VOIP Solution? 1

Im not using this setup, but if it's like any regular connection, it should use Robbed Bit signaling, ESF framing, and B8ZS Line Coding to connect to standardized devices like routers. That's a pretty serious router, so CISCO would be the ones to contact regarding what T1 properties you'd need for VOIP, but it absolutely CAN be done!
E&M (wink-start)is another common way to negotiate Voice over digital lines.

There's too little information given to help much further - sorry.
Hope this helps a little though!

Thanks! This should get me started. I am going to have a tech from cisco take a look at it also.

Thanks again....if you think of anything else,I'll be checking back.

Make sure the T1 card is the HDV-1T1 module. The HDV has the necessary DSPs. Hope that helps.

agoldberg@phonextra.com

Kelle
Hope the following conf. would help you. I have the following defined in my switch which connects with CISCO.

========================================
DS1 CIRCUIT PACK
Bit Rate: 1.544 Line Coding: b8zs
Line Compensation: 1 Framing Mode: esf
Signaling Mode: isdn-pri
Connect: network
Country Protocol: 1
Interworking Message: PROGress Protocol Version: a
Interface Companding: mulaw CRC? n
Idle Code: 01010100
DCP/Analog Bearer Capability: 3.1kHz

Slip Detection? n Near-end CSU Type: other
ESF DATA LINK OPTIONS

Network Management Protocol: tabs
Send ANSI-T1.403 One-Second Performance Reports? n
Far-end CSU Address: b
================================================
TRUNK GROUP

Group Number: Group Type: isdn CDR Reports: y
Group Name: COR: TN: 1 TAC:
Direction: two-way Outgoing Display? n
Dial Access? y Busy Threshold: 255 Night Service:
Queue Length: 0
Service Type: tie Auth Code? n TestCall ITC: rest
Far End Test Line No:
TestCall BCC: 4
TRUNK PARAMETERS
Codeset to Send Display: 6 Codeset to Send National IEs:6
Max Message Size to Send: 260 Charge Advice: none
Suppl Service Prot:a Digit Handg (in/out): enbloc/enbloc

Trunk Hunt: ascend
Connected to Toll? n STT Loss: normal DTT o DCO Loss: normal
Calling Number - Delete: Insert: Numbering Format:
Bit Rate: 1200 Synchronization: sync Duplex: full
Disconnect Supervision - In? y Out? n
Answer Supervision Timeout: 0
======================================================


Regards

Thanx-I am going to look into this config. and see what I can come up with....I spoke with the techs from Cisco and they weren't any help.

We've done it using a 2600 router.

ds1 board config
robbed-bit, esf, b8zs, leave the defaults

trunk group
wink/wink with defaults
cisco router
e&m wink start on the T1 controller
Ours works pretty well, just trying to figure out the best way to integrate Definity Audix.

Chris

Cwarren,

I'm going to end up having a 2620 router on both ends of my WAN... There's a Defintiy G3si on both ends as well. What module do I need to buy for the 2620 routers?

Thank you for your help.

Let me add a little more to my question... do you even have to by a module (such as NM-HDV-1T1-24) or can you just use a VWIC card? I'm a little confused as to how it all fits together.

Thanks agan.

YOu do need to get the NM-HDV module. The VWIC cards go inside of this. You would then need a VWIC-1MFT-T1. Connect this to a DS1 card on your G3. You will also need a 356A adapter for the PBX. This is an 8 port adapter that you will use to connect the router to the PBX. Use port # 8. Hope this helps.
Chris

Hi,

Sorry but I have no answers, only questions.

Has anyone in the UK done this? i.e. PBX to PA-VXC-2TE1+ using E1.

We have on Lucent Definity:

Bit rate 2.048
Signaling mode CAS
Line Coding HDB3
Country Protocol 10
Interface companding alaw
Idle code 11111111
CRC n
Slip Detection n

Router has PA-VXC-2E1+ and:

card type e1 2

dspint DSPfarm2/0

controller E1 2/0
framing NO-CRC4
ds0-group 0 timeslots 1-31 type e&m-wink-start


voice-port 2/0:0
!
dial-peer voice 2081 pots
destination-pattern 44208.......
forward-digits all
!
dial-peer voice 2071 pots
....more of these...

The thing is that the controller goes up and down all of the time.

I am particularly concerned about the country code. The voice switch is set to 10? What does this mean? Can this being miss-matched cause the controller to fail to come up?

What is the equivalent of 10 on the Cisco?


E1 2/0 is down.
Applique type is Channelized E1 - balanced
Receiver has remote alarm.
alarm-trigger is not set
Framing is NO-CRC4, Line Code is HDB3, Clock Source is Line.
International Bit: 1, National Bits: 11111
Active xconns: 0
Data in current interval (137 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 99 Unavail Secs


Ideas welcome, I am baffled.

Thanks.

Number7seven,
The country protocol on the voice switch is to mention the signalling bits , Country code 10 on voice switch is set for UK. However you can try with some other settings as well perticularly when you are connecting to CISCO.
The signalling bit would specify the ON hook & OFF hook bits. Following are the details about the various country code alongwith the signalling details.

Country code : 10
Status Forward Backward
IDLE(ON HOOK) 1101 1101
SIEZE(OFF HOOK) 0101 1101
ANSWERED(OFF) 0101 0101


Country Code : 7
Status Forward Backward
IDLE 1001 1001
SIEZE 0001 1001
ANSWERED 0001 0101

Country Code : 18 (You can try this as well
Status FORWARD BACKWARD
IDLE 1111 1111
SIEZE 0111 1111
ANSWERED 0111 0111

These are the ABCD bits for the CAS signalling. hope this might help you while checking in different country codes

Specifically for No7. - we certainly do it using the same method. The country code <I>Is</I> correct for the uk but we set ours to 1. Also your idle code is incorrect, it should be 01010100.

Hey,
Just wanted to post on this message because I have almost the exact same setup as listed in the beginning of this post. Almost everything works fine. The problem is I can't seem to get the PBX to pass the gateway (3640) the calling party number correctly. Basaically, When I place a call from my PBX phone to my Voip phones 99% of the time i get (9) as the number that shows up for caller id on the VoIP phone. 1% of the time I get some random number like (668),(3019832380), or (2026289775)... why isn't this working. I'm thinking its pulling the (9) from the dialpeer setup on the 3640.. The caller ID does work the other way. I need help zeroing in on this issue. I need to figure out what the correct configuration would be to pass the &quot;calling party&quot; number or extension down the PRI to the voice gateway. Any help is much appreciated. Thanks in advance.

Cold

Hey Cold (or anyone with a config to share)

Can you elaborate with a router config for your setup? I'm also starting a project very similar to the beginning of the post.

Thanks,
JHC