We're pleased to announce the availability of Asterisk: The Future of Telephony online!
The result of hundreds of hours of painstaking labour, this book represents the work of Jim Van Meggelen, Jared Smith, and Leif Madsen over the past year as part of the Asterisk Documentation Project.
Thanks to O'Reilly Media for publishing the book and agreeing to publish it under the Creative Commons license.
| What is the correct FXS impedance for the UK ? |
| Answer: | The first value in Hz is the Frequency and the second value is the On - Off Sequence in seconds| Country | Ring Frequency | Dial Tone | Busy Tone | Ring back Tone | FXS Impedance | CallerID Method | TX Gain | RX Gain | Onhook Voltage | | Australia | 25 Hz
0.4 - 0.2 - 0.4 - 2.0 | 425 Hz
Continuous | 400 Hz
0.375 - 0.375 | 400 Hz
0.4 - 0.2 - 0.4 - 2.0 | 220 + 820 Ohm || 120nF | | 0 dB | -7 dB | | | Austria | 50 Hz
1.0 - 5.0 | 420 Hz
Continuous | 420 Hz
0.4 - 0.4 | 420 Hz
1.0 - 5.0 | 220 + 820 Ohm || 115nF | | 0 dB | -7 dB | | | Belgium | | | | | 150 + 830 Ohm || 72nF | | | | | | Bulgaria | | | | | 220 + 820 Ohm || 115nF | | | | | | Canada | | | | | | | | | | | China | 20 Hz
1.0 - 4.0 | 350 + 440 Hz
Continuous | 450 Hz
0.35 - 0.35 | 450 Hz
1.0 - 4.0 | 200 + 680 Ohm || 100nF | | 0 dB | 0 dB | | | Cyprus | | | | | | | | | | | Czech Republic | | | | | | | | | | | Denmark | | | | | 400 + 500 Ohm || 330nF | | | | | | Estonia | | | | | | | | | | | Finland | | | | | | | | | | | France | 50 Hz
1.5 - 3.5 | 440 Hz
Continuous | 440 Hz
0.4 - 0.4 | 440 Hz
1.5 - 3.5 | 180 + 910 Ohm || 150nF | | -2 dB | -9 dB | | | Germany1 | 25 Hz
0.25 - 4.0 -1.0 - 4.0 | 425 Hz Continuous | 425 Hz
0.48 - 0.48 | 425 Hz
0.25 - 4.0 - 1.0 - 4.0 | 220 + 820 Ohm || 115nF | | +3 dB | -10 dB | | | Germany2 | 25 Hz
0.5 - 4.0 - 1.0 - 4.0 | 425 Hz Continuous | 425 Hz
0.15 - 0.475 | 425 Hz
0.5 - 4.0 - 1.0 - 4.0 | 220 + 820 Ohm || 115nF | | 0 dB | -7 dB | | | Greece | | | | | 600 Ohm | | | | | | Hungary | | | | | | | | | | | Iceland | | | | | | | | | | | India | | | | | 370 + 620 Ohm || 310nF | | | | | | Italy | 25 Hz
1.0 - 4.0 | 425 Hz
0.2 - 0.2 - 0.6 - 1.0 | 425 Hz
0.5 - 0.5 | 425 Hz
1.0 - 4.0 | 600 Ohm | | 0 dB | -7 dB | | | Japan | 20 Hz
1.0 - 2.0 | 400 Hz
Continuous | 400 Hz
0.5 - 0.5 | 400 Hz
1.0 - 2.0 | 600 Ohm | | 0 dB | -9 dB | | | Korea | 20 Hz
1.0 - 2.0 | 350 + 440 Hz
Continuous | 480 + 620 Hz
0.5 - 0.5 | 440 + 480 Hz
1.0 - 2.0 | 600 Ohm | | 0 dB | -9 dB | | | Lithuania | | | | | | | | | | | Luxembourg | | | | | | | | | | | Latvia | | | | | | | | | | | Malta | | | | | | | | | | | Netherlands | 25 Hz
1.0 - 4.0 | 425 Hz
Continuous | 425 Hz
0.5 - 0.5 | 425 Hz
1.0 - 4.0 | 600 Ohm | | 0 dB | -7 dB | | | New Zealand | 25 Hz
0.4 - 0.2 - 0.4 - 2.0 | 400 Hz
Continuous | 400 Hz
0.5 - 0.5 | 400 + 450 Hz
0.4 - 0.2 - 0.4 - 2.0 | 370 + 620 Ohm || 310nF | | +3 dB | -9 dB | | | Norway | | | | | 120 + 820 Ohm || 110nF | | | | | | Poland | | | | | | | | | | | Portugal | | | | | | | | | | | Rep of Ireland | | | | | | | | | | | Romania | | | | | | | | | | | Slovakia | | | | | 220 + 820 Ohm || 115nF | | | | | | Slovenia | | | | | 220 + 820 Ohm || 115nF | | | | | | South Africa | | | | | 220 + 820 Ohm || 115nF | | | | | | Spain | 25 Hz
1.5 - 3.0 | 425 Hz
Continuous | 425 Hz
0.2 - 0.2 | 425 Hz
1.5 - 3.0 | 600 Ohm | | 0 dB | -7 dB | | | Sweden | | | | | 270 + 750 Ohm || 150nF | | | | | | Switzerland | | | | | | | | | | | Turkey | | | | | | | | | | | UK | 25 Hz
0.4 - 0.2 - 0.4 - 2.0 | 350 + 440 Hz
Continuous | 400 Hz
0.375 - 0.375 | 400 + 450 Hz
0.4 - 0.2 - 0.4 - 2.0 | 370 + 620 Ohm || 310nF | ETSI-FSK with PR | +3 dB | -9 dB | 50v DC | | | Note that some non BT exchanges in the UK may need you to specify CTR21 (270 Ohm + 750 Ohm || 150nF) for Impedance and Bellcore for CID | | USA1 | 20 Hz
2.0 - 4.0 | 350 + 440 Hz
Continuous | 480 + 620 Hz
0.5 - 0.5 | 440 + 480 Hz
2.0 - 4.0 | 600 Ohm | Bellcore | +3 dB | -3 dB | 48v DC | | USA2 | 20 Hz
1.0 - 4.0 | 350 + 440 Hz
Continuous | 480 + 620 Hz
0.5 - 0.5 | 440 + 480 Hz
1.0 - 4.0 | 350 + 1000 Ohm || 210nF | Bellcore | 0 dB | 0 dB | 48v DC |
To notify us of corrections, additions or amendments to this information please click here 600c 600 Ohms complex
600r 600 Ohms real
900c 900 Ohms complex
900r 900 ohms real
complex1 220 ohms + (820 ohms || 115nF)
complex2 270 ohms + (750 ohms || 150nF)
complex3 370 ohms + (620 ohms || 310nF)
complex4 600r, line = 270 ohms + (750 ohms || 150nF)
complex5 320 + (1050 || 230 nF), line = 12Kft
complex6 600r, line = 350 + (1000 || 210nF)
see detailed doc |
|
| What is the Called ID Scheme in the UK ? |
| Answer: | North America - Bellcore
Canada - CID
China - Bellcore
Brazil - DTMF
Denmark - DTMF
Finland - ETSI-DTMF or DTMF
France - ETSI-FSK
Germany - ETSI-FSK
Norway - ETSI-FSK
Sweden - DTMF or ETSI-DTMF
Taiwan - ETSI-FSK
UK - ETSI-FSK with PR (Note that some non BT exchanges in the UK, notably NTL/Telewest/Virgin may need you to specify CTR21)
ETSI-DTMF with PR
ETSI-DTMF after ring
see detailed doc |
|
| What is VoIP? |
| Answer: | Voice over Internet Protocol (VoIP) - technology that enables one to make and receive phone calls through the Internet instead of using the traditional analogue PSTN (Public Switched Phone Network) lines. |
|
| What are the advantages of VoIP over analogue PSTN lines? |
| Answer: | The primary main advantage of VoIP over PSTN lines is cost (it's cheaper!) Other advantages of VoIP are as follows: digital features not commonly available on PSTN lines such as voicemail, caller ID, conference, music-on-hold, etc. |
|
| What type of service and equipment are needed for VoIP deployment? |
| Answer: | The following equipment and services are required for VoIP deployment: High-Speed Broadband connection, VoIP Phones (Softphones will require PC) or Analogue Telephone Adapters (ATAs) and VoIP Service Provider (terminate calls). |
|
| Can I use dial-up for VoIP or do I need broadband? |
| Answer: | Dial-up can be used for VoIP when necessary or if its the only type of connection available. However, we recommend using broadband since certain VoIP codecs (e.g. G.711) require higher bandwidth. |
|
| Do I need a computer to make/receive VoIP calls? |
| Answer: | The answer depends on whether or not you will be using a softphone with your VoIP integration. VoIP does not require any computer to make/receive phone calls (only ATA devices or VoIP Phones). If softphones are used instead of physical phones or ATA devices, then computers are needed. |
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| Can I surf the web during VoIP calls? |
| Answer: | Yes, VoIP allows web surfing while making and receiving VoIP calls simultaneously. It shares the bandwidth connection with other LAN computers and prioritizes voice. |
|
| Should I use an ATA or an VoIP Phone? |
| Answer: | It depends on your preference and budget. An ATA will allow you to use analogue phones for VoIP. While this might save money, they do not have one touch feature keys (e.g. transfer, hold, etc). On the other hand, using VoIP Phones will provide more features that are similar to digital phones. |
|
| Can I use VoIP for all the phones in my residence? |
| Answer: | Definitely, VoIP can replace every single phone in your residence. Both ATA devices and VoIP Phones can be used instead of regular analogue phones. This setup requires an account with a VoIP provider. |
|
| How I can make/receive free VoIP calls to/from remote location? |
| Answer: | Making and receiving free VoIP calls can be made possible by signing up with VoIP Service Providers such as Free World Dialup (FWD) that allow unlimited VoIP calling. These providers will sometimes allow making/receiving free VoIP to PSTN calls (and vice versa). In addition, VoIP end user devices such as ATAs and VoIP Phones can be set up to make point to point VoIP calls between one another. |
|
| Can VoIP make and receive calls to/from PSTN lines? |
| Answer: | Absolutely! VoIP users can definitely make and receive calls to/from PSTN lines. Any type of calls (e.g. local, long distance, international, etc.) are allowed. This requires an account with VoIP Service Providers that provide termination. |
|
| May I keep my existing phone number when migrating to VoIP? |
| Answer: | Most VoIP Service Providers will allow you to keep your existing PSTN phone number for VoIP. However, you will need to check with the provider since not all of them offer this service. A signed "Letter of Authorization" may be required by the provider when keeping your number. |
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| What are VoIP Service Providers (VSPs)? |
| Answer: | VoIP Service Providers (VSPs) are the next generation telcos that provide interconnection between VoIP and PSTN networks. They allow call origination and termination between these two networks. |
|
| Which VoIP Service Provider should I use? |
| Answer: | VoIP Service Providers can be selected based on the services and calling plans that they provide. The features they offer can greatly differ based on the price of the calling plan that you choose. Rates vary between providers and their pricing ranges from per minute charges to flat monthly bills. Choosing the right calling plan should be based on your monthly phone usage and company budget. The list shows current available VoIP Service Providers |
|
| What are IP PBXs? |
| Answer: | IP PBXs (Private Branch Exchanges) are complete phone systems that provide advanced telephony features and services between VoIP and PSTN networks. Common features and services include: call transfer, conference, voicemail, music-on-hold, auto-attendant, and auto call routing. |
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| What are VoIP Gateways? |
| Answer: | VoIP gateways are devices that take analogue voice signals and convert them to IP for transport over the LAN or WAN. |
|
| What are FXO and FXS ports? |
| Answer: | Foreign Exchange Office (FXO) ports are interfaces used to connect with the central office or PSTN analogue lines. Foreign Exchange Station (FXS) ports are interfaces used to connect with end user devices (e.g. phone or fax). |
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| What are PSTN failover lines? |
| Answer: | PSTN (Public Switched Phone Network) failover lines are used as backup connections in the event your VoIP or Internet connection goes down. These are optional ports on ATA devices or VoIP Phones that connect directly to the analogue PSTN lines coming from the phone company. This setup requires having both regular analogue phone lines and an account any VoIP Service Provider. |
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| Which VoIP signaling protocols are commonly used? |
| Answer: | VoIP signaling protocols are used to setup and tear down calls, carry the required information to locate end users, and negotiate device capabilities. The following list shows the most common VoIP signaling protocols available: SIP (Session Initiation Protocol), H.323, Cisco SCCP (Skinny Client Control Protocol), IAX (Inter-Asterisk Exchange), and MGCP (Media Gateway Control Protocol). |
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| Which VoIP codec should I use? |
| Answer: | VoIP codecs convert analogue voice signals to their digital encoded version. Codecs vary in size, sound quality, bandwidth and computation requirements. The most common VoIP codecs currently available are: G.711 (alaw & ulaw), G.723, G.726, G.729, GSM, and iLBC. |
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| What are Gatekeepers and Registrars? |
| Answer: | Gatekeepers and Registrars are gateways that provide authentication, authorization, call control and call routing, and session invites for end user devices. |
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