Avaya Definity -> Adtran T1 CSU -> Sangoma T1 -> IP PBX

[sixteamparlay]

How can we enable the physical link on our Definity to connect to the external T1 line linked to our IP PBX.
We also would like the Avaya to forward any calls to extension 7700-7899 thru the T1 connection to the IP PBX

 

[Midstream]

What brand is your IP PBX? There may be specific integration notes at support.avaya.com

Do you have the ability to use IP trunks?

Do you have an idle DS1 Interface circuit pack/blade?

You also want to search this forum for UDP threads. It's a bit of typing and others have done some great jobs, so I'm not going into it again.

 

[sixteamparlay]

The Avaya cannot use IP Trunks, we do have a DS1 interface card hooked up to the AdTRAN T1 CSU. How do we activate the DS1 physical link to adtran and what other steps basic things need to be setup on the Avaya. I'm trying to put a list together to better explain to the avaya partner to route 7700-7899 thru the new t1 to the IP pbx

thank you

 

[PeteBull]

sixteamparlay - You are asking a very broad question that has a variety of potential answers (some of them mutually exclusive). Try using this link to get you started, then let us help you when you are stuck on something specific.

 

[sixteamparlay]

Thank you, I see alot of info on how to setup the trunk, but what is the command i should be looking up to activate the physical link on the ds1 circuit pack?

thank you

 

[Mitch672]

there are 3 basic steps:

"add ds1 01A10" (or "add ds1 001V424" if you have a G style media gateway)

"add trunk group x"

"add signaling group x"

"change route x" (pick a new route for testing, put in the trunk group #), then "change ars ana 1xxxyyyzzzz" to put in a specific # you will be testing for outgoing.

that it, but there is a lot to fill in for those forms, not to mention UDP & AAR etc.

Mitch

AVAYA Certified Specialist

 

[sixteamparlay]

Thank you for those steps it will be helpful. Can I add the ds1 without interfering with anything else on the phone system?

I suspect I can add as many DS1's as I like but they won't be used for anything until they are associated with a TRUNK GROUP?

thank you

 

[Mitch672]

they won't do anything outgoing wise, until the trunk group is put in a "route". try "list ars ana", and "list route", you will see.

mitch


AVAYA Certified Specialist

 

[sixteamparlay]

What about incoming?

If the other PBX starts dialing extensions will the avaya try to route them as soon as I activate the DS1?

thank you

btw do you happen to have a screen shot of what the DS1 circuit page might look like in this basic situation?

 

[sixteamparlay]

Turned on DS1 circuit to Sangoma T1 card

The sangoma shows LAPD sync failed "Layer 2"

Does anyone have any experience in getting these T1 cards to talk to an Avaya DS1? Is there a circuit ID or something that needs to be established on both ends in order for the communication to take place?

thank you!

 

[Mitch672]

you need to match the parameters of the 2 circuits

Linecoding: B8ZS, ami-basic, ami-zcs, hdb3?
Framing Mode: esf or d4?
Bit Rate: 1.544 most likely, unless you are using e1, which would be 2.048
Signaling Mode: isdn-pri? common channel? robbed-bit?

not to mention a myriad of other settings on the DS1 form. Are you trying to connect a PRI, if so, the answers are most likely:

B8ZS, ESF, isdn-pri, connect: maybe pbx if you are emulating a CO. You most likely also have to build a T1 crossover cable (pins 1+2 to 4+5, and pins 4+5 to 1+2)

Then you have to get the signalling channel up first, until that works (in-service/idle), don't bother with anything else.

Mitch



AVAYA Certified Specialist

 

[sixteamparlay]

The T1 is another PBX but we have it set to ISDN-PRI, the other settings are identical on both ends (B8ZS, ESF, idle codes).

The Sangoma side seems like it is connected to a loopback since it receives almost the same amount ot data as transmitted. I have a crossover cable setup but perhaps I need a straight through. I will try that one next.

Thank you

 

[sixteamparlay]

OK thanks to your help the DS1 circuit from Avaya Definity to IP PBX is GREEN and has 23 IDLE voice channels.

We set our Avaya route plan for 4 digit numbers starting with 77 to be routed through this trunk #5, however when we dial 7700-7799 we get a fast busy signal.

Could you provide some insight on how we can trace what is actually happening when a user dials 7700 why it does not go down this new trunk?

Thank you!

 

[Mitch672]

Did you do a "change uniform", and put in "77", 4 digits, aar? then in the "change aar ana" you also need to put in the "77" 4 digits, and the aar route for that trunk group...

Mitch


AVAYA Certified Specialist

 

[sixteamparlay]

No we didn't. I'm afraid changing the plan to that would prevent our Avaya extensions 7500-7699 from being dialed. Right now uniform is setup for 4 digits starting with 7 are ext.

Do I need to choose another range like 8xxx ?

 

[Mitch672]

don't change the dialplan, do a "change uniform" (or change udp on older switches), just send 77XX to the new route/PRI

Mitch


AVAYA Certified Specialist

 

[Koplin]

I am dealing with something very similar currently at our location. We have a Definity G3 something and unfortunately we are not licensed for isdn/PRI ability so we were stuck with a "super trunk" aka AMI D4 type.

Our choice of integration for IP PBX's is Asterisk

We have one T1 from our provider going to our Asterisk PBX with a quad T1 digium card. We then slave the Definity off of Asterisk (using a cross T1 cable). Our DS1's are setup robbed-bit 1.5meg, AMI-Basic with E&M wink we had to changed our synchronization (before this step our operator was getting lots of calls with no audio as the line was not in sync with the network and so would generate a huge number of calls due to the mismatch.

I think the command was "change sync"

We then added a Trunk Group with the line setup as a WINK/WINK (trial and error mostly as its hard to find exactly what both asterisk and the Definity liked) gave it a TAC and assigned the channels. (I know a PRI would help here and help eliminate all most 10 seconds delay dialing from the Definity as the line is signaled)

we then setup a new route added in our newly created trunk and pre-appended a 9 so that asterisk would be happy
we changed ars analysis for our chose extension range's

we have 5 other Asterisk boxes tied in via IP using IAX trunks (but that's a different story)

On the asterisk side we had to change the /etc/dahdi/system.conf file to reflect our selected signaling type on the T1 and reload the drivers(or reboot system) in our case
IE
"
span=2,1,0,d4,ami (this is our provider)
span=1,0,0,d4,ami (this is our Definity)
e&m=1-48 (E and M signaling on both T1's)

;spans 3 and 4 are as yet unused"

The other file need on our AsteriskNOW install was a change to /etc/asterisk/chan_dahdi.conf

"context=from-internal
toneduration=150
signalling=em_w
rxwink=300
txwink=150
usecallerid=no
hidecallerid=yes
callerid=
restrictcid=yes
useincomingcalleridondahditransfer=no
sendcalleridafter=10
rxgain=0
txgain=0
group=0
channel=1-24

context=from-pstn
usecallerid=no
hidecallerid=yes
callerid=
restrictcid=yes
sendcalleridafter=10
useincomingcalleridondahditransfer=no
signalling=em_w
rxgain=0
txgain=0
group=1
channel=25-48
"
We encountered a few problems both with Asterisk but mostly with Definity related shortcomings or lack of knowledge on my part.

1st; incoming calls arriving at Audix do not play announcements/greetings. We had to place the incoming call first in an IVR that defaults in 0 seconds and passes it down to the Definity.

2nd; outgoing calls from the Definity if they are in a 7 or 10 digit format work as expected. But there is something odd with smaller digit sequences. I will admit knowing next to nothing on dial plans for Definity. We setup a new route and added in the trunk even have it adding a 9 to get past asterisk's dial plan's. But if say we choose a number less then 7 then for some reason the definity appends a # at the end of the number so if we call 7000 it dials 7000# on the trunk. After reading this forum I will see if I can change the Definity's uniform dialplan but I am guessing we also don't have that feature.


the command I used was "change ars ana 7" choose min and max of 4 etc. I changed the type to just about eveyone and nothing seamed to help (emmer, local, hpna). I put in the route number created earlier and watched as asterisk picked up the 4 digit numbers along with an extra #. Some nice Asterisk GURU's provided this sniped of code that I will try shortly to eliminate the # unless someone here has a better option (IE as stated above change udp/uniform) (I suspect I lack this option so will be forced to use stripping on the asterisk side.

I hope this helps someone and that some Definity GURU reading this might help me with my little # issue and as stated earlier there is a LOT of configuration dependent issues. IE our choose of line coding and signaling etc.

Thanks

Chris

P.S. Sorry for the long post.
P.P.S In frustration with our Definity management, maintenance costs and lack of easy remote management we chose to implement Asterisk and have yet to look back!