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Tecman: Not yet. It has something to do with aar ars as far as I can tell. Now what I CAN do and it does work is forward my phone to, catch this, 98xxxx00 where xxxx is my asterisk extension. It will not hang up if I do it this way. I just chop off the two end digits on the asterisk side...

Nope. Remote system is Asterisk (getting our feet wet before committing to a bigger voip system). Just 24 regular old voice channels. I was wrong actually... it will hang up on me whether I answer the phone or not. It somehow doesn't realize a valid call is taking place? -Matt

Calling out over E&M wink tie line. Far end sends ringing tone back, but the Definity hangs up with the classic two tone warble approximately 10 seconds after the last digit is dialed. I changed a lot of the answer supervision values to no avail. If I answer the phone on the other side then...

I have a tie line setup to another pbx (not Avaya). I am receiving *ANI*DNIS and that works fine INCOMING on my tie line. It is not, however, sending that information on an outgoing call. It simply sends the digits to dial. The feature I turned on on page two of the trunk group: Incoming...

Just to update everyone...checked over everything I can possibly imagine. Verified my Asterisk configs with the manufacturer of the T1 interface card. I ditched the a120 csu in favor of a direct crossover cable..same results. Tried different switchtypes (national, old national 1)..different...

RTMCKEE: It is actually a 120a2 CSU. My next step is going to be a direct wiring with a crossover cable just to test. This is my only space CSU presently and in an unknown condition. It is set for Network clock source on the Definity side. The other side is supposed to provide clock, but I...

okay... this could well be the case. Basically it is to attach an Asterisk phone system on to ours. My boss wants to evaluate it for a few special cases. I'm going out to a CSU 120 module. This essentially is wired with a crossover cable to the T1 interface in a server running Asterisk. One...

Okay... that I can do... I reseated the tn464f card in the system. I am using the csu module attached to the back. port maintenance name alt. name result err. code 01b1624 ISDN-LNK ABORT 1011 That is my result from release port 01b1624 Trying to...

unfortunately I don't have that command... It's old V6 software on this switch...maybe that was a later command. I'm wondering if I have a timing issue going on perhaps. It's weird! thanks for the advice though, -Matt

Okay... my second question of the week. I have created a new signalling group. When I view the status I have Group ID: 4 Type: facility associated signaling Group State: out-of-service Primary d-channel link: Port: 01b1624 Level 3 State: manual-out-of-service Do I need to somehow...

Hello all! I'm in the midst of doing some non-standard stuff with our definity. I am going to create a tie line, full 24 channel T1. To get extensions to our other tie line we use UDPcode 221. My question is...I've got the steps for almost all of the rest of the equation EXCEPT the 221 part...

I have defined a group page for about 10 phones. The group page has been busy for two days... I can't seem to figure out how to release it or even tell what's going on. Any advice would be great! Sometimes I feel so helpless *grin* -Matt

Great! I am almost positive I already have some of the a120's around that I can use. If not... the Adtran isn't a big deal to reprogram. Thanks again for the info! -Matt

Currently we use an adtran tsu 600 to multiplex a line. One of these goes to our router, one goes to the phone system (channels 13-24) Basic questions on the tn464f card... will this interface directly with the smartjack? I don't need the data channels anymore and eliminating the Adtran...

We are moving a remote proligix R6 system to a new building. They will primarily use a trunk to our building down the street for all calls. We are putting in some POTS lines at the new location for 911 and emergency calling if the link between us is broken. Is the TN747 the right card to use...

Thanks guys! I already have ample 110 blocks, just no cards in the slot. Plenty of analog cards, not enough digital. I'm actually cross connecting to one of those MCK PBX Gateways within a month or two. Branch office down the road. I didn't realize they were hot swap... very cool! Thanks...

Hopefully this isn't too beginner a question. Definity G3R6 I need to add another digital circuit pack. What all do I need to do? shutdown system and install card or what? Heck I don't even think I know how to shut this thing down. Never had to add a card before. Thanks for any help! -Matt

Thanks for all the replies! I only have 5 outstanding POTS fax lines that I need to route through the switch so I will probably just configure them in the trunk config screen. I tried it yesterday and it worked like a champ. We have 250 DID #'s already and remove the first 3 digits by...

Our former phone system admin retired so here I am trying my best. We have some old POTS lines coming in to the building for fax machines etc. and we'd like to move them to our switch. (a G3siR6) The phone company can send them to our switch, but how do I go about routing them there? I...