sundance-communications.com/forum Forums Other TDM Phone Systems Executone-Isoetec Telephones & Systems IDS tie over voip

Hello folx,

I am connecting a 648 w/3.0x9 to a 228 with same sw by using an Epygi t1/e1 box on either side.
Intercom calls are working out perfectly, but when I try to transfer a call from one side to the other, the calling party is disconnected after hearing 1-3 seconds of dial tone from the far end's voip box. The called extension sees the call ringing, but when it is picked up it immediately releases.

I have checked to see what is being sent by the 648 on transfer, it shows # plus last three digits of extn. I created routing tables in the voip interfaces to mimic this and am receiving the right information, but still can't complete a f*#$ing call...

Can't get to vms vie CVM, either.

I have RLS on both switches, line type is 01, drop pulse 20 (just in case), TIE,

It's got to be something stupid, I took my time preparing to do all of the complicated sh*t.

Any thoughts?

try changing line type and set drop pulse to zero for testing purposes. at zero it will not drop calls at all.

Good Luck!!

Which line type should I use? I tried lt 61, but since there's no cid coming across, all I get is a dt from remote side...
If I had hair, I'd be pulling it out.

I would first get a digit graber (Ziad) or a device that can read the hookflash, pause's and any DTMF that is passed between the two sites.
Then send a new post detailing all information.

Can't grab digits on a voip t1. We got most of it figured out. Someone told me that they turned on RLS on the other side, but they didn't program the ctrl-C/N screen properly.

the only issue I'm having now is turning mwi indicators on and off. The digits coming across are not what I expected. I have to tell the voip gateways what digits or combinations to expect before I can see what was actually sent.

I've tried: *763xxx, *76*23xxx, *762xxx, *76xxx, *7623xxx, and numerous other combinations.

I'm losing my mind here.

Go to the Control C "J" screen and look at the DID digits. Also look at the VMS Com Port screen and verify the "systems" are sending and receiving the correct information.

I would, but here's the thing.
The digits sent by the main system to the remote system must first be accepted and relayed by the voip gateway.
If I dial 123456789 and the voip gateway doesn't have a defined route to know what to do with this combination of numbers, it IGNORES it.
(I didn't choose the product, I just have to deal with it.)
In this case, I don't know what to expect, so I can't check the other side to see what's being received.
I know that I need *76xxx to turn on and *77xxx to turn it off. I can manipulate what's sent into whatever i want. i just don't know what's sent.

I'm trying to do the same thing what did you find out you had to do?

I'm trying to do this with Asterisk BTW. Connect a 432 to a 648. Would like to do this:

432 T1 Card -> Asterisk <- IAX -> Asterisk -> 648 T1 Card

Make sure both sets of lines are in the same trunk group. Turn on rls in CTRL-C /N screen there's a few other things you have to do, I'll pm you with my cell number.

I ran into similar issue a couple of years back. The * was getting stripped off by the cisco converter, so we set up the route any for 76xxxx or 77xxxx and inserted the * at the other end. Hope this helps.