There are several methods of having your legacy Nortel BCM/Norstar equipment communicate with your Asterisk PBX - depending on your needs. The deciding factor will usually be cost. Some of the things you will need to consider include where you want your main internal call routing to occur and what type of phones you want to end up using.

Integration via PRI

Larger BCM/Norstar systems (more than 25 users) will typically integrate with the PSTN using PRI circuits. Integrating Asterisk into a PRI environment can be done by either connecting it in between the C.O. and the BCM/Norstar - like this:

[Central Office] ---PRI--- [Asterisk] ---PRI--- [BCM/Norstar] (requires one PRI port in the BCM/Norstar, and two in the Asterisk)

or by connecting it to the BCM/Norstar like this:

[Central Office] ---PRI--- [BCM/Norstar] ---PRI--- [Asterisk] (requires two PRI ports in the BCM/Norstar, and one in the Asterisk)

The integration of the dial plans would be done on the BCM Norstar by use of Received Digits on Target Lines to deliver calls to the BCM/Norstar, and Destination Codes and Routes to deliver calls to the Asterisk.

PRI would support delivery of name and number, and would allow for a much more seamless integration of the systems than any analog solution can.

See T1 Crossover Cable for cable wiring to connect two PRI interfaces together.

Intergration with SIP

I have managed to get some basic SIP working between Asterisk and a CS1000 (aka. CS1K).

You will need to be running Asterisk 1.2.12.1 or higher. Lesser versions required a small 1 line patch to chan_sip.c available here: http://bugs.digium.com/view.php?id=8010

Also forget about using the Nortel Network Routing Service (NRS) for Asterisk -> CS1K calls. They dont get on so don't try to register Asterisk to it and don't send any calls to the CS1k via it. instead, you need to recreate routing entries in the extensions.conf file. From the Nortel side, after you configure Asterisk as a static SIP endpoint, you will be able to dial CS1K -> Asterisk through the NRS. Lets say you have two CS1K systems with Coordinated Dialing Plans (CDP dialing plans) where CS1K-A has the 2xxx range, CS1K-B has the 3xxx range and Asterisk uses the 4xxx range. You need to add the following configuration to the Asterisk sip.conf:

[nortelout1]
type=peer
host={node ip of first cs1k system - not the NRS!}
usereqphone=yes

[nortelout2]
type=peer
host={node ip of the other cs1k system - not the NRS!}
usereqphone=yes

and in extensions.conf:

exten => _2XXX,1,Dial(SIP/${EXTEN}@nortelout1)
exten => _3XXX,1,Dial(SIP/${EXTEN}@nortelout2)

Also, you need to be able to dial your own extensions:

exten => _4XXX,1,Dial(yourprotocol/${EXTEN}@yourextension)

Meridian/CS1000 integration using H.323 trunking

After giving up CS1000 to Asterisk interconnection using SIP (due to ";phone-context", maddr=XXX redirections and some other wired things), I've got an interconnection using H.323 to work.
Here are some notes:

• You'll get codec-problems (packet/frame-size incompatibilities during OLC-Handshake) with oohH323 and oh323, so NuFones h323 is your friend.
• You can telnet into the Norstal Signalling Server console and do "gkRegTrace ALL" if you need debugging information

But first, before you attempt this internetworking configuration, you need to do a little patch to the h323 channel driver:
In channels/h323/ast_h323.cpp you have to replace the single-quote at NuFone's, because this character is invalid for CS1000 (you'll get LOG0003 GKNPM: SOLID SQL Error 1: syntax error (line 1 near 'Network'S') at CS1000 console).

In h323.conf use the Signalling Server IP address as the gatekeeper. Asterisks' endpoint name, as configured in the CS1000 NRS, has to be used as the H.323 alias in h323.conf, not the name of the CS1000! Then you have to setup a CDP entry (of type DSC) in overlay 87 with your prefix pointing to the RLB (overlay 86), which in turn points to the route with the H323_IP_TRUNK's (but this isn't Asterisk specific, it's general NRS setup...)

Eventually, you can just Dial(H323/12345,45,t) in your extensions.conf

BCM integration using H.323 trunking

I have a friend who claims to have done this (yes, I believe him!). He says the trick was to set the system type to "Other" when chosing the H.323 trunking protocol (all the other protocol choices are Nortel-proprietary).

Integration via Analog Lines

Most likely, your Norstar PBX will have external lines for incoming and outgoing calls. You could run one of those lines to an FXS port on your Asterisk server (like a Digium TDM400P) which supplies dial tone or in larger systems, you might have a Norstar T1 Truck Cartridge which could connect to a T1 card on the Asterisk server (like a Digium TE410P, T100P, or E100P). When the Norstar system uses that line, it doesn't know what it's connected to - it naturally assumes a Central Office. For calls from the Norstar PBX to Asterisk, as far as Norstar knows it picks up a line, receives dial tone, and dials a number. Likewise Asterisk doesn't know what's connected to its FXS port. As far as Asterisk knows, something picked up a line, Asterisk gave out a dial tone, then received back some digits. The reverse is true for calls to the Norstar PBX. You can configure the Asterisk FXS port to use any context you like to get IP phones, outside lines, or a VoIP connection. You'll probably want to change the dial plan in the Norstar system so that integration line is in a special pickup group. One problem with this approach is that in a Norstar system running versions prior to 4.1 or so of the software, it isn't easy to forward an extension to an outside line, which means Norstar phone users will have to remember to do something different when they want to call a user who has been switched to an IP phone for example.

It's important to consider, also, that the Norstar FXO-type lines will often not disconnect without some sort of Disconnect Supervision (ie. an open loop). Thus, for example, if someone were to dial a Norstar extension in Asterisk, and then immediately hang up, the trunk would ring forever or until the voice mail system picked up, as Norstar would believe the calling party was still connected, and would continue providing the ring to its extension.

Integration via Extensions

Another approach, which works well for phasing in an Asterisk solution, is to interface the two systems over extensions rather than lines.

Nortel Meridian/Norstar digital PBX phones or extensions, such as their popular M series, are meant to use a propriety protocol to communicate with a Nortel/Norstar PBX. As such, it can be very difficult and costly to get those phones to communicate directly with an Asterisk server. In theory it would be possible using something like the Dialogic/Intel D/42-NSC board. This card communicates via the proprietary Norstar protocol that the Norstar digital phones understand over 4 ports, but the drivers for an Asterisk application of this product have not been written and it's quite a costly card.

A more feasible solution then is to attempt to communicate with Asterisk over an analog extension.

You can convert a single Norstar digital extension into an analog extension using a Norstar Analog Terminal Adapter (part#s NT8B90AL or NT8B90AC). One port on this box connects to where a Norstar digital phone used to be (digital extension) and the other port takes a standard analog phone.

For more than a few extensions you could use a Norstar 8 Port Analog Station Module (part#s NTBB51CA or NTBB51CB). Same concept here, the analog station modules let you plug in a standard analog phone.

With either of these devices, the analog extensions are giving out dial tone so you'd connect each extension to an FXO port on the Asterisk server or in larger systems, a channel bank connected to a T1 card. For calls from the Norstar PBX to Asterisk, as far as Norstar knows, it rang a digital extension. As far as the analog terminal adapter or analog station module knows, it rang an analog phone. As far as the Asterisk server knows, its FXO port rang so it picked it up. The reverse is true for calls to the Norstar PBX.

You can configure Asterisk's FXO port to immediately dial something like an IP phone, which allows for seamless integration of the two systems. A legacy Norstar user simply dials an internal extension as before and it rings on an IP phone. Likewise an IP phone can directly dial Norstar phones.

You can map more IP phones to Norstar extensions without purchasing an analog terminal adaptor for each one. You can forward several Norstar extensions to the same analog terminal adapter and subsequently, an Asterisk FXO, but the analog terminal adapter doesn't tell the FXO which extension was dialed so the Norstar dialer must re-enter the extension they dialed or follow prompts once they hit the Asterisk server.

You should know that most of the Nortel phones cannot send DTMF tones to the FXO interface in Asterisk. You need to implement "Long Tones" (feature 808) before Asterisk will act on the tones so you can dial o use something like Voicemail.

Voice Mail Interface

There is also a Norstar Voice Mail Interface device (part# NT8B89DA). At the time of writing the only information that could be found was:

"The Norstar Voice Mail Interface (VMI) allows the connection of a third party stand-alone auto attendant/ voice mail system to a Norstar system (DR2 or higher). This interface provides the basics of integration; forward to personal greeting, return to operator, and message notification. Each VMI supports 2 ports on the AA/VM device and uses 2 station ports on the Norstar. Up to 10 VMI units can be used on a Norstar to interface an AA/VM system."

The VMI is basically a two port analog terminal adapter that provides disconnect supervision and signalling via DTMF. It is no longer manufactured and thus can only be found on the secondary market.

Citel portico(tm) Telephone VoIP Adapter (TVA)

You can connect Nortel phones (and others) through a Citel portico Telephone VoIP Adapter (see http://www.citel.com/Products/Portico.asp). As mentioned above in 'Integration via Extensions', Nortel phones use a proprietary signalling protocol to communicate to the PBX. The Citel portico TVA can convert between this protocol and SIP signalling, thus allowing you to host the Nortel phones on a SIP PBX such as Asterisk.

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In developing your integration plan you must balance how much you want to invest in legacy hardware and how large a scale the integration needs to be.

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Nortel Phones
Asterisk unistim channels

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