Asterisk setup success 2

Asterisk without VOIP :-)


I have, as my first outside production site, just concluded a very (in my opinion) interesting and educational install as described below. There are still many tweaks which need to be done, and if anyone has any suggestions for improvement, I am always

The customer's request:

  • Connect the three separate offices with voice and data in an inexpensive and supportable manner. Specifically, the customer requested that there be NO VOIP.

The hardware used:

  • 1x Digium T400P $1500
  • 2x Digium T100P's $1000
  • 3x Digium TDM400P's each with 4 proSLICs $1050
  • 6x Digium X100P's (workaround for PBX feature problem) $600
  • 3x Athlon 2500+ systems, each with 512MB RAM and software RAID-1 80GB IDE HDD's, using Gigabyte 7VT600-L motherboards
  • Gentoo Linux 1.4
  • existing KSU at each location: Samsung DCS (series--- one was a DCS compact, the other two were DCS) $1500

The connections used:

  • 1 PRI from BellSouth with 100 DID numbers(Full 23 channels, which is way overkill-- not my decision) $950/mo
  • 2 Point-to-point T1s (price varies on distance) $1300/mo
  • 1 Frame relay T1 to Internet $650/mo
Setup:
  • PRI, frame, and one end of each of the P2P T1s comes into the T400P
  • One T100P takes the other end of the P2P T1s at each remote site
  • 1 TDM400P at each site goes to KSU trunk interface
  • 2 X100Ps at each site go to KSU analog station-side interface

P2P T1 dimensioning:

  • channels 1-4 = E&M trunks for voice
  • channels 5-24 = nethdlc for data (PPP encapsulation)

KSU interfacing:

Incoming main line calls:

  1. PRI -> main T400P
  2. Asterisk plays AutoAttendant stuff, then follows the following for DID

Incoming outside DID line calls:

  1. PRI -> main T400P
  2. Asterisk A to either local, Asterisk B, or Asterisk C, based on DID
  3. Asterisk -> T400P
  4. TDM400P -> DISA trunk on KSU (DID not supported by KSU on analog trunks...argh)
  5. KSU -> Digital Samsung KSU station

Outgoing outside calls:

  1. KSU Station -> KSU -> TDM400P -> Asterisk
  2. Asterisk B,C -> Asterisk A
  3. Asterisk A -> PRI

Local calls:

  1. station dials 9+extension
  2. KSU -> TDM400P -> Asterisk
  3. Asterisk to other Asterisk
  4. Asterisk -> TDM400P -> KSU DISA trunk -> KSU Station

To check voicemail from anywhere in-system:

  1. station dials 98+extension
  2. KSU -> TDM400P -> Asterisk
  3. Asterisk to other Asterisk, if necessary
  4. Asterisk -> VoicemailMain2(<extension>)

To take voicemail, I had to use X100Ps connected as stations, because the KSU cannot forward on Busy/NoAnswer to an external number, and because I had to use DISA instead of DID, asterisk thinks the call is answered when the KSU picks up... before the station even rings. This wouldn't be a problem in a native environment, but to scimp on the cost of handsets, the client wanted to keep the old KSUs.
  1. Incoming call creates a Busy or No Answer condition for the KSU
  2. KSU forwards the call to an "internal-to-the-KSU" extension
  3. this extension is connected to an X100P, which receives the KSU's DTMF voicemail routing digits (one of the saving graces of the Samsung DCS) and takes the call to Voicemail(<extension>)

Data setup:

The data side of things seems to be the least documented aspect of asterisk...probably because it isn't really in asterisk. It is a feature of the Digium cards and the zaptel drivers for them.

Each location has a separate private subnet and a shared transport private subnet nethdlc over T100Ps with PPP encapsulation (this requires sethdlc from zaptel CVS tree, which doesn't appear to be made by default... just 'make sethdlc')

sethdlc hdlc0 mode ppp
ifconfig hdlc0 <local transport IP> pointopoint <remote transport ip> up
route add -net <private supernet> netmask <private supernet mask> gw <remote transport ip>
if asterisk b or c, route add default <remote transport ip>

For the frame:
sethdlc hdlc3 mode fr-ansi create 16
ifconfig hdlc3 up
ifconfig pvc0 <local public ip> pointopoint <ISP gateway> up
route add default gw <ISP gateway>

Obviously, this proves the concept for a number of different features the the most excellect Digium hardware (Thanks, Mark and gang!)

This system has been in production use for a little over a month now, and although I have had a few problems (one bad DIMM, TDM400P revision, BellSouth fouling up the ownership of the PRI which caused a huge delay in getting the main telephone numbers ported over to the PRI), Digium has offered great support and with the source code so wonderfully available, I have been able to diagnose most of the problems with reasonable effort.

Ulexus 2003-11-05

Asterisk without VOIP :-)


I have, as my first outside production site, just concluded a very (in my opinion) interesting and educational install as described below. There are still many tweaks which need to be done, and if anyone has any suggestions for improvement, I am always

The customer's request:

  • Connect the three separate offices with voice and data in an inexpensive and supportable manner. Specifically, the customer requested that there be NO VOIP.

The hardware used:

  • 1x Digium T400P $1500
  • 2x Digium T100P's $1000
  • 3x Digium TDM400P's each with 4 proSLICs $1050
  • 6x Digium X100P's (workaround for PBX feature problem) $600
  • 3x Athlon 2500+ systems, each with 512MB RAM and software RAID-1 80GB IDE HDD's, using Gigabyte 7VT600-L motherboards
  • Gentoo Linux 1.4
  • existing KSU at each location: Samsung DCS (series--- one was a DCS compact, the other two were DCS) $1500

The connections used:

  • 1 PRI from BellSouth with 100 DID numbers(Full 23 channels, which is way overkill-- not my decision) $950/mo
  • 2 Point-to-point T1s (price varies on distance) $1300/mo
  • 1 Frame relay T1 to Internet $650/mo
Setup:
  • PRI, frame, and one end of each of the P2P T1s comes into the T400P
  • One T100P takes the other end of the P2P T1s at each remote site
  • 1 TDM400P at each site goes to KSU trunk interface
  • 2 X100Ps at each site go to KSU analog station-side interface

P2P T1 dimensioning:

  • channels 1-4 = E&M trunks for voice
  • channels 5-24 = nethdlc for data (PPP encapsulation)

KSU interfacing:

Incoming main line calls:

  1. PRI -> main T400P
  2. Asterisk plays AutoAttendant stuff, then follows the following for DID

Incoming outside DID line calls:

  1. PRI -> main T400P
  2. Asterisk A to either local, Asterisk B, or Asterisk C, based on DID
  3. Asterisk -> T400P
  4. TDM400P -> DISA trunk on KSU (DID not supported by KSU on analog trunks...argh)
  5. KSU -> Digital Samsung KSU station

Outgoing outside calls:

  1. KSU Station -> KSU -> TDM400P -> Asterisk
  2. Asterisk B,C -> Asterisk A
  3. Asterisk A -> PRI

Local calls:

  1. station dials 9+extension
  2. KSU -> TDM400P -> Asterisk
  3. Asterisk to other Asterisk
  4. Asterisk -> TDM400P -> KSU DISA trunk -> KSU Station

To check voicemail from anywhere in-system:

  1. station dials 98+extension
  2. KSU -> TDM400P -> Asterisk
  3. Asterisk to other Asterisk, if necessary
  4. Asterisk -> VoicemailMain2(<extension>)

To take voicemail, I had to use X100Ps connected as stations, because the KSU cannot forward on Busy/NoAnswer to an external number, and because I had to use DISA instead of DID, asterisk thinks the call is answered when the KSU picks up... before the station even rings. This wouldn't be a problem in a native environment, but to scimp on the cost of handsets, the client wanted to keep the old KSUs.
  1. Incoming call creates a Busy or No Answer condition for the KSU
  2. KSU forwards the call to an "internal-to-the-KSU" extension
  3. this extension is connected to an X100P, which receives the KSU's DTMF voicemail routing digits (one of the saving graces of the Samsung DCS) and takes the call to Voicemail(<extension>)

Data setup:

The data side of things seems to be the least documented aspect of asterisk...probably because it isn't really in asterisk. It is a feature of the Digium cards and the zaptel drivers for them.

Each location has a separate private subnet and a shared transport private subnet nethdlc over T100Ps with PPP encapsulation (this requires sethdlc from zaptel CVS tree, which doesn't appear to be made by default... just 'make sethdlc')

sethdlc hdlc0 mode ppp
ifconfig hdlc0 <local transport IP> pointopoint <remote transport ip> up
route add -net <private supernet> netmask <private supernet mask> gw <remote transport ip>
if asterisk b or c, route add default <remote transport ip>

For the frame:
sethdlc hdlc3 mode fr-ansi create 16
ifconfig hdlc3 up
ifconfig pvc0 <local public ip> pointopoint <ISP gateway> up
route add default gw <ISP gateway>

Obviously, this proves the concept for a number of different features the the most excellect Digium hardware (Thanks, Mark and gang!)

This system has been in production use for a little over a month now, and although I have had a few problems (one bad DIMM, TDM400P revision, BellSouth fouling up the ownership of the PRI which caused a huge delay in getting the main telephone numbers ported over to the PRI), Digium has offered great support and with the source code so wonderfully available, I have been able to diagnose most of the problems with reasonable effort.

Ulexus 2003-11-05

Created by: oej, Last modification: Wed 05 of Nov, 2003 (07:55 UTC)
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