1.4.x :-) - I think the next release is 1.8, but lots of stuff changed so we're staying with 1.4.x until we cannot avoid upgrading. That's the problem with stuff that just have to work - you don't get to work on it very often. -- Per Jessen, Zürich (21.2°C) -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Carlos E. R. wrote:
At least here the ID via POTs could be trusted, the network was closed.
Assuming the phone companies there allow 3rd party hardware to be connected, I could easily spoof a phone number. -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Per Jessen wrote:
Two risks -
1) the SIP sign-on (userid+password) is, AFAIK, not encrypted, so it could be intercepted, giving someone access to use our internal system. 2) brute force attack trying to guess the password. It is easily countered, but we had a case last year where someone managed to guess a SIP userid+password. It meant a slightly higher phone-bill that month:-)
Those problems have existed for years and are not unique to VoIP. Many companies have discovered large long distance bills when someone found out how to access their trunks. Last I heard, touch tones aren't encrypted either, so it's easy enough for someone to capture the dialed digits. Touch tone decoder chips are readily available. -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
James Knott wrote:
Per Jessen wrote:
Two risks -
1) the SIP sign-on (userid+password) is, AFAIK, not encrypted, so it could be intercepted, giving someone access to use our internal system. 2) brute force attack trying to guess the password. It is easily countered, but we had a case last year where someone managed to guess a SIP userid+password. It meant a slightly higher phone-bill that month:-)
Those problems have existed for years and are not unique to VoIP. Many companies have discovered large long distance bills when someone found out how to access their trunks.
No, the problem isn't new, but the exposure is much larger. I think I see a brute force attempt about every day. -- Per Jessen, Zürich (17.0°C) -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On Saturday, 2012-08-25 at 09:12 +0200, Per Jessen wrote:
Maybe we're are mixing things up - I don't quite see the security risk in receiving a VoIP call from someone@some.where on the internet?
Maybe the situation is somewhat similar to smtp. Some years ago I could have my sendmail send emails with a dynamic IP address, no problem. Nowdays most posts sent that way are rejected, via antispam rules. - -- Cheers, Carlos E. R. (from 12.1 x86_64 "Asparagus" at Telcontar) -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.18 (GNU/Linux) iEYEARECAAYFAlA5GjgACgkQtTMYHG2NR9XpbwCePI+TE++jXClEPGkSAYJXML2H G4UAnA/6zftdKGJGa287WEsngSceZZ1p =OCrD -----END PGP SIGNATURE----- -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Carlos E. R. wrote:
o make calls on the internet, the asterisk server does need an internet connection, but
it doesn't need to be known to other servers. It is actually fairly trivial:-) Making calls to somebody else registered on another server? Normally any sane asterisk will refuse an incoming call from another asterisk that is unknown.
How is accepting a call from an asterisk server different that one coming in on an analog line? Also, ideally, calls between VoIP systems should be via IP, rather than conversion to/from "POTS" service at each end. -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 2012-08-23 23:21, James Knott wrote:
Carlos E. R. wrote:
o make calls on the internet, the asterisk server does need an internet connection, but
it doesn't need to be known to other servers. It is actually fairly trivial:-) Making calls to somebody else registered on another server? Normally any sane asterisk will refuse an incoming call from another asterisk that is unknown.
How is accepting a call from an asterisk server different that one coming in on an analog line? Also, ideally, calls between VoIP systems should be via IP, rather than conversion to/from "POTS" service at each end.
The POTS system does not accept unauthentified calls. You, as user, may not know where the call comes from, but the telcos do. Plus, for a call going from a customer of one telco to a customer of another telco there has to be an explicit agreement between both telcos, on both technical and commercial sides. It is very complex. On the other hand, do you remember those movies where the bad guy makes a phone call to the police and the call has to last less than 20" or he is identified? That's rubbish, you are identified on on the first second, while ringing, at least on countries like Spain. Even if you tell your phone to hide your ID, you are identified if the recipient has the feature "do not accept hidden ID", which the police has, obviously. The problem nowdays is identifying calls from internet! From some gateways that allow the users to setup any calling number they desire without checking them. - -- Cheers / Saludos, Carlos E. R. (from 12.1 x86_64 "Asparagus" at Telcontar) -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.18 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAlA4GVAACgkQIvFNjefEBxqrQwCfUE56PCP9afw9A8EW6O1PjPXY GyMAnibn6tN8CV3fjHcrulXVeV1nh6na =AqkM -----END PGP SIGNATURE----- -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Carlos E. R. wrote:
On the other hand, do you remember those movies where the bad guy makes a phone call to the police and the call has to last less than 20" or he is identified? That's rubbish, you are identified on on the first second, while ringing, at least on countries like Spain.
Actually, it didn't used to be rubbish. I have worked in telecom going back 40 years and at one point early in my career, I actually worked on some of the old step by step relay systems and even learned to trace calls through them. There's no way it could be done in only 20 seconds. Modern, SS7 sytstems are of course quite different. Also, the phone ID can be spoofed. I get plenty of telemarketing calls where that is done. The originating switch or PBX is the one that provides the ID and it can be set to anything. For example, according to my phone's log, I received a call on Aug 10 from 10000000000. That is in no way a valid number, yet it managed to reach my phone. This is via a regular phone company that provides my home phone. Also, a few years ago, I worked on a phone system where the customer insisted there be no caller ID on outgoing calls, as it was for a woman's shelter and they didn't want anyway for someone to track where those women were. I have also set up VoIP PBXs, where someone in one location could call through another location and appear as though they originated at that 2nd location. It's not at all difficult to do with today's equipment, so accepting from the phone company is not quite as secure as you think. -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
James Knott wrote:
Carlos E. R. wrote:
On the other hand, do you remember those movies where the bad guy makes a phone call to the police and the call has to last less than 20" or he is identified? That's rubbish, you are identified on on the first second, while ringing, at least on countries like Spain.
Actually, it didn't used to be rubbish. I have worked in telecom going back 40 years and at one point early in my career, I actually worked on some of the old step by step relay systems and even learned to trace calls through them. There's no way it could be done in only 20 seconds. Modern, SS7 sytstems are of course quite different. Also, the phone ID can be spoofed. I get plenty of telemarketing calls where that is done. The originating switch or PBX is the one that provides the ID and it can be set to anything. For example, according to my phone's log, I received a call on Aug 10 from 10000000000. That is in no way a valid number, yet it managed to reach my phone. This is via a regular phone company that provides my home phone. Also, a few years ago, I worked on a phone system where the customer insisted there be no caller ID on outgoing calls, as it was for a woman's shelter and they didn't want anyway for someone to track where those women were. I have also set up VoIP PBXs, where someone in one location could call through another location and appear as though they originated at that 2nd location. It's not at all difficult to do with today's equipment, so accepting from the phone company is not quite as secure as you think.
Further on this. I have a fair bit of experience with the Adtran 550 series multiplexer. This device can connect to a variety of devices and services by plugging in the appropriate interface cards and making the appropriate configurations. I could take a DS1 (T1, E1) from the phone company and assign the channels as I desire. I could have local phones connected or as on one project, I could send voice lines via ISDN, that actually travelled back over the DS1 and on to some other location. The phone company would have absolutely no idea that I had done that or where the other location was. I'd then pick a number, including something like that 10000000000 number I mentioned and it would show as the caller ID. So, not even the police would know where that remote phone was located, as the phone company could not trace it beyond the end of that DS1 trunk. Normally, however, I would give that phone a proper phone number out of the customer's block of numbers. I have extended phone systems on many occasions, sometimes hundreds or even thousands of kilometres¹ from where the calls actually appear on the phone network. With some types of equipment, the user could even move their office phone to anywhere they had a decent internet connection or even an analog dial up phone. So, between moving the phones and providing dummy numbers, you really have no idea where an incoming call might be coming from, even though it comes to you via the local phone company. 1) Take a look at a map of Canada. I have extended phones as far a from near Toronto Ontario to Nova Scotia, about 1300 Km or Sarnia Ontario to St. John New Brunswick, over 1700 Km. -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 2012-08-25 04:45, James Knott wrote:
1) Take a look at a map of Canada. I have extended phones as far a from near Toronto Ontario to Nova Scotia, about 1300 Km or Sarnia Ontario to St. John New Brunswick, over 1700 Km.
Quite a distance. By the way, I was surprised, during my last visit to Ottawa, that phones displayed the name of the caller instead of the number, without having to enter a phone list on your terminal. Very interesting and nice. Long distance calls did not have that feature. It is curious that telcos in different countries offer so different features - like the one I want to block unwanted callers that we have not in Spain, but you already had in the 1990s Also, talking of these things, the UK telco is going the VoIP route, I understand. They are overhauling their network. But my news on that are old. - -- Cheers / Saludos, Carlos E. R. (from 12.1 x86_64 "Asparagus" at Telcontar) -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.18 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAlA4z+gACgkQIvFNjefEBxrJmwCeMR1y30iNPNQfJnv3ABjOxc9G 3A0AoJnE6i/VSaLi2o1Gva4BeVc3Q2UF =DwDo -----END PGP SIGNATURE----- -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Carlos E. R. wrote:
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1
On 2012-08-25 04:45, James Knott wrote:
1) Take a look at a map of Canada. I have extended phones as far a from near Toronto Ontario to Nova Scotia, about 1300 Km or Sarnia Ontario to St. John New Brunswick, over 1700 Km.
Quite a distance.
By the way, I was surprised, during my last visit to Ottawa, that phones displayed the name of the caller instead of the number, without having to enter a phone list on your terminal. Very interesting and nice.
I don't know about the Canadian setup, but it's pretty common on VoIP phones, Siemens Gigaset for instance. They use a local directory (in Switzerland, http://local.ch) to do a quick http lookup of the callerid. Our Asterisk server does the same. -- Per Jessen, Zürich (24.6°C) -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 2012-08-25 15:57, Per Jessen wrote:
Carlos E. R. wrote:
By the way, I was surprised, during my last visit to Ottawa, that phones displayed the name of the caller instead of the number, without having to enter a phone list on your terminal. Very interesting and nice.
I don't know about the Canadian setup, but it's pretty common on VoIP phones, Siemens Gigaset for instance. They use a local directory (in Switzerland, http://local.ch) to do a quick http lookup of the callerid. Our Asterisk server does the same.
VoIP is quite different, it is much more configurable. We tried to do something like that but it was not that easy (from scratch). I think it needed an ldap server. - -- Cheers / Saludos, Carlos E. R. (from 12.1 x86_64 "Asparagus" at Telcontar) -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.18 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAlA46GsACgkQIvFNjefEBxookgCff11xjR9T5nqHhhZspH2rb2r5 jqIAoK1pOp5wXuwKitMMSzCFdEau9d4F =etIU -----END PGP SIGNATURE----- -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Carlos E. R. wrote:
By the way, I was surprised, during my last visit to Ottawa, that phones displayed the name of the caller instead of the number, without having to enter a phone list on your terminal. Very interesting and nice. Long distance calls did not have that feature. It is curious that telcos in different countries offer so different features - like the one I want to block unwanted callers that we have not in Spain, but you already had in the 1990s
Canada has long been at the forefront of the telecommunications industry, going all the way back to when Alexander Graham Bell made his first long distance call between Paris and Brantford Ontario (via the telegraph lines owned by a predecessor of a company I used to work for). We were also the first to have a domestic communications satellite. Caller ID has been here for years. My home phone has displayed the caller's name for as long as I've had call display. My cell phone used to display number only and look up name in my phone book, until I recently changed plans. It now shows the callers name, without having to look it up in my phone book. It's only in the past several years that Europe has started to catch up with what has long been available in North America.
Also, talking of these things, the UK telco is going the VoIP route, I understand. They are overhauling their network. But my news on that are old.
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On 2012/08/25 15:36 (GMT-0400) James Knott composed:
My home phone has displayed the caller's name for as long as I've had call display. My cell phone used to display number only and look up name in my phone book, until I recently changed plans. It now shows the callers name, without having to look it up in my phone book.
At the start here when it was Bellsouth, callerid was # only. Later it became an extra cost option to also have name if the phone supported the different protocol it required. It's a rip now because less than half incoming calls display anything for name. Close to half the number matches up to "unknown name". Sometimes "name" is just the calling number without any spaces or hyphens. About half the time there is not a name it displays "cell service". AT&T claims it can't do anything for force incoming to provide name. If I was running the FCC the name assigned to the number would be absolutely required unless the caller's ancient technology prevented even the number from being displayed, IOW, callerid would be all or nothing, and those without could be blocked with a messages saying why. You wouldn't answer your door if the person was wearing a mask, would you? -- "The wise are known for their understanding, and pleasant words are persuasive." Proverbs 16:21 (New Living Translation) Team OS/2 ** Reg. Linux User #211409 ** a11y rocks! Felix Miata *** http://fm.no-ip.com/ -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 2012-08-25 22:27, Felix Miata wrote:
If I was running the FCC the name assigned to the number would be absolutely required unless the caller's ancient technology prevented even the number from being displayed,
The equivalent in Spain mandated that old exchanges be adapted to forward the caller ID number. They did that on the Alcatel 1240, which is the most used switch here, AFAIK (besides the electromechanical pentaconta). - -- Cheers / Saludos, Carlos E. R. (from 12.1 x86_64 "Asparagus" at Telcontar) -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.18 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAlA5QvcACgkQIvFNjefEBxpnLgCeIVJtiGh5hRPooaHH/Yom0z6/ h4MAnjOJ0sDCiRCBoVxsmDso1hPCBghW =wM7k -----END PGP SIGNATURE----- -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Felix Miata wrote:
If I was running the FCC the name assigned to the number would be absolutely required unless the caller's ancient technology prevented even the number from being displayed, IOW, callerid would be all or nothing, and those without could be blocked with a messages saying why. You wouldn't answer your door if the person was wearing a mask, would you?
What about that woman's shelter I mentioned, where that number might be used to help someone find a woman they're abusing? What about other situations where confidentiality might be desired, such as when a guy calls home from a bar and tells his wife he's at work? ;-) I have call display (it even shows up on my TV, if I'm watching it) and I don't answer calls from numbers or people I don't recognize. I have voice mail and if it's important, they can leave a message. Some people also get my cell phone number, which they can call me on. -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
On 2012/08/25 17:47 (GMT-0400) James Knott composed:
Felix Miata wrote:
If I was running the FCC the name assigned to the number would be absolutely required unless the caller's ancient technology prevented even the number from being displayed, IOW, callerid would be all or nothing, and those without could be blocked with a messages saying why. You wouldn't answer your door if the person was wearing a mask, would you?
What about that woman's shelter I mentioned, where that number might be used to help someone find a woman they're abusing? What about other situations where confidentiality might be desired, such as when a guy calls home from a bar and tells his wife he's at work? ;-)
I didn't say what I really meant. I have no problem with per call blocking of both name and number or private or unpublished numbers. My problem is with nameless numbers. If there was a way I would block all calls that don't provide both from even ringing my phone. Only about one in 15 incoming calls here are not pure annoyances. Without the do not call list it would probably be 1 in 30. Most that are not wrong numbers, which themselves number roughly half, seem to come from beggars and politikers, the rest from DNCL violations. -- "The wise are known for their understanding, and pleasant words are persuasive." Proverbs 16:21 (New Living Translation) Team OS/2 ** Reg. Linux User #211409 ** a11y rocks! Felix Miata *** http://fm.no-ip.com/ -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 2012-08-26 00:40, Felix Miata wrote:
On 2012/08/25 17:47 (GMT-0400) James Knott composed:
Without the do not call list it would
That "do not call" feature is one we do not have here. You are fortunate.
the rest from DNCL violations.
What is that? Ah, the wikipedia explains. In Spain we have the "robinson list". You know, the chap in the island, not my list, eh? ;-) You give your data to that list, and telemarketers are supposed to check that list before phoning; the problem is that they do get that list, so they have my data... and they can deny they saw me in the robinson list. It doesn't work :-/ And the wikipedia says the DNCL list has the same problem I describe, I understand. It would be better that the said telemarketers should register their phone numbers on a list. When one phones me, my system checks the list of callers, and if it is there, I deny access. Any non listed phone would get an automated response to leave a message with some other agency, and they decide to allow or not to allow (marketers: no way. Friends, maybe). It is a complicated life this we live... I considered using hylafax in Linux or something to check the caller ID and refuse or accept the call based on that. Asterisk would be nice, but I would need quite some hardware: dedicated PC, analog card... - -- Cheers / Saludos, Carlos E. R. (from 12.1 x86_64 "Asparagus" at Telcontar) -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.18 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAlA5bDwACgkQIvFNjefEBxoc8gCgzGRRjVfuA6yxkGIn4AhDvR+g JgQAoLIERriMJ5ndXiRiWxaDotbQk8to =EFTg -----END PGP SIGNATURE----- -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Carlos E. R. wrote:
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1
On 2012-08-26 00:40, Felix Miata wrote:
On 2012/08/25 17:47 (GMT-0400) James Knott composed:
Without the do not call list it would
That "do not call" feature is one we do not have here. You are fortunate.
We have something similar - if you tick the right box, your number is marked as do-not-call (for marketing purposes). It works fairly well. Asterisk will help with the rest :-)
I considered using hylafax in Linux or something to check the caller ID and refuse or accept the call based on that. Asterisk would be nice, but I would need quite some hardware: dedicated PC, analog card...
FYI, any old PC is sufficient. Our first Asterisk installation ran on an old desktop machine. -- Per Jessen, Zürich (16.9°C) -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 2012-08-26 10:10, Per Jessen wrote:
Carlos E. R. wrote:
FYI, any old PC is sufficient. Our first Asterisk installation ran on an old desktop machine.
Not if you load the answering robot :-p One of the features of asterisk, if I understood it well, is that it does codec translation. In normal VoIP, the the negotiation between two points uses a registrar, but the conversation itself only involves the end-points, the "phones". But with asterisk it is also involved, because it is used to match different codecs between bot ends. I suppose that if both can negotiate a common codec, they use it and free asterisk, but if not, asterisk does the translation. It means that with asterisk you can converse with anybody, but you load more the machine - which is (one of the reasons?) why the documentation say that the machine should not run X. One thing I looked to see if it existed, was separate terminals for video conferencing (with asterisk?). You know, phone terminal with a small display. Have only seen them in SciFi movies. - -- Cheers / Saludos, Carlos E. R. (from 12.1 x86_64 "Asparagus" at Telcontar) -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.18 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAlA6CYkACgkQIvFNjefEBxo25ACfa1Z6DfgYw7xO/ornTLwnQ4fU l54AoKIMl5AmtAaHItxBPViyR5uwS+lV =50Dv -----END PGP SIGNATURE----- -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Carlos E. R. wrote:
One thing I looked to see if it existed, was separate terminals for video conferencing (with asterisk?). You know, phone terminal with a small display. Have only seen them in SciFi movies.
Funny you should mention that. Just last week, I was experimenting with video chat with a friend, using our tablets. Many smart phones can also do that, as well as computers with a web cam. It's been here long enough to be used on some TV shows. I used Google Talk, but Skype also supports video chats IIRC. My "terminal" in that experiment was a 7" Android tablet. -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Carlos E. R. wrote:
On 2012-08-26 10:10, Per Jessen wrote:
Carlos E. R. wrote:
FYI, any old PC is sufficient. Our first Asterisk installation ran on an old desktop machine.
Not if you load the answering robot :-p
Voicemail you mean? It obviously depends on the number of people in your location, but our server is a very elderly 1U HP Proliant 2x800MHz with 512M RAM. We have 12 people, 4 ISDN BRI lines, voicemail and the usual stuff (music-on-hold, parking, diversion etc etc).
One of the features of asterisk, if I understood it well, is that it does codec translation. In normal VoIP, the the negotiation between two points uses a registrar, but the conversation itself only involves the end-points, the "phones". But with asterisk it is also involved, because it is used to match different codecs between bot ends. I suppose that if both can negotiate a common codec, they use it and free asterisk, but if not, asterisk does the translation. It means that with asterisk you can converse with anybody, but you load more the machine
That is correct, but you just make sure you pick the right codec, and no conversion is required. I can't remember which one, but it's called G729-something. It's also used for ISDN and mobile phones, iirc.
- which is (one of the reasons?) why the documentation say that the machine should not run X.
Well, it would usually be a server anyway. -- Per Jessen, Zürich (20.6°C) -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
James Knott wrote:
Also, the phone ID can be spoofed. I get plenty of telemarketing calls where that is done. The originating switch or PBX is the one that provides the ID and it can be set to anything.
Yep, Asterisk does that, but it is limited to the number-block assigned to our lines, I think.
Also, a few years ago, I worked on a phone system where the customer insisted there be no caller ID on outgoing calls, as it was for a woman's shelter and they didn't want anyway for someone to track where those women were.
Suppressed caller-id is a pretty normal thing. In Asterisk you can tell whether CLID was suppressed or unavailable.
I have also set up VoIP PBXs, where someone in one location could call through another location and appear as though they originated at that 2nd location.
Yep, that's also not a problem with Asterisk. -- Per Jessen, Zürich (19.9°C) -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 2012-08-25 04:07, James Knott wrote:
Carlos E. R. wrote:
On the other hand, do you remember those movies where the bad guy makes a phone call to the police and the call has to last less than 20" or he is identified? That's rubbish, you are identified on on the first second, while ringing, at least on countries like Spain.
Actually, it didn't used to be rubbish. I have worked in telecom going back 40 years and at one point early in my career, I actually worked on some of the old step by step relay systems and even learned to trace calls through them. There's no way it could be done in only 20 seconds. Modern, SS7 sytstems are of course quite different. Also, the phone ID can be spoofed. I get plenty of telemarketing calls where that is done. The originating switch or PBX is the one that provides the ID and it can be set to anything. For example, according to my phone's log, I received a call on Aug 10 from 10000000000. That is in no way a valid number, yet it managed to reach my phone. This is via a regular phone company that provides my home phone. Also, a few years ago, I worked on a phone system where the customer insisted there be no caller ID on outgoing calls, as it was for a woman's shelter and they didn't want anyway for someone to track where those women were. I have also set up VoIP PBXs, where someone in one location could call through another location and appear as though they originated at that 2nd location. It's not at all difficult to do with today's equipment, so accepting from the phone company is not quite as secure as you think.
I also have worked on that field, but not that long. My experience is with the 5EEE only. It is precisely due to that work that I became interested in Linux ;-) Legislation in Spain is such that law insists calls are identified because it is needed to know whom to charge the phone call to, across different companies. It started when the government forced the only telco in existence here (Telefónica), in 1997, to accept indirect phone calls routed via another company (you prefix your call with a 3 digit number, and the call is routed via another telco for the long distance part). I too have seen those strange numbers, like 000000, in my terminal, but unfortunately, after I stopped working there, so I don't have inside knowledge on them. I believe they come via internet gateways. I know that with VoIP machines you have a lot of liberty, you can do anything with numbers and names and routings. It is up to you what you do or fake, which also kind of scares me. Of course, the typical is having a company with two or more sites, and routing the call, inside your network, to the POT network from the site closest to the destination, so that the call is charged local charges instead of long distance. - -- Cheers / Saludos, Carlos E. R. (from 12.1 x86_64 "Asparagus" at Telcontar) -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.18 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAlA4zaAACgkQIvFNjefEBxokjwCgyKIs65/ZmxnW+JgYI4lxkCXh 4j0An1brmrNNKpDT7lSmn2+ffyK6KdYm =o1HK -----END PGP SIGNATURE----- -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Carlos E. R. wrote:
I know that with VoIP machines you have a lot of liberty, you can do anything with numbers and names and routings. It is up to you what you do or fake, which also kind of scares me.
VoIP is completely independent of POTS, but maybe of course be connected to it or carry the traffic.
Of course, the typical is having a company with two or more sites, and routing the call, inside your network, to the POT network from the site closest to the destination, so that the call is charged local charges instead of long distance.
Even more typical is probably running all internal telephony as VoIP - it saves running separate telephone-wiring, and is easily connected to existing ethernet ditto. Also very popular is VoIP connections for individuals who have a fibre or cable connection any way. -- Per Jessen, Zürich (24.6°C) -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Per Jessen wrote:
Even more typical is probably running all internal telephony as VoIP - it saves running separate telephone-wiring, and is easily connected to existing ethernet ditto. Also very popular is VoIP connections for individuals who have a fibre or cable connection any way.
individual households. -- Per Jessen, Zürich (24.8°C) -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 2012-08-25 15:51, Per Jessen wrote:
Carlos E. R. wrote:
Even more typical is probably running all internal telephony as VoIP - it saves running separate telephone-wiring, and is easily connected to existing ethernet ditto. Also very popular is VoIP connections for individuals who have a fibre or cable connection any way.
Ah, yes, that's the obvious use, that's why I did not mention it. - -- Cheers / Saludos, Carlos E. R. (from 12.1 x86_64 "Asparagus" at Telcontar) -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.18 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAlA46ZIACgkQIvFNjefEBxo5ywCeLgNd7rp8BR2Whuw1EN5+XeZa RWUAoNzUZYmogqZPWa9VPcivY/zf8pzy =LMzd -----END PGP SIGNATURE----- -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Per Jessen wrote:
Carlos E. R. wrote:
I know that with VoIP machines you have a lot of liberty, you can do anything with numbers and names and routings. It is up to you what you do or fake, which also kind of scares me. VoIP is completely independent of POTS, but maybe of course be connected to it or carry the traffic.
Actually, my home "POTS" is delivered via VoIP. I have a terminal, sitting on my shelf, that connects between the cable TV network and my analog phones. There's a pair of regular RJ11 jacks on the back of it. I have also seen 8 port versions used in businesses.
Of course, the typical is having a company with two or more sites, and routing the call, inside your network, to the POT network from the site closest to the destination, so that the call is charged local charges instead of long distance. Even more typical is probably running all internal telephony as VoIP - it saves running separate telephone-wiring, and is easily connected to existing ethernet ditto. Also very popular is VoIP connections for individuals who have a fibre or cable connection any way.
I recently did some work for a major insurance company, where they are moving to VoIP for inter office calls. The local offices get their local trunks from the phone company via analog POTS, T1 or POTS/VoIP similar to what I have at home. There is no reason why they couldn't use local VoIP trunks directly, but they weren't yet doing that. -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
James Knott wrote:
Per Jessen wrote:
Carlos E. R. wrote:
I know that with VoIP machines you have a lot of liberty, you can do anything with numbers and names and routings. It is up to you what you do or fake, which also kind of scares me. VoIP is completely independent of POTS, but maybe of course be connected to it or carry the traffic.
Actually, my home "POTS" is delivered via VoIP. I have a terminal, sitting on my shelf, that connects between the cable TV network and my analog phones.
Yeah, POTS is rapidly going out of fashion along with broadband gaining territory.
I recently did some work for a major insurance company, where they are moving to VoIP for inter office calls. The local offices get their local trunks from the phone company via analog POTS, T1 or POTS/VoIP similar to what I have at home. There is no reason why they couldn't use local VoIP trunks directly, but they weren't yet doing that.
Sounds familiar. A couple of years ago, a large supermarket chain here in Switzerland switched to VoIP for all telephony to and between branches. -- Per Jessen, Zürich (16.5°C) -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Carlos E. R. wrote:
I also have worked on that field, but not that long. My experience is with the 5EEE only. It is precisely due to that work that I became interested in Linux;-)
I have worked with a wide variety of systems, though not on the big phone switches.
Legislation in Spain is such that law insists calls are identified because it is needed to know whom to charge the phone call to, across different companies. It started when the government forced the only telco in existence here (Telefónica), in 1997, to accept indirect phone calls routed via another company (you prefix your call with a 3 digit number, and the call is routed via another telco for the long distance part).
The phone company can still identify the phone trunk a call came in to them on and use that for billing. However, they have no idea what may be connected to that trunk at the customer's end. It could be a local PBX or it could be extended just about anywhere in the world, by a variety of means.
I too have seen those strange numbers, like 000000, in my terminal, but unfortunately, after I stopped working there, so I don't have inside knowledge on them. I believe they come via internet gateways.
I know that with VoIP machines you have a lot of liberty, you can do anything with numbers and names and routings. It is up to you what you do or fake, which also kind of scares me.
Of course, the typical is having a company with two or more sites, and routing the call, inside your network, to the POT network from the site closest to the destination, so that the call is charged local charges instead of long distance.
That has been quite common for years. Also, the old "TDM" systems were also quite flexible, as I have mentioned, including giving whatever number to a phone. It's just cheaper and easier with SIP. -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 2012-08-25 21:42, James Knott wrote:
Carlos E. R. wrote:
I also have worked on that field, but not that long. My experience is with the 5EEE only. It is precisely due to that work that I became interested in Linux;-)
I have worked with a wide variety of systems, though not on the big phone switches.
Legislation in Spain is such that law insists calls are identified because it is needed to know whom to charge the phone call to, across different companies. It started when the government forced the only telco in existence here (Telefónica), in 1997, to accept indirect phone calls routed via another company (you prefix your call with a 3 digit number, and the call is routed via another telco for the long distance part).
The phone company can still identify the phone trunk a call came in to them on and use that for billing. However, they have no idea what may be connected to that trunk at the customer's end. It could be a local PBX or it could be extended just about anywhere in the world, by a variety of means.
The snag is that the identification has to be carried forward to the next telco receiving the phone call, which is why caller ID is mandatory here. What I do not know is to what extent the PBX can forward an ID or not, because although my knowledge is becoming fuzzier with time, I think that the ID was assigned on the big exchanges, not on the PBX. You say you can do that on the PBX, but I simply do not know if it is possible here or not. - -- Cheers / Saludos, Carlos E. R. (from 12.1 x86_64 "Asparagus" at Telcontar) -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.18 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAlA5PEsACgkQIvFNjefEBxrTYACfWF0z+0g2Q1F9Rif3l34d8A4u FxYAnjFRqTyJ0lvZglXnvOKkBWWE7eDw =QvAe -----END PGP SIGNATURE----- -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
Carlos E. R. wrote:
that for billing. However, they have no idea what may be connected to that trunk at the customer's end. It could be a local PBX or it could be extended just about anywhere in the world, by a variety of means. The snag is that the identification has to be carried forward to the next telco receiving the
The phone company can still identify the phone trunk a call came in to them on and use phone call, which is why caller ID is mandatory here. What I do not know is to what extent the PBX can forward an ID or not, because although my knowledge is becoming fuzzier with time, I think that the ID was assigned on the big exchanges, not on the PBX. You say you can do that on the PBX, but I simply do not know if it is possible here or not.
That number is carried over SS7 on the POTS network and a similar method exists for SIP. If a carrier does not pass on that number, they're violating the SS7 specs. -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 2012-08-25 23:48, James Knott wrote:
That number is carried over SS7 on the POTS network and a similar method exists for SIP. If a carrier does not pass on that number, they're violating the SS7 specs.
Pre 1997 the telco in Spain did not. I worked for one of the newcommer telcos, and one of the problems was that some calls did not carry the ID via SS7 signaling. I was in the department that received those problems, we found out that. Of course, in most of those occasions, it happened that the caller was on one of the few analog switches remaining (rotary?), so caller ID was impossible. Telefónica had to supply us with a list of all the phone ranges for which caller ID was impossible, so that our telemarketers did not offer service to them. - -- Cheers / Saludos, Carlos E. R. (from 12.1 x86_64 "Asparagus" at Telcontar) -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.18 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAlA5cAMACgkQIvFNjefEBxocCwCdH+s6QALAIX9D28Z97w5TWSIL JcwAoNs0AoFkgazdeSwjFzTZttozwGvP =C9fc -----END PGP SIGNATURE----- -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org
participants (7)
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Carlos E. R. -
Dsant -
Felix Miata -
Hans Witvliet -
James Knott -
Per Jessen -
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