"AsteriskNow + FreePBX Connection With Siemens Hicom 150

Hello to everybody here,

I am new to this forum and AsteriskNow as well so i need your help for the follows:

my company we have as primary PBX system the “Siemens Hicom 150” but
due to additional needs I have setup one “AsteriskNow + FreePBX” server
in order to communicate with our local offices ( worldwide).
I would
like to ask you how I can connect the existing PBX system “Siemens Hicom
150” with “AsteriskNow + FreePBX” server so that our staff in
headquarter can be call the our worldwide offices from existing analog

We need from this interconnection, at least 20 phones, at same time be working and making calls.

In your reply, kindly requested to inform me what peripheral units I need ( such: extra card or fxo gateway etc)

Thank you in advance for your kind attention to this subject matter


Hi, your request, considering the age and type of Siemens PBX, is a challenge!

I presume your system could be a Siemens Hicom 150E R2.2 or, eventually, a 150H V1 - formerly known as 150E R3.0 (either the case…your Hardware/Software system was available years before Siemens introduced, really many years ago, the “new” HiPath 3000 V1.2 - then upgraded to V3, V4, V5, V6, V7, V8 and V9 - family). The HiPath 3000 family, reached V9, was recently Phased Out in favour of the new and shiny over-licensed linux based half open half closed source Unify OpenScape Business V1 (a masked HiPath 3000 V10 with new Software and a truly re-engineered Mainboard).

In the past Siemens introuced a Media-Gateway sub-module called HXGS (Small)/HXGM (Medium) formerly known as Siemens Hicom Xpress@LAN respectively available for Hicom 150E OfficePoint/OfficeCom or OfficePro (basically an H.323/CorNet-IP <-> system media gateway), that card then was renamed into Siemens HiPath HG1500 (V1, V2 and V3) with an Hardware completely re-engineered (same Form Factor for HiPath 33x0/35x0 or HiPath 37x0) and a new Firmware and Software (with new Features and Services, like SIP support for Trunking and Devices when coupled with HiPath 3000 V6 and newer) both enhanced to cope with new HiPath 3000 features.

All this long preamble to say that walking the track of finding and using an integrated media gateway is a no go (due to your system age and its intrinsic Hardware/Software/Interoperability limits): you should upgrade your system to (at least) HiPath 3000 V9 (so changing Hardware and Software, worrying also about licensing if you have ISDN PRI Lines) just to have it equipped with an HG1500 V3 (Support of SIP was introduced along the way only with latest HG1500 V3 Software releases)…and all this just to say “WOW, now I’ve finally SIP support on my integrated Siemens Media Gateway!”. IMHO it’s a no go…in term of labour, costs and troubles you’re going to experience…more…thinking that HiPath 3000 was declared Phased Out adds some real blue to all this.

Finding, configuring and using an old Siemens Xpress@LAN is not an option: no SIP Support, just H.323/CorNet-IP (which is proprietary CorNet-N incapsulated into IP, called HFA when used with clients and CorNet-IP when used for IP trunking) …and I don’t know if you would go down that sandy road!

The other way could be using an external SIP to ISDN (TE mode) Media Gateway so you can connect your Hicom 150E R2.2 unused Hicom system ISDN trunks channels (2, 4 or more Channels per Trunk…it depends also on Media Gateway supported ports) and have it registered to your Asterisk+FreePBX system. Results may vary due to Media Gateway configuration/interoperability.

Just my 2 cents (or just 1 Euro cents!).

An advice: Clearly you should provide the topology of your systems/nodes in order to receive good feedback…but I’ve no fear to say that you should throw away that old PBX (indeed it is really rock solid on its basic digital features) and migrate all your nodes to IP each one with/without an integrated ISDN Media Gateway (and, keep in mind, if you have lot of DECT Terminals implemented through Hicom Cordless Office board consider that Gigaset Pro has a DECT/IP solution that integrates flawlessly with Asterisk+FreePBX)…at this point all I wrote is a function of costs your company have to sustain to throw away a (small/medim/large) set of old (end EOL) optiSet E (or old and Phased Out optiPoint 500) digital propietary phones in favour of new SIP IP phones.

Thank you parnassus for your reply.

I am fully agree with your suggestion, to replace the old PBX Hicom 150 with newest.

But, the management push me so hard, in order to found a solution for the next 2-3 years
until company moves to our own new building. Believe me is a big headache for me this
upgrade something I like to avoid.

The company who have install existing PBX has been already stop the services, and any other
company that I have search in the local market does not like to run the project, from all I receive
the same answer: “it is too old, sorry we can’t support you”.

Anyway I will try with your suggestion/instructions to see if I can found a solution.

Thank you so much for time you have spent to my problem.


Hi idown, you’re welcome.

I gave the above suggestions/instructions not because the Hicom 150E is too old or unsupported (it’s natural…after so many years…it’s not easy to manage a such old piece of Hardware) but because I think the only way to avoid a bad series of strong headaches is to keep the whole design as simple as you can (the KISS method) and let the Hicom 150E configuration to stay as it is now (I mean: as more as possible, at least) while introducing an external Media Gateway in the middle (there are many, like Patton, AudioCodes, etc) that will interface the Asterisk (IP) world with the Siemens digital one…acting in this way, IMHO, you should only be worried to configure relevant LCR Routes on the Hicom 150E to let it routes some calls (once a matching rule is satisfied) to the ISDN S0 Trunks connected to that ISDN/SIP Media Gateway.

Sure that problems could arise to grant a symmetrical behaviour (calls completion and payload of originating calls from Hicom to Asterisk and vice versa), but nothing so bad as to look for (and then implement) an old Hicom Xpress@LAN or to move your whole system (PBX plus digital Terminals) to HiPath 3000 just to try SIP Trunking directly through HG1500.

Good luck!

Based on what you have posted so far, and assuming that you want 20 concurrent calls between the systems, this could be a simple matter of adding an E1 (a.k.a. ISDN30, primary rate access or S2M) card to both systems and running a 2-pair cable between them.

If there are constraints that make this impossible (Hicom chassis is full, card not available, systems are far apart, etc.), please provide details about the setup.

Conceivably, you may not need to interconnect the systems at all. For example, if you have plenty of local trunk capacity and unmetered local calling, you could set up the Asterisk system with local DIDs that would ring to the remote offices.

What spare extension / trunk ports do you have on the Hicom? Spare card slots? What country is the Hicom in?

It all depends on his network topology, his Hicom system placement (inside this topology) and the way it is actually equipped.

I thought that he probably has a central node (Headquarter) historically Hicom based and then various Branch Offices each one with a IP-PBX (connected via IAX2 to Headquarter?) or just - more probably - he has various IP Phones installed on each Branch offices worldwide (connected straight through VPN Tunnels to the Headquarter’s Asterisk IP-PBX node): if so the requirement is “Asterisk reaches all the Branch Offices, but the Headquarter’s Hicom terminals are cut off from the picture, how can I interconnect them to the rest of our IP based infrastructure used in our Headquarter/Branch Offices?”…and 20 terminals maybe don’t mean 20 (really) concurrent calls at all (if so you should have then a plenty of very good - QoS/SLA - bandwidth too) and think about a PRI/SIP Media Gateway in order to support that voice traffic.