Brooktrout SR140 configuration notes

[VoipMenace]

Hi ,
anyone out there configured an SR140 with a CM4. Outbound calls / faxes work, inbound does not work at all

any notes or gotcha's are appreciated

 

[1a22ip]

Are you using UDP to route the calls to the Brooktrout trunk group? What kind of transport (Analog, Tie T1, PRI) between the Brooktrout and the CM?

ED

1a2 to ip I seen it all

 

[Midstream]

Hey Menace,

I've done it on my S8720 running this release:

R014x.00.1.731.2

One of my neighbors is a Captaris Business Partner and asked if I'd hook his server up to my CM4 to test. We got it working in minutes, both inbound and outbound. I've heard of some other problems with this though. I've got the integration docs around here somewhere if you don't have them.

Phil

 

[atwebb]

Rockspop any chance you can share the config screenshots for the CM and the ip setup for the SR140. I can't get outbound to work.
The udp table and routing in and out would help as well as the sig and trunk groups,but particularly a working config of the sr140 woulh help me out.

 

[Midstream]

Here's the Signal Group:

Voice System name: Cronus - SIGNALING GROUP

Group Number: 5 Group Type: h.323
Remote Office? n Max number of NCA TSC: 0
SBS? n Max number of CA TSC: 0
IP Video? n Trunk Group for NCA TSC:
Trunk Group for Channel Selection: 5
TSC Supplementary Service Protocol: a
T303 Timer(sec): 10

Near-end Node Name: S8720_CLAN Far-end Node Name: RIGHT_FAX
Near-end Listen Port: 1720 Far-end Listen Port: 1720
Far-end Network Region: 5
LRQ Required? n Calls Share IP Signaling Connection? y
RRQ Required? n
Bypass If IP Threshold Exceeded? n
H.235 Annex H Required? n
DTMF over IP: out-of-band Direct IP-IP Audio Connections? y
Link Loss Delay Timer(sec): 90 IP Audio Hairpinning? n
Enable Layer 3 Test? n Interworking Message: PROGress
DCP/Analog Bearer Capability: 3.1kHz

ADMINISTERED NCA TSC ASSIGNMENT

Service/Feature: As-needed Inactivity Time-out (min):
TSC Local Adj. Mach.
Index Ext. Enabled Established Dest. Digits Appl. Name ID
1: n
2: n


And here's the trunk group:

Voice System name: Cronus - TRUNK GROUP

Group Number: 5 Group Type: isdn CDR Reports: y
Group Name: Right Fax COR: 92 TN: 1 TAC: 705
Direction: two-way Outgoing Display? n Carrier Medium: H.323
Dial Access? n Busy Threshold: 255 Night Service:
Queue Length: 0
Service Type: tie Auth Code? n
Member Assignment Method: manual

Group Type: isdn

TRUNK PARAMETERS
Codeset to Send Display: 6 Codeset to Send National IEs: 6
Charge Advice: none
Supplementary Service Protocol: a Digit Handling (in/out): enbloc/enbloc


Digital Loss Group: 18
Incoming Calling Number - Delete: Insert: Format:

Disconnect Supervision - In? y Out? n
Answer Supervision Timeout: 0

TRUNK FEATURES
ACA Assignment? n Measured: none
Internal Alert? n Maintenance Tests? y
Data Restriction? n NCA-TSC Trunk Member:
Send Name: y Send Calling Number: y
Used for DCS? n Send EMU Visitor CPN? n
Suppress # Outpulsing? n Format: public
UUI IE Treatment: service-provider

Replace Restricted Numbers? n
Replace Unavailable Numbers? n
Send Connected Number: n
Hold/Unhold Notifications? n
Send UUI IE? y Modify Tandem Calling Number? n
Send UCID? n
Send Codeset 6/7 LAI IE? y

QSIG TRUNK GROUP OPTIONS

SBS? n


QSIG Value-Added? n

TRUNK GROUP
Administered Members (min/max): 1/2
GROUP MEMBER ASSIGNMENTS Total Administered Members: 2

Port Name Night Sig Grp
1: T00087 5
2: T00088 5



I don't have the screen shots for the Right Fax set up, the server is no longer connected to my network. I followed the integration docs to set up the signal group and the trunk group. The whole setup took about 20 minutes to complete, if that.

 

[phonegoober]

Hello All,

quick question.

I am kinda in the same mode right now. Have the faxination product. Looking at how to setup on our PBX. It is setup as a Tie, but i am trying to figure out how to get over to the Tie trunk group from a DID trunk group. Right now, we have each fax line setup as a VDN that routes to a vector that inserts the TAC Of the trunk group, but there has to be a better way.

 

[Midstream]

Goober,

Time to sharpen up your UDP skills. If your switch is networked to any other switch, use

display uniform 1

display aar analysis 2

display route X

Try to decipher what you see there and how calls are directed to particular trunk groups based on digit ranges and aar codes.

 

[phonegoober]

thanks Rockspop,

so i have to put in every Fax extension in UDP 1?
Also, on the fax extensions, i need no coverage path as well? thanks.

 

[Midstream]

Not every extension, you can cover a range of extensions if they are continuous.

There won't even be a physical or logical extension to put a cover path on. You program the extension number in the aar analysis/uniform dial plan form to exist "over there", on the other side of the trunk group that ties to your fax server.

 

[phonegoober]

Thanks rockspop,
we don't have continuous extensions.
What about i just create a new coverage path, have it point to a new extension, then have AAR/UDP use the digits of that new extension and point it to the trunk group?

 

[Midstream]

That's some serious "out of the box" thinking there Goob. Give it a shot and let me know what happens!

How many extensions do you have?

 

[phonegoober]

rockspop,

we have about 300 extensions and they all are currently vdn's that have their own vector. That's a big pain to manage. I can give it a whirl. If it works, then i only have one entry in UDP versus 300. I will let you know if it works.

thanks a bunch.

 

[phonegoober]

rockspop,
i am getting a busy. My stations are defined with an
X port. I dialed cov pat ext and got a busy. if i remove the station from the pbx and just use the digits in UDP, it works.
thoughts?

 

[Midstream]

Thoughts?

Get a clerk to start typing! =)

I can't get my head around it right now, but what you're trying to do to shortcut the technology isn't making sense to me. TGIF.

 

[phonegoober]

Thanks. Even if i put the individual extensions in UDP, i still get a busy. The test station is defined as a FAX with an X port, no coverage..

 

[1a22ip]

That is the way is supposed to work. If the extension is in translation the system will use this first. If it cannot find the extension it will look in the udp table. If found off it goes. Always remove the extension from Translation. Another thing is to keep a list of UDP extensions. The system will allow the extension to be assigned even if it is in the UDP table. I instruct my Tech's when adding an extension the first command is to change the extension. If the extension is in the Udp table the system is give a message that the extension must be local.

Hope this helps
ED

1a2 to ip I seen it all

 

[Midstream]

Delete the station. You can't have a local station defined AND have a UDP record for a station on a networked system.

 

[phonegoober]

thanks rockspop,

i am going to have to come up with another way then. I have to have our extensions programmed for billing purposes. our call accounting system polls our PBX for valid extensions and matches that up with billing, department number.etc.
looks like i am going find another way such as defining the Tie Trunk ports as a station (DS1).

Or create a hunt group, assign those Tie Members in the group with a lead ext. Then point the stations to the cov path that points to that lead number.. There's several different ways, i am sure..

 

[phonegoober]

Rockspop,
i pulled a channel off our our existing setup that is defined as tie (i pullled channel 24).

I then defined a new test station as X11111, using port 01d1724, and station type of DS1FD. I called this extension and it is just doing a ring no answer.
I don't see my test call hitting the TAC.

Do i need to go down the road and setup a new hunt group, agent ID, AAS, etc?

 

[Midstream]

Stop everything.

Go get this 16 page manual that spells it out.

http://www.avaya.com/master-usa/en-us/resource/assets/applicationnotes/sr140-h323-acm.pdf

Have a nice weekend, rockspop is off to Happy Hour!