sundance-communications.com/forum Forums TDM Business Telephone Systems Toshiba Telephones & Systems Integration Questions

I have been tasked with researching something for one of our customers that is a little out of my comfort zone so to speak.

High level scope:

Traditional Side:
Customer has an existing StataNet network of CIX/CTX systems and they are currently sharing a PRI circuit with some copper trunks scattered in there.


SIP Side:
They also have a Cudatel 470a system in use being fed by an emergency call out system. This side is using 100 SIP trunks.

The customer would like to leverage the SIP trunks for use with both systems as the PRI circuit is being deleted.

I have a couple of questions to start with.

1. Does anyone have any experience linking Toshiba systems to other systems using a Gateway such as the Digium G100.

2. If this is possible, would it be better to leave the interface on the Toshiba side as PRI or look into QSIG? The Digium gateway will accept either.

I should add that the next thing they are looking at is set to set calls between the 2 systems but that is secondary to getting the Toshiba's using the SIP trunks.

Any input would be greatly appreciated.

Soooo, we dove head first into this last night!

Results:

We now have a Stratanet network sharing a SIP trunk service being managed by a Cudatel SIP system.

The main CTX is linked to a Digium G100 gateway via the existing PRI card, this gateway is directing traffic to the Cudatel which is managing the SIP trunk service. All inbound and outbound traffic is routing as it should with very little work done on the strata side.

I have also been able to route station to station calling between systems, still a few bugs to work out with this part but I have ringing sets with 2 way voice.

A few things of note on the Digium config:

In the SIP endpoint configuration make these selections in Call Settings > Caller ID Settings:
Trust Remote-Party-ID = Yes
Send Remote-Party-ID = Yes, send Send Remote-Party-ID Header
Caller ID Presentation = Allowed, not screened

In the T1/E1 configuration for the Port:
Signaling > Switchtype = National ISDN 2
Advanced Signalling > Enable Caller ID = Yes

It seems we are making headway with this project.

Node access from the SoftIPT is working as it should now, eliminating some dialing delays experienced from remote nodes to the Digium.

I think we are getting close to making this a fairly good integration even if it does have it's limitations.