ARFNET

From 71a0e862ded2041b23f2b63603c85203434b3535 Mon Sep 17 00:00:00 2001 From: arf20 Date: Tue, 26 Aug 2025 23:55:30 +0200 Subject: Contact add, cv, projects --- projects/telephony/index.html | 180 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 180 insertions(+) create mode 100644 projects/telephony/index.html (limited to 'projects') diff --git a/projects/telephony/index.html b/projects/telephony/index.html new file mode 100644 index 0000000..e0f576f --- /dev/null +++ b/projects/telephony/index.html @@ -0,0 +1,180 @@ + + + + + + ARFNET + + + + +

ARFNET +

Projects

telephony

+ + +

Intro

+ +

+ We've all had landlines at home, but what first got me interested in telephony, + was a laptop with an internal modem which I got to play with, and a InfoVia Plus information card + with my cousin, which in my failing memory I think is whom I owe my love for retrocomputing and by extension + everything technologically dated, although my tasted may shift towards professional and enterprise hardware. +

+ +

Dial-Up backstory

+ +

+ The InfoVía Plus thing is rather interesting. As far as I know its a dial-up service created by Telefónica in 1999 + for the price of the phone call, which it was still a monopoly of Telefonica. Of course, I wasn't alive to + see it in its time but rather, much much later in the mid 2010s, by which time the service was still very much + in service, although I'd imagine with few users given that the DSL, VDSL and mostly ADSL broadband services and later + HFC of other, emerging ISPs took much of the clientele of Telefónica itself. I'd use a Fujitso Siemens laptop from + my father with an internal modem, and using Windows XP dial my town's node and connect with if I remember correctly + something like infovia@123 via PPP, and get 33.8 or 48kbps sometimes of internet connectivity. With which you could + navigate good ol' sites like sdf.org, frogsearch, stoff.pl? And of course ARFNET. At the time of writing, I made a + small survey and I could only find one node still running and answering, the Toledo one. +

+ +

Meta and first steps

+ Consider this document not a writeup, but something like an evolving tale, which starts with the online + finding and holy installation of sacred Asterisk Open Source PBX and telephony toolkit; followed by the interpretation + of the holy configuration scriptures, and sacred online documentation. As you may expect, my 17-year-old + ass got bored of not getting that nonsense to work pretty quick with the ancient, arcane chan_sip module of what is now + Debian oldstable's package. +

+ +

+ Thankfully, the story didn't end there. Thanks to the inspiration of SDF and its people, I pressed on + but in reality it took several attempts, fixations, frustrations and abandonment over the course of + several years until today, when I can finally say I did it. I here described how I did it do the silly things + a telephone system does. +

+ +

+ After finding how useless the chan_sip module is I decided to just build a modern version of Asterisk from source (20 at the time) + I followed many tutorials that used the wizard configuration but I never got that to work either. So I asked for help + at some friend's nerdy telephony (among other things) discord and set up the chan_pjsip module, which actually did work, with + softphone clients (MicroSIP on windows). I didn't go much further than that, and left it like that. +

+ +

+ Later I went looking for real hardware to use with my newly setup PBX, like ATAs and IP phones, + but I didn't have any purchasing power at the time, until someday at school I was given two Cisco 3911 IP phones from school, + from my legend, Rosendo, the guy himself. However, I immediately discovered Cisco and its shenanigans. + A time of despair. +

+ +

ATAs

+ +

+ Another time, at a local morrocan flea market, I found a god damm Linksys PAP2T ATA somehow??? You never know what you find + in those places. Old vidicon cameras? Rotary phones? Yamaha keyboards? A bit of a time machine these markets are, + from I guess the fleeing old rich population of the south, since the increasing collapse of the economy and + opportunities, too bad! Their time is over. With the newly acquired ATA for a whoopping 3€, I figured out + from the manual that it had a password, and I had to reset it with a god forsaken code from an old forum, + dialing with a god knows brand 2000s cheap small gondola-style phone I got from a friend. I then proceeded to set it + up at home with its ancient webfig, and it pretty much just worked. I was able to make calls and all. +

+ +

+ I then got another ATA bought from another friend, a Grandstream HT702, hoping that it would + work with rotary phones (haven't tested that yet, but the webfig makes no mention to pulse dialing) + and I set it up at my student place, to register remotely over the internet (since I had no router + capable of doing site-to-site tunneling (asaide from that damm Mikrotik that I could + never get to work)) and for the first time, I could make a real phone-to-phone call, over the internet + using the PBX, to a remote place (my parent's) so it finally could have purpose. Note: for the + purposes of the story, I am telling it in a rather linear fashion trying to make it make sense + but in reality some of these events are very mixed and intertwined with much more failure than success. +

+ +

IP phones

+ +

+ At a different unspecified time, rather parallel, I got another, Cisco IP phone, a 7941, from a friend + and got told this one COULD work with Asterisk if I flashed it with its SIP firmware (Asterisk does not ship with SCCP or skippy support). + So anyway, I bricked it in the process. And it stayed so until recently where I had the brilliant idea of + flashing its original firmware back, and that got it unbricked. The brick mode was rather silly, + because it would stay in flashing mode whenever it booted, try to download firmware, fail and reboot again. + The reason I got back to the original SCCP firmware is that I got told about the existance of the out of tree chan_sccp asterisk module, which is actually really really good. +

+ + +

SEPfiles

+ Cisco IP phones automatically get their IP addresses via DHCP, and from it take a TFTP server, + from which they try to download among other things, a file named SEP.cnf.xml, being + the phone's MAC address in uppercase hex without separators. This is the so called sepfile and + it describes the phone's configuration, including but not limited to its time server, Cisco UCM server, + timezones, directory URLs, and stuff. I was able to find a sepfile for the 7941 but not for the 3911 phones. + Which launched me on a very long detour that took around 51 years. +

+ +

Cisco 7941

+ So, having burnt the SCCP firmware back again, I installed chan_sccp from source to the asterisk and tested the module. + And with a SEP file for the 7941 I got off the internet, I modified it and pointed the phone to the Asterisk running chan_sccp masking as + CUCM, and allll the SCCP features just work out of the box! I'd even dare to say that the experience is better than chan_pjsip. Awesome +

+ +

Cisco Unified Communications Manager

+ For the purposes of making the 3911s work, I got a CUCM ISO from the internet and spent too much + time trying to virtualise it in KVM, first installing it on VMWare and then moving it over to KVM trying to + virtualise the devices it expects. After getting it to boot, I proceeded to set up all the required objects + and configuration to add a phone device, the 3911. +

+ +

Cisco 3911

+ When I confirmed that the 3911 indeed registers with CUCM, I used a tftp client to fetch its SEP file + and modified it to my needs to point it to asterisk, having added the corresponding endpoint, aor, auth and stuff. + And to my surprise, ancient Cisco SIP works with Asterisk. Damm. No fancy features though. +

+ +

Trunks

+ +

+ The original inspiration in the project was to have a PBX with an SDF SIP trunk. I think I did get it to work a while ago, but + in the meantime of trying stuff I guess it broke, but I recently fixed it! Copying another working configuration from a posterior trunk. +

+ +

+ After having the ATAs working roughly as they are now, someone told me about TandmX, and it was simple enough to join, + being an IAX2 trunk, very simple and stable, and met a whole lot of people in their discord. +

+ +

+ A bunch of years after hiring AvanzaFibra and being told by someone somewhere that I could ask for the SIP credentials for my fixed + phone line that comes off the router (lines lying about being VoIP) which my ISP formally does not offer VoIP lines; I insisted over the phone and raised + a ticket 3 times until eventually I could talk to an actual tech that knew what the hell I was talking about. He had to generate new creds for my line (the ISP's CPE ATA would stop working (thats fine)) + and he would have it sent to my email, so they did. It registered correctly but after a minute the trunk would stop working, originating or terminating calls for some reason. The reason was keepalive I think. + So I made Asterisk do keepalives (SIP OPTIONS) every 58 seconds). Then applied the same config to the SDF trunk apparently. Now I have PSTN calls!!! +

+ +

+ + +

+ + + -- cgit v1.2.3