News items for tag asterisk - Koos van den Hout

There is a museum in Seattle called the Connections Museum and it is on my "If I ever visit that part of the world" list. The reason I found it because one of the volunteers likes to make videos for youtube about the equipment in the museum and the youtube suggestions are on to me.

with an interest in phone phreaking in my history this is a very interesting channel. They recently had a video on how blue boxing *actually* worked, including a demonstration of how the switch actually responds to the blue box tones. This made me go "oh now I get it" for details on blue boxing.

In the latest youtube video is an explanation that they run asterisk as one way of connecting all their historic phone exchanges. The historic phone exchanges are also connected using direct interconnects. Video announced in In case you haven't seen the latest bit of ridiculous hacking ;) - Connections Museum on Twitter. Video at Is this the world's oldest Linux peripheral? - Connections Museum

If I understand the remark about asterisk and Collectors' Net / Phreak Net correctly it should be possible to dial into the old exchanges at the museum from either of those networks.

From 2008 to 2013 I played for a while on the Collectors' Net to test my asterisk experiments but when I got less interested and reduced my phone setup at home to a simple voip base again I stopped being a member of Collectors' Net. Maybe I should get back on one of those networks and get something going again! It would be awesome to have an option to dial into the old hardware at the Connections Museum and actually end up in a phone switch from 1923 using a VoIP phone on my side. Or dig up a pulse-dial capable ATA and dial in using the original T65 rotary phone.

This weekend was about the last weekend that I could have access to Ubuntu 10.04 LTS for upgrading my homeserver greenblatt from Ubuntu 8.04 to Ubuntu 10.04 which will run out of support this month, making it unavailable as an upgrade path. I postponed this for way too long because I expected a lot of work fixing things, especially asterisk which runs our home phone system.

The solution to asterisk was simple: I disabled it and reprogrammed a Gigaset C450IP base to deal with our VoIP provider directly. After all the upgrading is done I'll go fix things with the OpenVox ISDN card in a test machine and when all is stable again I will move things to production.

For the rest it was a matter of typing 'do-release-upgrade' and hope for the best. Reconfiguring packages took the longest and the biggest issue was that the upgrade from Postgres 8.3 to Postgres 8.4 wasn't done automatically but I had to do this myself using the hints from How tu [sic] upgrade Postgresql 8.3 database file to 8.4 - stackoverflow.

The documentation says to do this beforehand but Postgresql 8.4 isn't available before the upgrade and it took a bit of fiddling to have Postgresql 8.3 available after the upgrade. But then pg_upgradecluster ran. It complained about a few tables and the end result I noticed was that those tables were dropped out of the upgrade completely. I re-enabled Postgresql 8.3 and migrated those databases or tables using pg_dump. Not a complete smooth upgrade but I think it went ok.

I dug up all the tools needed to test the isdn setup in the test server: an old sitecom ISDN card with HFC-S chipset, an ISDN cross cable, a fritzbox with external S0 bus and an analog phone set.

It took me a while to get all 3 channels in the ISDN card active in Asterisk, I 'missed' the fact that the oslec echo canceller wasn't loaded due to a module versioning problem. At first it showed:

root@metcalfe:~# lsdahdi
### Span  1: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: HRtimer) 1" (MASTER)
### Span  2: ZTHFC1 "HFC-S PCI A ISDN card 0 [NT] " AMI/CCS
  1 BRI        Clear       (In use)
  2 BRI
  3 BRI

The switch from oslec to mg2 fixed things:

root@metcalfe:~# lsdahdi
### Span  1: ZTHFC1 "HFC-S PCI A ISDN card 0 [NT] " AMI/CCS
  1 BRI        Clear       (In use) (EC: MG2 - INACTIVE)
  2 BRI        Clear       (In use) (EC: MG2 - INACTIVE)
  3 BRI        Hardware-assisted HDLC  (In use)
### Span  2: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: HRtimer) 1" (MASTER)

But whatever I tried: no dialtone. Time to also hook up a SIP phone to initiate calls the other way.

An interesting twist in the microsoft support scam calls: 7 call attempts within 2 seconds. So asterisk can only forward the first 2 calls to the isdn phones in the house and the next five go to voicemail instantly.

sqlite> select src,start,answer,end from cdr where .. order by start;
0016077329064|2013-08-14 13:12:16|2013-08-14 13:12:47|2013-08-14 13:12:53
0016308599364|2013-08-14 13:12:16|2013-08-14 13:12:47|2013-08-14 13:12:52
0015852439807|2013-08-14 13:12:17|2013-08-14 13:12:18|2013-08-14 13:12:40
0017187455293|2013-08-14 13:12:17|2013-08-14 13:12:18|2013-08-14 13:12:23
0016073249764|2013-08-14 13:12:17|2013-08-14 13:12:17|2013-08-14 13:12:23
0016073639777|2013-08-14 13:12:17|2013-08-14 13:12:17|2013-08-14 13:12:22
0016265749227|2013-08-14 13:12:17|2013-08-14 13:12:17|2013-08-14 13:12:22

Bug in the call handling on the scamming side? Nobody was available to be scammed.

De laatste tariefsverhoging van KPN voor het vaste net, met als excuus uitleg:

Steeds minder mensen maken gebruik van de vaste telefoon waardoor de kosten per gebruiker stijgen. Daarom zijn wij genoodzaakt om enkele tarieven te verhogen.

maakte het tijd om een portering aan te vragen naar een VoIP aanbieder. Per maand betaal ik voor ISDN1 met belvrij weekend straks EUR 23.43 (was EUR 21.24).

ISDN is een hele mooie technologie, en ik vind het jammer om er minder mee te doen, maar de tarieven zijn dusdanig dat het de moeite is om met VoIP te gaan bellen. SIP geeft me zeker bij een provider die het goed implementeert en weet dat er mensen asterisk gebruiken vergelijkbare signalering en informatie. Een Internet aansluiting willen we toch altijd wel hebben dus de telefonie kan daar ook overheen.

Upside of using asterisk for the home telephony: it's quite easy to browse the call detail records and do some calculations on them. So when our fixed line provider came with yet another price increase it was time to shop around for better options. And comparing rates is a lot easier when you have an exact log of how many calls for how long to which destinations were made in the previous months.

I'll miss the high level of control and call-progress indication ISDN offers, but prices for SIP accounts are a lot better and call-progress for SIP is comparable to ISDN.

Interesting new twist between all the attempts to reach Palestinian cell phone numbers: one try to reach the US embassy in The Hague. I guess someone attempting to abuse my SIP server thought maybe just international calls are blocked and used a number which is easy to find from abroad. Incoming audio was recorded, but it's a recording of pure silence.

New attempts to get a call out via the unauthenticated SIP context on my asterisk testserver, this time attempts to get connected to +96277xxxxxxx numbers. These are Orange Jordan mobile phone numbers. The originating IPs are also in Jordan so I guess the attacker is waiting for the mobile phone to ring.

Vandaag een PTT Ericsson model 51 DTMF geleerd met behulp van de DTMF omzetter van picbasic.nl met ondersteuning voor PTT Ericsson model 51. Een wandtoestel wat een ontwerp is uit 1951. Het toestel wat we onderhanden hadden had een productiedatum 'VII 1964', vermoedelijk dus juli 1964. De PTT W65 is de opvolger hiervan.

Na het werkend krijgen van de DTMF omzetter hebben we het toestel aangesloten op een Cisco ATA 186 die gekoppeld is met de asterisk testcentrale zodat we de sprekende klok in asterisk en het weerbericht in asterisk konden bellen. En natuurlijk de telefoon laten rinkelen!

I just found Intercept Service with Jane Barbe where ElmerCat has put a lot of time and energy into saving, splitting and digitizing phone phreaking recordings. My first thought was to take the Jane Barbe recordings and set up a few intercepts of my own. Maybe for playing with the people who try to break in to my asterisk testserver or (more constructive) to set up a Jane Barbe intercept service which can be used on Collectors*Net.
Found (unsurprisingly) via "1000 Abstract Machines" ... and a New Generation of Phone Phreaks? - The History Of Phone Phreaking.

Update: Ok, using the 'Jane Barbe' digits in Asterisk isn't very hard. Download the .mp3 files from soundcloud and convert them to the asterisk .gsm format:

$ mkdir janebarbe
$ sox JB-0-neutral.mp3 -r 8000 -c 1 janebarbe/0.gsm

..

$ sox JB-is-not-in-service.mp3 -r 8000 -c 1 janebarbe/is-not-in-service.gsm 
$ sox JB-the-number-you-have-reached.mp3 -r 8000 -c 1 janebarbe/the-number-you-have-reached.gsm

And put that entire janebarbe directory in the directory where asterisk expects the digit files for language 'janebarbe' which is /usr/share/asterisk/sounds/digits/janebarbe/ in the 'old' directory structure and /usr/share/asterisk/sounds/janebarbe/digits/ in the 'new' directory structure. Look at Asterisk multi-language - voip-info.org for details on directory structures.

Using the digits is now simple, a test:

exten => s,1,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(digits/janebarbe/the-number-you-have-reached)
exten => s,n,Set(CHANNEL(language)=janebarbe)
exten => s,n,SayDigits(1234567890)
exten => s,n,Playback(digits/janebarbe/is-not-in-service)
exten => s,n,Hangup

Will have Jane Barbe telling you what you expect. This can be used as an invalid-number intercept.