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<P>From: <I>"Trevor G. Hammonds" <trevor@concipient.net></I><BR>> > From: Yuan LIU<BR>> > Sent: Tuesday, February 06, 2007 8:11 PM<BR>> ><BR>> > After reading through several recent threads, I started to wonder why the<BR>> > Cisco document (and other VoIP documents) appears to present this issue as<BR>> > VoIP gateway specific. Don't (plain old) PBX' face the same issue if they<BR>> > use analogue interfaces? If there are analogue PBX' at all, how do they<BR>> > solve the problem?<BR>><BR>>Yuan,<BR>>Well engineered analogue PBXs typically do not use standard loop start<BR>>subscriber lines. When digital trunks are not an option, they use analogue<BR>>PBX and/or DID trunks. At the very least, ground start circuits are<BR>>preferred to avoid
"glare". The best call quality for analogue is achieved<BR>>by using four-wire E&M trunks that provide answer and disconnect<BR>>supervision. There are two-wire trunks (which are probably more common), as<BR>>well as different signalling methods. These trunks require special<BR>>interface hardware, and I am unaware of any that work with Asterisk. As the<BR>>cards are typically very expensive, it is usually better to go with digital<BR>>if you require that functionality. It would be nice to see a BRI interface<BR>>for Asterisk that works in North America, as BRI circuits are often<BR>>comparable in price to analogue lines.<BR></P>
<P>Thanks for the enlightment, Trevor. I always thought that standard ISDN cards (presumably BRI) work with Asterisk if they work under Linux?</P>
<P>> Sincerely,<BR>> Trevor Hammonds<BR></P></FONT></DIV></div></html>