VINA Integrator

The VINA Integrator is the original product model. It is now called the Larscom Integrator 300, but it is the same item, without change. Regarding the Lucent ConnectReach variety…according to Larscom technical support:

“The Lucent T1 Integrator also known as the ConnectReach integrator, allows the user to connect to either a SLIC-200 or an AnyMedia.”

AnyMedia is hardware on the telco side designed to talk to any piece of network gear. I have not found information about SLIC-200. Time/Warner Telecom technical support also claims that the ConnectReach speaks a “proprietary” protocol and cannot be used to terminate a standard T1 of any sort. Good thing we are interested in using it only as a basic channel bank. 😉

The ConnectReach used for this research was purchased from eBay for about US$50. This seems to be the going rate for ConnectReach hardware. I was able to access (and reset to defaults) the piece, and nose around a bit in the web-based configuration pages. I can verify that you are able to select, in software, the following settings for arbitrary ranges of analog channels (coming out of the Amp50 25-pair connector) — however, see below for more on the FXO/FXS configuration section:

(cut-n-paste from the Channel Bank configuration page)

signaling:

loop-gnd
e&m
unconfigured

incoming-type:

wnk-s
imm-s
dly-d
wnk-d

outgoing-type:

wnk-s
imm-s
dly-d

analog-interface:

fxs-loop
fxs-ground
fxo-did
fxo-loop
fxo-ground

far-end-disconnect:

disable
enable

loop-reversal:

disable
enable

tx-gain-db:

0
-3
-6

rx-gain-db:

0
-3
-6
-9
-12

on-hook-threshold:

300…2000 msec

alarm-state:

idle
busy

unconfigured-state:

idle
busy

lp-gnd-disc-delay:

20…1200 msec

on-hook-tx:

The VINA has two T1 ports. On the ConnectReach, one is “Network” (Primary, intended to plug into the SmartJack), and the other is “Secondary” (intended as a general T1 that I can tell). “Network” T1 Configuration includes the following settings:

enable: off

framing:

D4
ESF
TR08-Framing
TR08-ESF

linecode:

B8ZS
AMI

buildout:

0-133ft(DSX)/0dB(CSU)
133-266ft(DSX)
266-399ft(DSX)
399-533ft(DSX)
533-655ft(DSX)
-7.5dB(CSU)
-15dB(CSU)
-22.5dB(CSU)

clock:

network
internal

pulse-density-enforce:

alarm-operation:

alarm-delay:

0…3600 secs

is-slave:

with a link to another page, labelled:
“IP-over-FDL”

The Secondary T1 has fewer config options:

enable:

framing:

linecode:

B8ZS
AMI

buildout:

0-133ft(DSX)/0dB(CSU)
133-266ft(DSX)
266-399ft(DSX)
399-533ft(DSX)
533-655ft(DSX)
-7.5dB(CSU)
-15dB(CSU)
-22.5dB(CSU)

pulse-density-enforce:

alarm-operation:

alarm-delay:

0…3600 secs

has-slave:

and no “IP-over-FDL” option.

So far (as of today, 26 Oct 2004) it looks good, in my less-than-expert opinion, that the ConnectReach could be used as a simple channel bank with Asterisk. When I get a T100P to test, I can verify if this is true or not.

3 Nov 2004 (yes, it’s starting to look more like a blog than an article….I will clean it up when I have everything figured out).

Really, though, it’s not complicated. Assuming you get your Integrator/ConnectReach with the Carrier password set to blank, it’s really simple to figure out. If you don’t have the Carrier password, or you lost yours, don’t bother calling Larscom. They simply will not help you, especially with previously-owned/auction gear (for good reason, actually….would you want them giving out the carrier password for your telecom equipment?).

So. It comes with analog interfaces in groups of 8, each group set for FXO or FXS, not software selectable. Mine came with 16 FXS. The software selectability is in the T1 interfaces. As mentioned above, the “Network” interface supports more T1 formats than does the Secondary. The Secondary typically is used for trunking between/slaving Integrator/ConnectReach units; I have not yet tried to use it as the primary T1 interface for the unit (no reason, really; the default configuration maps the Network T1 to the Analog Voice ports starting with DS0 #1 mapped to Analog #1, so it doesn’t get much easier). There are myriad configuration possibilities, and it is possible to custom-map on a one-by-one basis, DS0 to analog ports.

Short answer: I am 95% positive that yes, this unit can be used just fine as a channel bank with Asterisk. Just be sure of what analog port types you are getting (ask the seller before bidding if you are looking at auction items). I would be willing to bet that 99% of the used Integrator/ConnectReach units out there for sale are FXS-only analog ports; this item typically is intended to have phones connected to it — it provides dial tone on the analog ports, in other words. The DS0 can be configured however you like them, just match it up in zapata.conf. It works just fine with a T100P, a T1 crossover cable and ESF/B8ZS. I say 95% above because at this point I can cause events to happen in the Asterisk console when I change modes in the Diagnose menu, and I can call out from Asterisk (over the T1) and watch the line status change to Ringing, so it’s clear that the two work together just fine. The rest will be an exercise in telecommunications system design to figure out where I want to insert this unit.

Keep in mind that you only get as many DS0 as you have actual analog ports; it will not let you configure any more than that. So in my setup, I have 16 analog FXS, and therefore can only configure 16 DS0.

For US$50, it’s hard to beat if all you want are FXS ports. And having used the Time Warner Telecom ConnectReach on my wall for our phone service for two years now, I can say that they are quite reliable and the sound quality is just peachy. 🙂

Works through a Cisco 1760 Router

11/26/06
I’ve had the Lucent ConnectReach for some time now. Probably about a year. I was able to contact Lucent and somehow get them to give me the user/pass to log into the machine to clear out the carrier password (sorry, don’t ask for it. I won’t give it out as I told them I wouldn’t, and I stick to my word).
Recently I got a Cisco 1760 router. Along with it I bought a PVDM-12 (DSP chip, does 12 channels), VWIC-2MFT-T1-DI, VWIC-1MFT-T1, VIC-2FXO, VIC-2FXS. Anywho, I managed to get the ConnectReach working with Asterisk through the Cisco. The 1760 connects via SIP, then has everything setup, and uses the ConnectReach as a channel bank. Since the version of ConnectReach I got doesn’t have any DSP’s in it, I can only use it as a channel bank with fxs on the pots side. I might be able to do fxo, but I haven’t tried, and I don’t have land-line access.

Relavent Cisco configs:


!
tdm clock T1 1/0 both export line
!
voice service pots 
!
voice service voip 
 sip
  bind all source-interface FastEthernet0/0
! 
no voice hpi capture buffer
no voice hpi capture destination 
!
!
voip-incoming translation-rule calling 1
!
!
controller T1 1/0
 framing esf
 fdl ansi
 linecode b8zs
 channel-group 17 timeslots 24
 ds0-group 1 timeslots 1 type fxs-loop-start
 ds0-group 2 timeslots 2 type fxs-loop-start
 ds0-group 3 timeslots 3 type fxs-loop-start
 ds0-group 4 timeslots 4 type fxs-loop-start
 ds0-group 5 timeslots 5 type fxs-loop-start
 ds0-group 6 timeslots 6 type fxs-loop-start
!
interface FastEthernet0/0
 ip address 10.0.3.9 255.255.255.240
 speed auto
 no cdp enable
!
interface Serial1/0:17
 no ip address
 no cdp enable
!
voice-port 1/0:1
 output attenuation 0
!
voice-port 1/0:2
 output attenuation 0
! 
voice-port 1/0:3
 output attenuation 0
!
voice-port 1/0:4
 output attenuation 0
!
voice-port 1/0:5
 output attenuation 0
!
voice-port 1/0:6
 output attenuation 0
!
dial-peer cor custom
!
dial-peer voice 5 pots
 incoming called-number 269
 destination-pattern 269
 port 1/0:2
!
dial-peer voice 7 pots
 incoming called-number 265
 destination-pattern 265
 port 1/0:1
!
sip-ua 
 calling-info pstn-to-sip from name set POTS
 sip-server ipv4:10.0.3.2
!

This makes port 1 extention 265 and port 2 extention 269
(ports on the ConnectReach, it bascially passes through how you mark it)

And the relavent config on the ConnectReach


Network-t1
    enable on
    framing ESF
    linecode B8ZS
    buildout 0-133ft(DSX)/0dB(CSU)
    clock network
    pulse-density-enforce off
    alarm-operation on
    alarm-delay 15
    is-slave off
    !
Modify-tr08-data
    shelf ASHELF
    bpv-threshold 155
    !
IP-over-FDL
    IP-address Disable
    netmask 255.255.255.0
    !
DS0-Provisioning
    analog-begin 1
    analog-end 16
    sec-voice-begin 0
    sec-voice-end 0
    sec-data-begin 0
    sec-data-end 0
    data-begin 17
    data-end 24
    alignment same
    !
ConnectReach
    1-4 none
    5-8 none
    9-12 none
    13-16 none
    17-20 none
    21-24 none
    ClearReach-enable off
    !
Synchronous-interface
    encapsulation HDLC
    HDLC-inversion off
    !
Modify-encap-data
    IP-address Disable
    netmask 255.255.255.0
    RIP Disable
    version-RIP 1
    !
Ethernet
    IP-address 10.100.31.169
    netmask 255.255.255.0
    secondary-ip 0.0.0.0
    sec-netmask 255.255.255.0
    RIP Disable
    version-RIP 1
    Link-integrity-test on
    !
Channel-bank
    begin 1
    end 16
    signaling loop-gnd
    incoming-type wnk-s
    outgoing-type wnk-s
    analog-interface fxs-loop
    far-end-disconnect enable
    loop-reversal enable
    tx-gain-db -3
    rx-gain-db -3
    on-hook-threshold 1250
    alarm-state busy
    unconfigured-state busy
    lp-gnd-disc-delay 125
    on-hook-tx on
Ch   Signalling   Inc   Out     Analog    FarEdD Loop-rev TG RG  OHT AlSt Line
 1     loop-gnd wnk-s wnk-s   fxs-loop    enable   enable -3 -3 1250 busy    1
 2     loop-gnd wnk-s wnk-s   fxs-loop    enable   enable -3 -3 1250 busy    2
 3     loop-gnd wnk-s wnk-s   fxs-loop    enable   enable -3 -3 1250 busy    3
 4     loop-gnd wnk-s wnk-s   fxs-loop    enable   enable -3 -3 1250 busy    4
 5     loop-gnd wnk-s wnk-s   fxs-loop    enable   enable -3 -3 1250 busy    5
 6     loop-gnd wnk-s wnk-s   fxs-loop    enable   enable -3 -3 1250 busy    6
 7     loop-gnd wnk-s wnk-s   fxs-loop    enable   enable -3 -3 1250 busy    7
 8     loop-gnd wnk-s wnk-s   fxs-loop    enable   enable -3 -3 1250 busy    8
 9     loop-gnd wnk-s wnk-s   fxs-loop    enable   enable -3 -3 1250 busy    9
10     loop-gnd wnk-s wnk-s   fxs-loop    enable   enable -3 -3 1250 busy   10
11     loop-gnd wnk-s wnk-s   fxs-loop    enable   enable -3 -3 1250 busy   11
12     loop-gnd wnk-s wnk-s   fxs-loop    enable   enable -3 -3 1250 busy   12
13     loop-gnd wnk-s wnk-s   fxs-loop    enable   enable -3 -3 1250 busy   13
14     loop-gnd wnk-s wnk-s   fxs-loop    enable   enable -3 -3 1250 busy   14
15     loop-gnd wnk-s wnk-s   fxs-loop    enable   enable -3 -3 1250 busy   15
16     loop-gnd wnk-s wnk-s   fxs-loop    enable   enable -3 -3 1250 busy   16
    !
t1-traps
    ESs-threshold 0
    SESs-threshold 0
    SEFSs-threshold 0
    UASs-threshold 0
    CSSs-threshold 0
    PCVs-threshold 0
    LESs-threshold 0
    BESs-threshold 0
    DMs-threshold 0
    LCVs-threshold 0
    !
Telnet
    enable on
    Port 23
    !
HTTP
    enable on
    Port 80
    !

The IP listed above is for management only. Its also for if you use it as your gateway (e.g. you by a fractional T1 from the telco, half voice, half data), you would plug your switch (or router) here.

Here is the HW output on the ConnectReach:


  Hardware version: 0023
  Serial number 000000100xxxx
  Ethernet address 00.60.49.00.xx.xx
  Analog channels 1 - 8 are type FXS
  Analog channels 9 - 16 are type FXS
  Analog channels 17 - 24 are type FXS
  4194304 bytes of main memory
  No SIMM memory present
  Primary T1 is present
  Secondary T1 is not present
  DSP is not present

It would almost seem I can do all 24 channels on this, but I haven’t figured it out yet.

Relavent Asterisk configs (sip.conf, extentions.conf, respectfully):


[cisco]
type=friend
context=default         ; Where to start in the dialplan when this phone calls
qualify=yes
host=10.0.3.9           ; we have a static but private IP address
                                ; No registration allowed
nat=no                          ; there is not NAT between phone and Asterisk
canreinvite=yes         ; allow RTP voice traffic to bypass Asterisk
dtmfmode=rfc2833                        ; either RFC2833 or INFO for the BudgeTone
;incominglimit=1                ; permit only 1 outgoing call at a time
                                ; from the phone to asterisk
allow=all


exten => 269,1,Dial(SIP/[email protected],25,rt)
exten => 265,1,Dial(SIP/[email protected],25,rt)

Hopefully this can help get your ConnectReach/VINA Integrator/whateverelsethisthingiscalled working. I don’t have a Digium T1 card to try on there, but this should help configure that.
Also, I have tried to get the Data side of the T1 working, but I haven’t gotten it to work yet. Either I don’t have my clock setup right (the 1760 is totally different than any router I’ve worked on before (only 2600 and 2500))
This has been working with both a wired phone, and a cordless phone.

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