Connecting Asterisk With The PSTN
Asterisk connects you to the world of VoIP with its included support for a number of protocols including SIP, IAX, H.323, and more. Did you know that Asterisk is more than just a platform for VoIP? Asterisk is a hybrid platform that is also capable of connecting to traditional telephony interfaces like analog lines and phones, ISDN lines, and T1/E1 lines.
To connect to VoIP phones and service providers, Asterisk uses channel drivers. For each channel type including SIP, IAX, H.323, etc., Asterisk has a corresponding channel driver - chan_sip, chan_iax, chan_ooh323, etc. To make connections to traditional telephony interfaces, Asterisk uses a channel type called chan_dahdi (included with your Asterisk download) and a separate set of software drivers collectively referred to as DAHDI - Digium Asterisk Hardware Device Interface.
DAHDI provides drivers for a number of traditional telephony interface cards, most notably the telephony cards manufactured by Digium®, the creator of Asterisk.
Analog Telephony
Support for regular POTS (Plain Old Telephone Service) is provided by Digium’s series of analog telephony cards. Digium’s analog cards utilize separate capability modules for Foreign Exchange (FXO) trunk lines and Foreign Exchange Station (FXS) telephone sets. The analog cards are provided in four (4), eight (8), and twenty-four (24) modular port varieties for both PCI and PCI-Express slot types. An optional DSP module provides hardware-based echo cancellation for Digium’s analog cards.
Digium Analog Telephony Cards
Digium, the creators and primary sponsors of the Asterisk project offer a complete line of analog telephony interface cards.
| PCI 3.3/5.0V | PCI-Express | |
| Four Port |
TDM410 Up to Four FXO/FXS Loop Start ports, Half-length PCI   |
AEX410 Up to Four FXO/FXS Loop Start ports, Half-length PCI-Express x1 |
| Eight Port |
TDM800 Up to Eight FXO/FXS Loop Start ports, Half-length PCI |
AEX800 Up to Eight FXO/FXS Loop Start ports, Half-length PCI-Express x1 |
| Twenty-Four Port |
TDM2400 Up to Twenty-Four FXO/FXS Loop Start ports, Full-length PCI |
AEX2400 Up to Twenty-Four FXO/FXS Loop Start ports, Full-length PCI-Express x1 |
Digital Telephony
Support for Robbed Bit Signaling (RBS), Basic Rate ISDN (BRI) and Primary Rate ISDN (PRI) lines are provided by Digium’s series of digital telephony cards. The digital cards are provided in one (1), two (2), and four (4) port varieties for both PCI and PCI-Express slot types. An optional DSP module provides hardware-based echo cancellation for Digium’s digital PRI cards; digital BRI cards include on-board DSP-based echo cancellation. The following cards are provided:
| PCI 3.3/5.0V | PCI-Express | |
| Basic Rate ISDN |
B410 Four S/T EuroISDN ports, Half-length PCI |
|
| Basic Rate ISDN / Analog Hybrid |
HA8 Up to 8 analog or digital BRI ports, Half-length PCI |
HB8 Up to 8 analog or digital BRI ports, Half-length PCI-Express x1 |
| Single Span |
TE122 One E1/T1/J1 CCS/CAS span, Low-profile, Half-length PCI |
TE121 One E1/T1/J1 CCS/CAS span, Low-profile, Half-length PCI-Express x1 |
| PCI 5.0V | PCI 3.3V | PCI-Express | |
| Dual Span |
TE205 Two E1/T1/J1 CCS/CAS spans, Half-length 5.0V PCI |
TE210 Two E1/T1/J1 CCS/CAS spans, Half-length 3.3V PCI |
TE220 Two E1/T1/J1 CCS/CAS spans, Half-length PCI-Express x1 |
| Quad Span |
TE405 Four E1/T1/J1 CCS/CAS spans, Half-length 5.0V PCI |
TE410 Four E1/T1/J1 CCS/CAS spans, Half-length 3.3V PCI |
TE420 Four E1/T1/J1 CCS/CAS spans, Half-length PCI-Express x1 |
VoIP Connectivity
Codec Translation -
When a telephone call is placed across a digital network, including VoIP calls, the human voice is transformed from its analog form into a digital form. This rendering of speech sound into digital form utilizes a computer algorithm to encode the signal - compression. The decoding of the digital signal is decompression. The word codec is a combination of these two functions compression – decompression.
All VoIP calls use some kind of codec, the most universal is a codec called G.711, available in two forms: u-law (used in North America) and a-law (used everywhere else). G.711 renders spoken voice into a data stream that utilizes 64kbit/s of bandwidth and is not computationally intensive. G.711 is also the codec used to transmit data across digital PSTN links – ISDN BRI and PRI lines.
As bandwidth is typically very expensive, and processor MIPS are cheaper, it became convenient to invent algorithms capable of high-compressing audio. The two most universally acceptable compression algorithms for compressing audio for telephony are G.729a and G.723.1. Digium sells a telephony card that integrates with Asterisk to perform codec translations of the G.729a and G.723.1 codecs.
| PCI 3.3/5.0V | PCI-Express | |
| Codec Translation |
TC400 120 G.729a or 92 G.723.1 Translations, Low-profile, Half-length PCI |
TCE400 120 G.729a or 92 G.723.1 Transformations, Low-profile, Half-length PCI-Express x1 |