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<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>
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