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Linux-Call-Router
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What we do
Again we will provide external telephony connection to the congress by linking the DECT base station of Eventphone (POC) to Versatel via ISDN over IP tunnel. You will be able to dial to the national fixed and mobile telephone network for free, except value added numbers and emergency numbers. Just use your DECT phone as you would do it at home.
The connection is realized with Linux-Call-Router, a software based PBX, specially designed for ISDN and low latency. In conjunction with mISDN driver, it connects to E1 links (S2M), S0 links, virtual IP links, as well as to Asterisk PBX using chan_lcr channel driver for SIP and other protocols.
Use your DECT Telephone
The outdial has changed this year, so read this: To dial to a German number, just dial the area code and number as you would do elsewhere in Germany. For Berlin you also need to dial the area code! Example: 030 1234567 (dialing to Berlin) For our foreign visitors: In Germany the area code always starts with the '0', so if you like to dial an German number like +49-xxx-yyyy, then you remove the country code (49) and add the '0' for accessing the area: 0-xxx-yyyy.
To dial inside the congress, you need to dial the four digits code of the phone you like to reach. (Don't add 0 or any other prefix in front of it.)
To dial to the congress from German telephone network, dial 0461-5056623-xxxx, where xxxx is your phone number at the congress. To dial from outside Germany, dial your international access code and then +49-461-5056623-xxxx.
Special Projects this year
Interoperability with OpenBSC
This year I spent some time with improving and writing an application interface for the GSM project OpenBSC. This interface allows Linux-Call-Router to use OpenBSC and a Base Station to build a GSM network with access to ISDN phones and the public fixed network. We like to test and improve stability of both GSM and Linux-Call-Router.
Blueboxing
You say: "Blueboxing does not work anymore!". This is not true. I have a little supprise. Bring your blue box (software) for some retro phun. Don't worry - it will not be illegal.
Reducing Latency
To improve SIP and IP telephony, we will introduce an mISDN driver improvement that has theoretically no latency, except for the duration of one audio frame plus the network latency. (This is because we need to wait for one frame until we can transfer it to the remote.)
Visit us
Further more we will show the Linux-Call-Router software and how to connect it to Asterisk. Other toppics are ISDN tunneling, real-time audio processing, VoIP.
We would like to sit close to the POC for better communication and easier linking the DECT PBX with our PBX. We are four people but other friends and visitors join us from time to time.
Contact us
- jolly AT eversberg DOT eu
- before and after congrees: +49-4644-973171
- during the congress dial 9000 or from outside: +49-461-50566239000