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!
