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| author | arf20 <aruizfernandez05@gmail.com> | 2025-08-26 23:55:30 +0200 |
|---|---|---|
| committer | arf20 <aruizfernandez05@gmail.com> | 2025-08-26 23:55:30 +0200 |
| commit | 71a0e862ded2041b23f2b63603c85203434b3535 (patch) | |
| tree | 1e5f006ce948d0c97f2248fd66858f8858a52a8f | |
| parent | 5ba722a6bd1a291a3c8426191c5f73a778ad184b (diff) | |
| download | arfnet2-web-71a0e862ded2041b23f2b63603c85203434b3535.tar.gz arfnet2-web-71a0e862ded2041b23f2b63603c85203434b3535.zip | |
Contact add, cv, projects
| -rw-r--r-- | contact.html | 3 | ||||
| -rw-r--r-- | cv.html | 11 | ||||
| -rw-r--r-- | projects/telephony/index.html | 180 |
3 files changed, 190 insertions, 4 deletions
diff --git a/contact.html b/contact.html index bf9f175..b2b64b1 100644 --- a/contact.html +++ b/contact.html @@ -18,7 +18,8 @@ <ul> <li>e-mail: arf20@arf20.com, webmaster@arf20.com, dmca@arf20.com, arfnet@proton.me, aruizfernandez05@gmail.com <a href="/arf20.ecc.asc">PGP</a></li> - <li>Phone: +34 637 63 21 36</li> + <li>Mobile phone: +34 637 63 21 36</li> + <li>Office operator: +34 868 92 55 54 (press 1 for IVR, press 2 for V.34 BBS)</li> <li>Brandmeister DMR: 2149160 (EA5JGX)</li> <li><a href="https://github.com/arf20">GitHub</a></li> <li><a href="https://www.linkedin.com/in/%C3%A1ngel-ruiz-fernandez-562470229/">LinkedIn</a></li> @@ -76,7 +76,12 @@ <h2>Certifications and licenses</h2> <ul> - <li><b>Jan 2024: </b>Hurricane Electric IPv6 Certification</li> + <li><b>Aug 2025: </b>RIPE NCC IPv6 Certification<br> + <div data-iframe-width="150" data-iframe-height="270" data-share-badge-id="67df9682-9cc4-4198-8c90-6a9d6a2d9c37" data-share-badge-host="https://www.credly.com"></div><script type="text/javascript" async src="https://cdn.credly.com/assets/utilities/embed.js"></script> + </li> + <li><b>Jan 2024: </b>Hurricane Electric IPv6 Certification<br> + <img src="https://ipv6.he.net/certification/create_badge.php?pass_name=arf20&badge=2" style="border: 0; width: 250px; height: 194px" alt="IPv6 Certification Badge for arf20"></img> + </li> <li><b>2022: </b>CEPT Radioamateur Licence (EA5JGX)</li> </ul> @@ -107,7 +112,7 @@ <li><b>Build systems:</b> CMake, Make</li> <li><b>Version control systems:</b> Git</li> <li><b>Operating Systems:</b> GNU/Linux, *BSD, Solaris, IRIX, AIX, plan9, MS-DOS, Windows</li> - <li><b>OS building:</b> Kernel development, embedding Busybox/Linux</li> + <li><b>OS building:</b> Kernel development, embedding Busybox/Linux</li> <li><b>Server and hypervisor software:</b> Proxmox VE, nginx, (all manner, see ARFNET)</li> <li><b>Networking:</b> Cisco, Mikrotik, pf/OPNSense, DELL, SONiC, P4lang</li> </ul> @@ -118,7 +123,7 @@ <li><b>Multimedia:</b> Video recording, editing and streaming</li> <li>Music theory and interpretation</li> </ul> - </div> + </div> </div> </main> </body> 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 @@ +<!DOCTYPE html> +<html> + <head> + <meta charset="UTF-8"> + <link rel="stylesheet" type="text/css" href="/style.css"> + <title>ARFNET</title> + <style> + .sect { + margin-left: 20px; + } + + .pics { + display: inline-block; + } + </style> + </head> + + <body> + <header><a href="/"> + <img src="/arfnet_logo.png" width="64"><span class="title"><strong>ARFNET</strong></span> + </a></header> + <hr> + <h2><a href="../index.html">Projects</a></h2> + <h2>telephony</h2> + + + <h3>Intro</h3> + + <p> + 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. + </p> + + <h3>Dial-Up backstory</h3> + + <p> + 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. + </p> + + <h3>Meta and first steps</h3> + <p> + 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. + </p> + + <p> + 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. + </p> + + <p> + 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. + </p> + + <p> + 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. + </p> + + <h3>ATAs</h3> + + <p> + 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. + </p> + + <p> + 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. + </p> + + <h3>IP phones</h3> + + <p> + 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. + </p> + + + <h4>SEPfiles</h4> + <p> + 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<MAC>.cnf.xml, <MAC> 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. + </p> + + <h4>Cisco 7941</h4> + <p> + 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 + </p> + + <h4>Cisco Unified Communications Manager</h4> + <p> + 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. + </p> + + <h4>Cisco 3911</h4> + <p> + 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. + </p> + + <h3>Trunks</h3> + + <p> + 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. + </p> + + <p> + 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. + </p> + + <p> + 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!!! + </p> + + <div class="pics"> + <img src="cam.jpg" width=49%> + <img src="vcr.jpg" width=49%> + </div> + + + <div class="pics"> + <img src="in1.jpg" width=32%> + <img src="in2.jpg" width=32%> + <img src="bat.jpg" width=32%> + </div> + </body> +</html> + |