Difference between revisions of "Telecom"
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| − | + | Telecom in LinuxMCE is enabled by the integration of an open-source PBX system for Linux called [[http://en.wikipedia.org/wiki/Asterisk_%28PBX%29 Asterisk]]. | |
| − | + | The basic Asterisk software includes many features available in proprietary PBX systems: voice mail, conference calling, interactive voice response (phone menus), and automatic call distribution. | |
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| − | + | To attach ordinary telephones to a Linux server running Asterisk, or to connect to PSTN trunk lines, the server must be fitted with special hardware. Digium and a number of other firms sell PCI cards to attach telephones, telephone lines, T1 and E1 lines, and other analog and digital phone services to a server. | |
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| − | + | Perhaps of more interest to many deployers today, Asterisk also supports a wide range of Voice over IP protocols, including SIP, MGCP and H.323. Asterisk can interoperate with most SIP telephones, acting both as registrar and as a gateway between IP phones and the PSTN. Asterisk developers have also designed a new protocol, Inter-Asterisk eXchange (IAX2), for efficient trunking of calls among Asterisk PBXes, and to VoIP service providers who support it. Some telephones support the IAX2 protocol directly for communicating with an Asterisk server (see Comparison of VoIP software for examples). | |
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| − | + | VoIP telephone companies have begun to support Asterisk; many[2] now offer IAX2 or SIP trunking direct to an Asterisk box as an alternative to providing the customer with an ATA. | |
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[[Category:Telecom| ]] | [[Category:Telecom| ]] | ||
Revision as of 23:28, 19 May 2008
Telecom in LinuxMCE is enabled by the integration of an open-source PBX system for Linux called [Asterisk].
The basic Asterisk software includes many features available in proprietary PBX systems: voice mail, conference calling, interactive voice response (phone menus), and automatic call distribution.
To attach ordinary telephones to a Linux server running Asterisk, or to connect to PSTN trunk lines, the server must be fitted with special hardware. Digium and a number of other firms sell PCI cards to attach telephones, telephone lines, T1 and E1 lines, and other analog and digital phone services to a server.
Perhaps of more interest to many deployers today, Asterisk also supports a wide range of Voice over IP protocols, including SIP, MGCP and H.323. Asterisk can interoperate with most SIP telephones, acting both as registrar and as a gateway between IP phones and the PSTN. Asterisk developers have also designed a new protocol, Inter-Asterisk eXchange (IAX2), for efficient trunking of calls among Asterisk PBXes, and to VoIP service providers who support it. Some telephones support the IAX2 protocol directly for communicating with an Asterisk server (see Comparison of VoIP software for examples).
VoIP telephone companies have begun to support Asterisk; many[2] now offer IAX2 or SIP trunking direct to an Asterisk box as an alternative to providing the customer with an ATA.
