H323 trunk in openscape 4000 2

[bitapn]

Hi all.
i have a openscape 4000 ver 7.
i need to connect it to another PBX by using a IP trunk via H323.
I need to know the commands for configuring the H323 trunk in openscape 4000.
could you please help me and make me aware of the trunk details which i need to set in TDCSU and LWPAR or any AMO related?

 

[Iamnothere]

H.323 trunking is no longer supported on the OpenScape 4000 V7 - except if used to provide the voice path to Xpressions (Unify voice mail). You can use SIP trunking, for which the configuration is 90% similar to that of H.323.
You will need a physical STMI board, or if using an Access 500/Softgate server - a "virtual STMI".

I do not have time to re-create every possibility or each tiny step, so here are the BASIC steps:

Use AMO BFDAT to configure the function(s) required on this STMI board. To support SIP Trunking, the BFDAT "function block" (which is a fancy word for "index number"), must include the function "HG3550". This series of BFDAT commands may be tricky. To create a BFDAT entry, you must first perform ADD-BFDAT to create the "function block", assign the function(s) such as "HG3550", determine which size STMI board may be used for this situation (either 60-channel or 120-channel, or BOTH), and specify whether this STMI board will be installed on a standard shelf (using AP3700 hardware) or a SOFTGATE (Access 500 or server-based IPDA). If using AP3700 hardware, then this last ADD-BFDAT parameter is "ATTR=NONE". If using Access 500 or Softgate/server-based IPDA, then "ATTR=SOCO".
Now that the BFDAT "function block" is created, you must continue the process by specifying exactly what you intend to create using CHA-BFDAT. After entering the function block number, specify the Function to be further-defined, such as "HG3550". Three very confusing parameters will follow. First parameter = "LINECNT": this parameter represents the number of circuits to be created. SIP Trunking circuits are based on the international "E" carrier with 30 B-Channels. Therefore a 60-channel STMI may support TWO 30-Channel SIP Trunking circuits. Next parameter is "UNITS": to assign 30-Channels to each circuit, enter a "3" here, as there is a behind-the-scenes multiplying factor of 10, where an entry of "3" really means "30". The next parameter, "BCHLCNT", is ignored for HG3550. Example 1: To create ONE SIP Trunking circuit with 30-Channels each, then "LINECNT=1" and "UNITS=3". Example 2: To create TWO SIP Trunking circuits with 10-Channels each, then "LINECNT=2" and "UNITS=1". And finally, close the BFDAT configuration with a final CHA-BFDAT. Indicate that the BFDAT configuration is "OK", and "YES" = ready for processing. Now the BFDAT function block is ready to be assigned to one or many STMI boards.
Next, use AMO BCSU to add the STMI board to the system and assign the previously-built BFDAT "function block" to the STMI. Unless the function HG3570 is also built in your BFDAT function block, the AMO BCSU parameter "IP Address" will be ignored; therefore, use AMO CGWB to add the board's IP Address, netmask, default router, and other board-specific parameters. Also in AMO CGWB, you must assign ALL of the trunk channels to one specific trunk protocol, such as parameter "TRPRSIP" - which represents Native SIP. Example: If creating ONE SIP Trunking circuit with 30-Channels, then parameter "TRPRSIP=30".
This completes the board setup. You may now insert the STMI board, or if already inserted, you must now reset the STMI. Ensure that LAN Cable is plugged in (if physical STMI board).

Now you may begin to build the trunk circuit on the STMI, starting with AMO BUEND (trunk group), AMOs COT & COP (trunk settings), then AMO TDCSU to build the trunk circuit(s). In AMO TDCSU, there are some critical parameters:

PROTVAR=PSS1V2 (in the USA) or PROTVAR=ECMAV2 in most other countries.
SEGMENT=8
DEV=HG3550IP (for a Point-to-Point trunk circuit like this one)
SRCHMODE=DSC (I do not recall the exact parameter name, but it "means" SearchMode! Value should be "DSC" = Descending)
BCHAN=1&&30 (enter all of the B-Channel numbers - not just a "30", as this would enable ONLY
TGRP= the trunk group number that you created using AMO BUEND
INS=Y (circuit is UP by default. No ACT-DSSU command is required to activate the circuit)

Then you may proceed to configure the call routing. Assuming that you will be using an Access Code such as "8" to start the dialing process for these SIP Trunks (when using a trunk access code, this type of dialing is known as "OPEN DIALING"), the configuration process may involve AMO WABE to configure the trunk access code - if needed- such as "8", AMO RICHT to create the actual call route, and for "permission" to use the trunk group via this new route. Then AMO LODR is used to create a set of Out-Dial Rules (determines digits/info to be sent to the far end system). If the user dials "8" plus a 4-digit station number at the far end, then typically only the 4-digit station number needs to be sent to the far end via the trunk circuit path. The digit pattern here is "8-xxxx", where the "8" represents the fixed trunk access code which was created in AMO WABE - called "Field 1", and the "xxxx" represents any 4-digit station number which resides at the far end - called "Field 2". Therefore, the AMO LODR out-dial rule may consist of as few as two commands: "ECHO 2", and "END". Next, AMO LDAT is used to assign the trunk group number, the Out-Dial Rule number, LCR Authorization, and other parameters that "fine-tune" the route. AMO LDAT may also be used to create routing "choices" (if this trunk circuit is busy, then try a different path using a different trunk arrangement). Each choice is known as a "route element". Having multiple choices/route elements is a LUXURY - not a standard occurrence. Finally, AMO LDPLN is used to create a dialing pattern such as "8-xxxx", where all matching calls are sent to the route number which was defined and carefully-crafted using AMOs RICHT, LODR, and LDAT.

I have now provided the CONFIGURATION SEQUENCE, which is different from how the system processes a call. Here is how the AMOs flow when someone actually places a call:
User dials 8-2345 to reach station "2345" at remote system via SIP Trunks. The first digit "8" is analyzed by the Switching Unit, and the Digit Analysis Result (known as "DAR" in AMO WABE) is "TIE", which starts the Least Cost Routing process. The system then searches AMO LDPLN for a match on the dialed Pattern (8-2345). In LDPLN, the "8-xxxx" entry is the perfect match. This pattern points to a specific route number, which will lead directly to AMO LDAT. In AMO LDAT for that specific route number, the Trunk Group number represents the SIP Trunking circuit. The call is sent to the STMI via the shelf's TDM backplane. The call WAITS there for further instructions! Also in AMO LDAT, the Out-Dial rule number specifies that only "Field 2" digits be sent, therefore when ready, "2345" will be sent. If the call is allowed (authorizations and LCOSV settings can block users from placing certain calls), then the call is sent to the Trunk Group number specified in AMO LDAT, which is the Trunk Group on our OWN STMI board. How does the STMI know where to send the call? The process continues below!

The native SIP protocol can be "tweaked" to support certain features which may be available on the far-end PBX by selecting/configuring a Native SIP Profile, which is available by accessing the Web-Based Management on the STMI board. You can access the STMI via browser using several methods, including Assistant --> Expert Access --> HG35xx Management (no user/password required when using this connectivity option). Ultimately when connected to the STMI boards' Web-Based Management, you will click Explorers --> Voice Gateway --> SIP Trunk Profiles, then look for a matching "profile" describing the scenario at the trunk circuit's far end. Here, right-click, and select "EDIT". The typical process here is to enter the far end Gateway's IP Address, along with SIP Signaling port number "5060", then scroll down and click "Accept". As the final step, return to the SIP Trunk Profile in the Left Pane, then right-click, and select "Activate".
If the trunk circuit is properly configured, the routing is properly configured, the STMI's web-based settings are properly configured, the path between the two locations is UP, and the far end PBX is configured correctly, then a call from the 4K to the far end should be successful. As you can see, there are many potential points of failure! One tiny mistake can kill the entire process.

When the STMI board is UP, and all configuration is complete, it is advisable to check the status of the SIP Trunking circuit using AMO SDSU. Example: If the SIP Trunking circuit is installed on the STMI @ LTG 1, LTU 2, Slot 3, Circuit 0 (the first circuit), here is a sample SDSU status command:
DIS-SDSU:ALL,,PEN,PER3,1,2,3,0; (All 30 Channels should indicate "READY")

Returning to the explanation of the call process; when the call is sent to the STMI via the TDM backplane, the STMI examines its Web-Based Management settings, where YOU configured a destination IP Address for the far end Gateway. The STMI sets up a connection to this IP Address via its LAN connection to the customer's network. Ultimately the call is sent to the far end, and the call is complete.

Those are the BASICS! I assume that someone else on this forum can provide more details as needed. There is a Unify training course for this topic: IP Trunking - where the student can learn how to build Point-to-Point/Peer-to-Peer trunking (like this scenario), and also the much more complex Point-to-MultiPoint trunking - where the 4K's internal Large Enterprise GateKeeper must be configured to distribute calls to the proper location. This training course is typically 4 days, and covers a wide variety of IP Trunking scenarios that may also include SIP Load Balancing, a new feature that is required when the 4K is connected to a Unified Communications server with more than 1 Media Server.

I have semi-retired from contributing on this forum; therefore I-AM-NOT-supposed-to-be-HERE! But I was bored today.... so.....
GOOD LUCK!