You might want to try the ozmod_pri instead of ozmod_isdn until the new revision of ozmod_isdn is published into the source tree.
On Thu, Oct 1, 2009 at 9:37 AM, <[email protected]> wrote: > 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 > > > ________________________________________________________ > Concordia University, Nebraska > See http://www.cune.edu/ for the latest news and events! > > > _______________________________________________ > FreeSWITCH-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:[email protected] <msn%[email protected]> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[email protected] <sip%[email protected]> iax:[email protected]/888 googletalk:[email protected]<googletalk%3aconf%[email protected]> pstn:213-799-1400
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