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unfix.org VoIP

Abaddon runs an Asterisk for linking up to INOC-DBA, SIPBroker, IAXTel, SIPphone/GizmoProject, ISN (ITAD Subscriber Numbers), ENUM and Google Talk. Asterisk provides voicemail, conferencing, echo latency test and of course the screaming monkeys and the weasels that take over your PBX. Thanks must go to all the Asterisk developers for the main tool, the Debian VoIP Packaging Team and als VoIP-info.org for great documentation and tips.

Debian
Asterisk PBX
got ISN?
600*522
freenum.org

There are several clients connected using it, amongst others:

Some of the numbers that are used:

Number SIP URI INOC-DBA e164.org SIP Broker ISN Description
103monkeys8298*10388299 002064 103*512 103103*522Monkeys
104weasels8298*10488299 002064 104*512 104104*522Weasels
600echo8298*60088299 002064 600*512 600600*522Echo Latency Test
601callerid8298*60188299 002064 601*512 601601*522Caller ID pronounciation
602conference8298*60288299 002064 602*512 602602*522Conference
603tell8298*60388299 002064 603*512 603603*5221-800-555-tell: Tell-Me Informations
604google8298*60488299 002064 604*512 604604*5221-800-goog-411: Google Local

My dutch cell phone number, if you know it, is also redirected to SIP using e164.org's ENUM directory.

VoIP Call Testing

For testing purposes, some known VoIP Test Numbers:

For other numbers check the SIP Broker or VoIP-Info.org for other Phone numbers lists.

Additionally one can try calling yourself by using the following form. Note that your IP address is included in the Caller ID + in the spoken words. You will also need to prefix your number properly using the below table.

PrefixTelco
902...IAXtel
903...SIP Broker
*...SIP Broker
904...SIPPhone/Gizmo
905...ISN
000...ENUM
+...ENUM
name@domain...SIP Direct based on domain

ENUM Lookup

Want to know if a phone number has an enum entry? Then try the following tool.

(+CC XXXXXXXXXX)

VoIP Clients

As my primary VoIP client I have at home.nl a Siemens Gigaset S450 IP, this is a perfect phone which does way too much and is still quite affordable and comes in a stylish design. At home.ch I have a Siemens Gigaset S675 IP which is the newer version of the S450IP which comes with an integrated answering machine (which I don't use as Asterisk handles that) but also has a 65k color screen and supports RSS.

As a laptop-client I am currently using SJPhone (available for Windows/Linux/MacOS) which I picked after having tried out a couple of softphone clients and this being the one that worked properly: settings are clear, dialing and Caller-ID worked, audio configuration has a nice wizard, audio is perfect and clear. Most of the other softphones I tested usually didn't work at all, broke standards, crashed or had cracky audio.

I've attached a Logitech Mobile Pro Headset to the a bluetooth available on Spaghetti and this makes everything work handsfree, I can even walk downstairs and it keeps on working, then the concrete stops the signal though, also very important: BlueTooth devices have blue leds.

For a client with video capabilities check Voice Sistem SRL's excellent SIP and Windows Messenger tutorial document (local copy, original). A client which works pretty well on Linux is Twinkle.

Siemens Gigaset VoIP/DECT Notes

The following are a couple of 'hidden' things inside the Siemens Gigaset S450IP/S675IP/S685IP which I found around the Internet.

To change the answering machine language see page page 66 of the International Full Manual, and page 67 of Swiss French manual. it says: Press Right Arrow, then 8 5 9 2 then: 1+ok for German, 2+ok for French, 3+ok for Italian. This might only work with Swiss Editions of the device as then there is a requirement for three languages. You will find at the bottom of the basestation a number which starts with S30852... If the 3rd group (composed of 4 characters) starts with an 'F' it is a device produced for Switzerland. German devices will start with a 'B'.

Menu (Right Arrow), 8 5 9 and then:

"Menu * # 06 #" allows one to read out Service Information, which includes Serial Number (IPUI), operating hours and variant/version of handset software.

To see what the CLIR (Caller Line Identification Repression) prefix is, use the Menu, "Selected Services", "Next Call", "Anonymous". Then dial a number, the phone will show in the display what the prefix (generally *31 in Europe, *67 in the US).

Some of the Sxxx series phones might have an initial '5-10 second delay' before audio starts. This happens in cojunction with Asterisk. A Solution to the "10 Second Lost Audio Problem" seems to be adding dtmfmode=inband to the sip.conf entries that are having this problem.

If one has more handy details, then don't hesitate to contact me.

Some useful links:

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Last Revision: Tue 20 Jan 2009 00:06:42 CET ©1998-2009 Unfix