The Hitchhiker's Guide to Asterisk | ||
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The Asterisk software is what gives a computer system the logic required to run a PBX system. IP based channels, dialplans, AGI scripting and timing insensitive parts of Asterisk can be used without any additional hardware.
The drivers for Digium hardware can be obtained from the CVS server. These will allow you to integrate many types of legacy telephony equipment such as T1/E1, PSTN, FXO and FXS devices.
Libpri will allow Asterisk to work with Primary Rate ISDN interfaces. It is based on the Bellcore specification SR-NWT-002343 for National ISDN. You will need to compile and configure these drivers if you need Asterisk to speak with an ISDN interface.
Zaptel hardware is designed and built by Digium, the owners of Asterisk. The Asterisk PBX system is designed to work with these devices and so are fully supported. Drivers are provided to run the devices on a Linux based operating system.
The X100P and X101P are Foreign Exchange Office (FXO) devices which allow you to connect the Asterisk PBX to a PSTN line. The only difference between an X100P and X101P is a slight difference in their respective chips, but this has no effect on their use with Asterisk. The X100P uses a single PCI slot and supports FXS Loopstart and "Kewlstart" (Loopstart with far end disconnection supervision). With the X100P Asterisk supports both incoming and outgoing calls and supports ring detection and remote hangup.
![]() | These cards can not be used as an FXS device for attaching analog telephones to the Asterisk PBX. |
The S100U is a single port FXS interface for connecting an analog telephone to your Asterisk server via a USB port. This allows you to use any analog telephone for VoIP when connected to Asterisk.
The TDM400P is a half-length PCI 2.2 compliant card which allows you to connect standard analog telephones and analog lines to a computer. The card uses small modules to activate the 4 ports on the card. Which daughter card is plugged onto the board will determine whether the port acts as an FXO or FXS interface. The boards are not selectable between modes; the module used determines the type of interface.
There is an alternate naming convention used as well to reference the type of modules attached to the TDM400P. This is in the form TDM##B where the first hash is the number of FXS (0-4) interfaces and the second hash is the number of FXO (0-4) interfaces.
With the development of the FXO module for the TDM400P it has become the preferred FXO interface device.
The T100P is a single span T-1 (24-channel) card. This card supports both voice and data modes on its single-T span. The T100P supports standard telephony and data protocols, including both RBS and Primary Rate ISDN (PRI) protocol families for voice and PPP, Cisco HDLC and Frame Relay modes. The T100P can also be connected to channel banks for use with Asterisk.
The E100P version is essentially the same card supporting the E-1 European standard.
The T400P and E400P are 4 port versions of the T100P and E100P respectively. The TE410P is a quad T1 or E1, selectable per card or per port, allowing you to do both signaling formats in a single card. This card only supports a 3.3v PCI bus available on newer machines or in 64-bit PCI bus architectures.
IAX stands for Inter-Asterisk eXchange and was developed as an alternative to SIP and H.323. Currently there exists 2 versions of IAX, where IAX2 is the most common used. IAX is not submitted by any standards group, but is currently being adopted by different manufacturers for both soft- and hard-phones.
The biggest advantage for IAX is that it uses only one UDP port and thus works very well behind NAT firewalls. It allocates only the the minimum of bandwidth, that is used at any time.
SIP, or the Session Initiation Protocol, is specified by the IETF. It allows text, voice and multimedia sessions and uses port 5060 udp and tcp, but may use other ports. Most VoIP devices on the market currently support this protocol.
This protocol is not always easy to deploy in a firewalled environment, but with the help of a STUN server not impossible. Asterisk is able to translate the information in the packet headers so that it is possible to run in a NAT'd environment. See chapter 8 for more information.
H.323 is specified by the ITU-T (International Telecommunication Union Stanardization Sector) and was meant for teleconferencing (Speech and Video). It basically should enable you to terminate voice, video, fax and much more over IP, depending on what features your client offers.
There are two implementations for H.323 that can be used with Asterisk: asterisk-oh323 and chan_h323, these are described in detail in a later section.
You might ask yourself now, what module you should choose and quite frankly, there is no all-round answer to that. Implement whichever you have in hands and test it, if you not are happy with that, try the other one.
Asterisk comes with many applications built into it. These applications are used for manipulating calls and giving the user an interactive system. These applications are then used to build a custom dialplan. You will be learning how applications like Dial and other basics work in a dialplan, plus the actual dialplan scripting.
The voicemail system, called Comedian, will be extensivly covered. This will include the backend configuration of the voicemail system for use within Asterisk as well as the actual use of the Comedian mail system.
You can extend the capabilities of Asterisk through the AGI scripting interface. This allows you to create custom applications in virtually any language and tie them into your Asterisk PBX system.
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