Siemens Hicom / HiPath

Siemens HiPath use the CorNet-IP proprietary protocol.
Someone is trying to write a Asterisk chan_cornet at the moment.
These are simple phones that act like a terminal (or thin client), the whole intelligence is in the server.
The nice thing with this system phones is - compared to standard protocols like H.323 or SIP - that they support a lot of features that the standard protocols do not.

Until the chan_cornet module is finished ...

You can use a 4 (quadBRI boards) or 8 (octoBRI cards) port card by e.g. Junghanns (a pri card should work as well) and put the Asterisk in between the NTBAs and the Siemens.
Configure as many ports as you have NTBAs to work in P2P (point to point (Anlagenanschluss)) or P2MP (point to multipoint (Mehrgeräteanschluss)) in TE mode. Configure the same number of ports on the card in the same way (P2P or P2MP) but NT mode and connect them to the Siemens.
You do not need to change anything on the Siemens - except the 0 for getting a line for convenience (let the Siemens automatically get a line for you - you have to configure the internal key on your system-phones for dialing them from another one directly (meaning not via Asterisk - because that would occupy 2 lines from and to the Siemens into Asterisk))

FAQ for HiCom 100E, 112E and 118E

The original site www.hicom-faq.de went off-line, hower the Internet archive still has it (in German). These PBX are sold on ebay in great quantities for cheap money and thus are a good start if you need a number of system phones to link to an Asterisk box.

Notes

If you want to send a * by capi you have to replace it by 75.

Be warned: I do have problems with ISDN cards connected to the Siemens using HDLC or X.75. They do not work! Fax and othe protocols seem to work well. If anyone as a solution for that, please let me know. (Hinnack)

HG1500

The HiPath HG 1500 is an integrated IP gateway for a HiPath 3000 system.
Asterisk can be easily interfaced with it using ooh323 or SIP.
First you will need to use Manager E to configure an IP trunk and the dial plan on the HiPath.
I have set up my HiPath so all '2XX' numbers are sent to the Asterisk trunk.
Once the trunk and dial plan are configured, you will need to configure the HG 1500.
Point your browser at http://IPOFHG:8085/ and login.

See: Explorers - Voice Gateway - PBX - IP Networking Data
If the HG 1500 is using the TCP alive monitoring method, be sure to disable it for the Asterisk node.
If the HG 1500 is using the ICMP alive monitoring method (ping), make sure the Asterisk server will respond to them.

See: Explorers - Voice Gateway - PBX - Nodes
Create a new node and enter the IP address of the Asterisk server.

See: Explorers - Voice Gateway - PBX - Routing
Add a new Call Address for the node you just created.

See: Explorers - Voice Gateway - Codec Parameters - Misc
To get DTMF to work you will need to check the following options...
Be sure that the 'Transmission of Fax/Modem/DTMF tones according to RFC2833' options are enabled.
Also make sure the 'ClearChannel' and 'Redundant Transmission according to RFC2198' options are disabled.

Note DTMF doesn't work properly.

• IAX phone - Asterisk - SIP - HiPath = WORKS
• Siemens phone - HiPath - SIP - Asterisk = WORKS
• IAX phone - Asterisk - SIP - HiPath - Siemens phone = WORKS
• Siemens phone - HiPath - SIP - Asterisk - IAX phone = BROKEN

For the SIP method:
    Example sip.conf
         hg1500sip
         type=peer
         host=192.168.0.51
         canrenvite=yes
         insecure=very
         qualify=no
         nat=no
         disallow=all
         allow=alaw
         allow=ulaw
         context=default

    Example extensions.conf
         macro-tohg1500
         exten => s,1,Answer()
         exten => s,2,Dial(SIP/${ARG1}@hg1500sip,30,tTr)
         exten => s,3,Goto(s-${DIALSTATUS},1)
         exten => s,4,Hangup()
         exten => t,1,Hangup()
         exten => T,1,Hangup()
         exten => h,1,Hangup()
         exten => s-BUSY,1,Playback(the-party-you-are-calling)
         exten => s-BUSY,2,Playback(is-curntly-busy)
         exten => s-BUSY,3,Goto(s,4)
         exten => s-CONGESTION,1,Playback(the-party-you-are-calling)
         exten => s-CONGESTION,2,Playback(is-curntly-unavail)
         exten => s-CONGESTION,3,Goto(s,4)
         exten => _s-.,1,Goto(s-BUSY,1)
         exten => _.,1,Goto(s,4)
         default
         exten => 209,1,Goto(mainmenu,s,1) ; Example incoming exten
         exten => _5.,1,Macro(tohg1500,${EXTEN}) ; Out to HG - mine uses a 5XX number range

For the H323 method:
    Example ooh323.conf
         hg1500
         type=friend
         host=192.168.0.51
         context=default

    Example extensions.conf
         macro-tohg1500
         exten => s,1,Answer()
         exten => s,2,Dial(OOH323/${ARG1}@192.168.0.51,30,tTr) ; Note the different Dial
         exten => s,3,Goto(s-${DIALSTATUS},1)
         exten => s,4,Hangup()
         exten => t,1,Hangup()
         exten => T,1,Hangup()
         exten => h,1,Hangup()
         exten => s-BUSY,1,Playback(the-party-you-are-calling)
         exten => s-BUSY,2,Playback(is-curntly-busy)
         exten => s-BUSY,3,Goto(s,4)
         exten => s-CONGESTION,1,Playback(the-party-you-are-calling)
         exten => s-CONGESTION,2,Playback(is-curntly-unavail)
         exten => s-CONGESTION,3,Goto(s,4)
         exten => _s-.,1,Goto(s-BUSY,1)
         exten => _.,1,Goto(s,4)
         default
         exten => 209,1,Goto(mainmenu,s,1) ; Example incoming exten
         exten => _5.,1,Macro(tohg1500,${EXTEN}) ; Out to HG - mine uses a 5XX number range

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Asterisk legacy integration