login | register
Wed 01 of Aug, 2007 [09:23 UTC]

voip-info.org

Search with Google
Search this site with Google. Results may not include recent changes.

Web www.voip-info.org
Shoutbox
  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.38s
  • Memory usage: 2.20MB
  • Database queries: 29
  • GZIP: Disabled
  • Server load: 3.12

T1

T1 (also referred to as DS1) is a standard for digital transmission over phone lines at 1.544 Mbps.
It is split into 24 channels of 64Kbps each. In the original standard, signalling was inband (robbed bit signalling). Now T1s are often "clear channel" and all bits are available for data.

Each 64Kbps channel (also known as a DS0) can carry data or voice traffic, and two or more channels can be combined into one higher speed data channel.

T1s are often used to deliver phone connections to a PBX in blocks of 24 lines.
ISDN Primary Rate PRI lines are delivered over T1 circuits in the USA.

A channelbank can convert a T1 to 24 analog phone lines.

T1s are also used to deliver high-speed data service.

A good overview on the fundamentals of digital telephony, digital voice, basic TDM, and T-1 and E-1 applications is the T-1/E-1 Technology Primer, published by Intel Corporation in 2001.

See Also

Created by jht2, Last modification by Paul Gillman on Fri 25 of May, 2007 [13:42 UTC]

Comments Filter

Re:

by T1 on Wednesday 18 of April, 2007 [05:12:30 UTC]

Great article on T1 Lines

by T1 on Tuesday 02 of January, 2007 [06:53:58 UTC]

by T1 on Thursday 16 of November, 2006 [03:19:35 UTC]



by T1 on Thursday 16 of November, 2006 [03:18:26 UTC]

Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

Page Changes | Comments

Sponsored by:

Terms of Service Privacy Policy
© 2003-2007 Arte Marketing, Inc.

Powered by bitweaver