Abaddon runs an FreeSWITCH for linking up to INOC-DBA, SIPBroker, IAXTel, SIPphone/GizmoProject, ISN (ITAD Subscriber Numbers), ENUM and Google Talk. The FreeSWITCH 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 FreeSWITCH developers as it supports IPv6, SCTP and ZRTP out of the box.

There are several clients connected using it, amongst others:

• sip:<name|number>@unfix.org
• INOC-DBA: 8298*xxx
• e164.org: 88299 002064 xxx, e164.org ENUM enabled
• SIPbroker: *512xxx or sip:*512xxx@sipbroker.com
• ISN: xxx*522
• From PSTN one can use of the SIPBroker Gateways, eg +31-30-711-0327 or +353-1-4361726, then enter *512xxx to call the wanted number.

Some of the numbers that are used:

Number	SIP URI	INOC-DBA	e164.org	SIP Broker	ISN	Description
103	monkeys	8298*103	88299 002064 103	*512 103	103*522	Monkeys
104	weasels	8298*104	88299 002064 104	*512 104	104*522	Weasels
600	echo	8298*600	88299 002064 600	*512 600	600*522	Echo Latency Test
601	callerid	8298*601	88299 002064 601	*512 601	601*522	Caller ID pronounciation
602	conference	8298*602	88299 002064 602	*512 602	602*522	Conference
603	tell	8298*603	88299 002064 603	*512 603	603*522	1-800-555-tell: Tell-Me Informations
604	google	8298*604	88299 002064 604	*512 604	604*522	1-800-goog-411: Google Local

My Dutch and Swiss cell phone numbers, if you know them, are also redirected to SIP using e164.org's ENUM directory.

For testing purposes, some known VoIP Test Numbers:

• INOC-DBA: 2914*600 (echo @ NTT), 2914*601 (callerid @ NTT), 2914*602 (callername @ NTT), 30374*100 (time and date @ Modwest)
• SIP Broker: *011-1-88888, *266-300, *013-1-516-687-5089

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.

VoIP number:

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

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

Telephone number: (+CC XXXXXXXXXX)

At home.ch I have a Siemens Gigaset S685 IP with two handsets. At home.nl a Siemens Gigaset S450 IP and a Siemens Gigaset S675 IP do their work, these are perfect phones which does way too much and is still quite affordable and comes in a stylish design.

As a laptop-client (Windows) I am currently using Phonerlite 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. It supports IPv6 and TLS/SRTP thus what more do you want ;)

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

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:

• 1 1 Configure pause after the recall key (1 = 1 second, 2 = 2 seconds, 3 = 3 seconds, 4 = 6 seconds)
• 1 6 Configure pause after line seizure (1 = 1 second, 2 = 3 seconds, 3 = 7 seconds)
• 1 9 [not sure] Anonymous mode on the next call
• 2 0 will show the MAC address [Russian manual claims CLIP setting: 0 FSK, 1-9 also possible) [swiss=7]
• 2 1 / 2 2 / 2 3: select answering machine language as per above
• 2 6 allows you to edit the '926' parameter (SMS on '1' or off '0')
• 2 7 allows you to edit the '927' parameter (unknown what it is)
• 2 9 [not sure] 0/1 (CLIP maybe/AON)
• 4 7 6 2 0 0 +
  • 1 = EEPROM, followed by 09447 01636 13979 to apparently change CLIP to FSK (ETSI ETS 300 648, ETS 300 659-1 CLIP FSK standard); I am looking for the code for DTMF (variant 1 or 2 for the Netherlands). (List of different CLIP codes)
  • 2 = Show version number
  • 3 = FSK/DTMF switch (0/1/2 are the settings, but which is which is unknown) [swiss=0]
  • 4 = Timing/pause of tone (1.5:1 or 2:1)
  • 5 = Defined list of numbers on/off
  • 6 = Seizure Activation

"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:

• Caller ID FAQ