[Freeswitch-users] Connecting FS to Hicom 300

[Russell.Mosemann at cune.org]

We have connected FS to a Siemens Hicomm 300. As you might guess, it's
not working right. Here is what we are working with.

Dell 1750 (dual socket, dual core Xeon 2.8GHz)
Debian 5
FS (15029), OpenZAP (without libpri)
TE110P T1 card (Zaptel driver)
Handles 71xx extensions

Siemens Hicom 300
TMDN64P T1 card
Handles 74xx extensions

We are pretty much using the stock FS configuration, yet, because we're
trying to get this to work. I have configured OpenZAP and the associated
files like the examples on the wiki (see below) to work with a PRI T1.
There are 23 B channels and 1 D channel. The Zaptel side looks fine.
OpenZAP is able to open the channels when FS boots. So far, so good.

When a call is made from 74xx (Rolmphone 624) to 71xx (X-Lite 4.0 beta
from CounterPath on an office PC), X-Lite rings. The call can be
answered, and the conversation sounds fine. That means the routing,
registration and authorization are working on the network between X-Lite
and FS. It also means that FS is able to communicate with the Hicom over
the T1. Great.

When the caller presses the transfer button on the 74xx phone, the Hicom
sends a message over the D channel, and the call is disconnected
(watching with fs_cli). As best I can interpret the bytes in the message,
the Hicom sends a disconnect message when 74xx presses the transfer key.

In order to call 74xx, I created dialplan/default/02_hicom.xml. The
contents are

<include>
  <extension name="hicom">
    <condition field="destination_number" expression="^(74\d{2})$">
      <action application="bridge" data="openzap/1/a/$1"/>
    </condition>
  </extension>
</include>

If a call is made from 71xx to 74xx, the Hicom forwards the call to the
switchboard with "7100->7445 connection not possible" (or whatever
extensions) in the switchboard display.

1. Are these issues related to the way I have configured FS?

The Hicom is maintained by the local phone company. I do not have access
to view or configure the T1 card on the Hicom. According to the phone
guy, there isn't anything else that needs to be configured on the Hicom.
He believes that if 74xx can call 71xx, then 71xx should be able to call
74xx.

I suspect that something more needs to be done on the Hicom in order to
accept calls from FS and bridge/transfer them to a local extension on the
Hicom. It's as if the Hicom doesn't know how or is not permitted to route
incoming calls on the T1 to local extensions. I have no way to know,
though. I'm hoping someone else has connected FS to a Hicom 300 and can
provide configuration details. If I could tell the phone guy something
like, "You need to look at <this>," that would help him out.

2. Should I receive CID/ANI from the Hicom?

X-Lite displays "OpenZAP" as the call and "1" as Other when the call
comes in, which is the information for the endpoint. Is there something I
need to do in the FS configuration to capture CID/ANI information from
the Hicom and make it available (or is it not being provided by the Hicom)?

3. When dialing from the Rolmphone is there a way for FS to send the
called name back to the Hicom for it to appear in the display?

When dialing 74xx to 74xx, of course, it shows the called number and name
in the display. We also have a HiPath 4000 connected to the Hicom 300.
When dialing an extension on the HiPath from the Hicom, the HiPath ships
the called name back to the Hicom for display on the phone. It would be
nice to do that from FS.

Let me know if you need additional information. Thanks for any pointers
or insight as to how things work.

-- 
Russell Mosemann


openzap.conf
[span zt PRI_1]
name => OpenZAP
number => 1
trunk_type => t1
b-channel => 1-23
d-channel => 24

zt.conf
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
echo_cancel_level => 64
rxgain => 0.0
txgain => 0.0

openzap.conf
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="0"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
   <pri_spans>
     <span name="PRI_1">
       <param name="q921loglevel" value="alert"/>
       <param name="q931loglevel" value="alert"/>
       <param name="mode" value="user"/>
       <param name="dialect" value="national"/>
       <param name="dialplan" value="XML"/>
       <param name="context" value="public"/>
     </span>
   </pri_spans>
</configuration>

zaptel.conf
# Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER)
span=1,1,0,esf,b8zs
# termtype: te
bchan=1-23
dchan=24

# Global data
loadzone        = us
defaultzone     = us


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