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Thursday, July 13 2006, 02:42

Junghanns.net quad GSM PCI arrived!

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oh, what nice day: the Junghanns.net quad GSM PCI arrived!

No, i'm not swimming in money, my company bought it, not me personally ;) It's ~ 2k EUR, a little expensive for a toy - but worth it's money in a production environment (or so i hope - time will tell).

The moment i heard the package arrived at the office, i went to the mobile phone store right around the corner of where i live and asked for a GSM antenna - ha, ha, like they sell those. "Damn!" i thought, knowing that running the device without an antenna will most likely not work, and may even break it. So i went to the office without a GSM antenna, and opened up the box:


Tadaa! 4 small, but sufficient antennas were included in the package! The card is robust and well-manufactured, the SIM card holders are on the back, the topmost one being connected to the topmost antenna connector, as can be seen in the jumper settings sheet.

For the ztgsm module to work, you'll need a bristuffed asterisk, forget about the stable 0.2 branch (it's asterisk 1.0, yuck!), get the latest version of 0.3 instead, it's asterisk 1.2.9.1 and zaptel 1.2.6, so it's up-to-date. It's a little uncomfortable to be forced to switch to the bristuffed asterisk, but maybe the ztgsm driver will find it's way into mainline zaptel. Trying to use the ztgsm with mainline asterisk and zaptel failed for me because of a missing libgsm-like library inlcuded in bristuff... ztcfg already failed to load it's configuration.

once ./install.sh of bristuff finished, copy over the ztgsm/zaptel.conf.unoGSM, ztgsm/zaptel.conf.duoGSM or ztgsm/zaptel.conf.quadGSM to /etc/zaptel.conf (assuming you didn't already have device configs in there, then just append them, of course...), depending on which model you own. If you have multiple GSM card with only part of the sims installed, you don't need to alter the settings here, you can do that in zapata.conf, which is also located in ztgsm, again in the zapata.conf.unoGSM, zapata.conf.duoGSM and zapata.conf.quadGSM versions. Copy the according one to /etc/asterisk/zapata.conf (again, don't overwrite if you already configured a Zap device, stupid!) and edit the file. Enable as many sections as you have sim cards installed for (or less ;), assuming you installed them from slot A-D in row - I didn't test other configurations, it's pretty time-consuming un-screwing all the antennas to get the PCI card out again to swap sim cards, you know, and it's very dusty down there under my desk... oh well, back on topic. Make sure the pin codes are correct...

With /etc/zaptel.conf and /etc/asterisk/zapata.conf configured, it's time to modprobe ztgsm. this took about 30 - 40 seconds here...

ztgsm: iomem at d5204000 size 8192
ztgsm: ioport size 256
ztgsm: Junghanns.NET quadGSM card configured at io port a000 IRQ 17 io mem f8ae8000 HZ 250 CardID 0
ztgsm: Powering up all spans... done.
ztgsm: VERSION aa32
ztgsm: 1 multiGSM card(s) in this box, 4 GSM spans total.

the leds on the rear of the card finally light up and show some of their colors! then, ztcfg -v and asterisk -vvvvvvvc. If you entered a wrong PIN code in zapata.conf, you will pay for your education now. On the console you should note some debug messages about the GSM spans connecting to the network. If you now issue a Dial(Zap/(#channelr from zapata.conf)/012345) the device should call the phone nr.

In my case, though, it said it "unregistered from the network" instead, and i heard a blipblipblip sound on the snom phone i placed the call with - after call termination, the following error was displayed: 

chan_zap.c: GSM: 1 !+CME ERROR: operation not allowed!

Oh jesus, i thought, how am i going to debug this? After some digging and googling for the CME Error codes, i found out that the answer is easy: screw out the card again, and insert the SIM card into a usual cellphone. voila - the card was not valid anymore, because someone at my company re-ordered the same card and didn't tell me. "Network registration failed" was what the Nokia cellphone told me after it accepted the PIN code. So i tried another SIM card, and dialing out and in worked!

Unfortunately it's not possible to suffix extensions to the telephone number on gsm calls, so it seems you'll always have to use either the "s" extension or the individual telephone number of the SIM card recieving the call on incoming calls in the from-gsm context (if you left the default in zapata.conf, that is).

I'm still a bit puzzled about how smsq, the queuing tool for app_sms works, but i'll figure that out once i have the correct SIM card.

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Saturday, July 8 2006, 19:25

First VoIP SIP Phone sighted at consumer electronics store!

By fake - Permalink



just yesterday i saw the first SIP VoIP phone (not Skype, USB, or all that junk) at "Saturn", a german/austrian consumer electronics store with many shops around those two countries. The poor guys filling up the shelves are obviously confused by the merger of IP and telephony, so they just put it in line with all the other DECT phones: The Siemens C450IP. That decision isn't totally braindamaged, as the phone has the capability for standard analogue telephony, too.

It's 99 EUR, and i'm probably going to get one for my employer. Expect a review..

It took quite some research to find out that phone is using SIP and is not bound to any Skype or Windows Live Messenger junk like the M34 USB DECT Dongle for Skype Siemens sells, too, or the stuff Philips is selling as "VoIP phones".

Surprisingly, Siemens is using libosip in those phones. According to the LGPL libosip is licensed under, you can download the source used, and a toolchain to build firmware images at a special Siemens Developer page. The .tar.gz didn't work for me with tar xfvz for some reason, but gunzip and tar xfv separately work.

It's cool Siemens is playing by the rules!

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Thursday, January 12 2006, 19:50

Netgear TA612V user report

By fake - Permalink



i bought a netgear ta612v a while ago, and i totally regret it. the device is called a 'voice over ip telephony adapter', and is sold in austria and germany, afaik. the box doesn't say it, but the device is bound to work ONLY with sipgate.de/sipgate.at 's services.

Once you connected the device (which only works if it gets to do the dial-in via pppoe/pptp, yuck!), you visit a web page at sipgate, where you enter the device's MAC address. A configuration file for your associated account is then sent to your device automagically. you can't disable this service - neither can you configure *any* of the important voice over ip settings. netgear promised to ship a drilled-open firmware a time ago, but as of yet they didn't, and several polite requests and later rants to the support page (i'm a customer, damn it!) didn't help.

so i let the white box do my dial-in and routing, feeling *very* uncomfortable. though, i have to admit, the routing part worked pretty well, the box only froze once in about 6 months of uptime.

another thing is the availability of the SIP lines. the phone line worked, say, 70% of the time, and i had to do *very* dirty tricks using DISA (direct inward system access) on my main asterisk box to get it working the way i wanted. also, i had to register a second account at sipgate just to get my regular sip calls forwared to the connected phone. getting every DTMF digit twice (or not at all) at the recieving end made things even worse.

so that's it for the netgear ta612v. apart from it being a proprietary dsl dialup router you can rely on, it sucks big time. finally doing the next order of voip hardware for my company over at beronet's online shop, i ordered an IAXy for my private use, and gave the netgear junk away.

stay tuned for a report, the shipment should arrive next week ;-)

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Thursday, February 17 2005, 20:35

something to call 'home', calling home

By fake - Permalink

after thinking into it for some time, i noticed that sharing a flat with clifford - or anyone, for that matter - is something i'm not so keen about after all. how am i supposed to call something my 'home' when it's not my home? it would be our home, not mine. i don't know what this is all about, i can't really argue about it logically, it's just a necessity i feel and i can't withdraw by logic. so i looked around the net on monday, made a few calls, and ended up looking at 3 flats on monday evening, one of which i have now complied to rent. it's amazing, totally into the flat situation in munich, i was stunned by the offers - 3 flats near the main street in the district i work in for a really affordable price - empty! weird.

my 'home' (i like the sound of that!) will be 61 square meters large, with an inner height of 3.60 meters, consists of 2 rooms and - tadaa - a bathtub! i never had an own bathtub before, just stand-and-hurry showers. it's going to be great. i hope ;)

i tried the shopping thing again last friday, after that complete failure on thursday, and i searched and called people and spoke to morons, until i finally found a company that would sell me a snom phone right away. i hurried there, picked it up, and finally got to deploy an asterisk setup along the VPN on the weekend. it's still not perfect, especially my friends are having problems with their Softphones and / or sound equipment (holli is in crystal-clear quality, but much too low in volume, whereas hannes is loud enough, but sounds like he's talking through a tunnel with many rats in it).

the trick to achive a low delay is, that i use IAX2 'outside' the vpn. though it's not encrypted (which can be done with srtp), it's possible to set the tcp lowdelay/etc flags in a sane manner, plus the overhead of encryption that caused some delay and would even eventually let the stream congest on high load (i.e. lousy compression) is gone, too.

if you wanna call me, my freeworlddialup-account is 612990, so you can also set up an asterisk and use iax2 to the outside if you've got problems with nat traversal, or you could directly call me via SIP at sip:fake@f4k3.net. let me know if it doesn't work ;)

the phone is really amazing, i'm stunned by the possibilities i have. a tricky thing was to set up the message waiting indicator (MWI) on the phone to react when there's a message on a remote server (as you may have guessed, storing my voicemail on clifford's soekris router is not an option ;). i tricked around this by entering the phone as a peer in asterisk's sip.conf on the server i store the voicemail, it's reachable via the VPN from there. i had to fake the fromdomain= and fromuser= lines in the peer section, so the phone calls the voicemailbox application through the soekris' mini-asterisk again.

i also got the cisco voip ringtone (or something that sounds exactly like it, to be correct) i got used to so much thanks to 24 - the phone can stream any 8000Hz 16bit mono wav file off an http server, it's set via a sip header ;-)

there's a lot of information about how to get the most out of the phone with asterisk here.

i'll be driving to germany tomorrow, holli aka blacksun is celebrating his birthday with a classical lan-party, so i may have something to write on sunday (being utterly bored by lan-party-games *g*).

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