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<DIV><FONT face=Arial size=2>Hi Simon,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Point-to-Point (P2P) ist set if you want to use DID
(In Germany called "Anlagenanschluss") Point-to-Multipoit (ptmp) is if you want
to use the asterisk with singelpoint-entry (in Germany called
Mehrgeräteanschluss) (also possible to dial extensions after this, but look
at the maximum length for international calls!).</FONT></DIV>
<DIV><FONT face=Arial size=2>NT or TE is another question, one (NT) gives
the Synchronisation, the other (TE) gets it.</FONT></DIV>
<DIV><FONT face=Arial size=2>But you dont need an NTBA, cause Hicom and
HFC-Chipset can synchronise depending on configuration.</FONT></DIV>
<DIV><FONT face=Arial size=2>Crossover cable is needed anyway!</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>Regards Marco</FONT></DIV>
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style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=cyberenergy2k@hotmail.com
href="mailto:cyberenergy2k@hotmail.com">Simon D</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Sunday, September 18, 2005 10:57
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> RE: [Asterisk-Users] Integration
between Asterisk andSiemensHiCom150e over ISDN</DIV>
<DIV><BR></DIV>
<DIV>
<P class=RTE>Hi,</P>
<P class=RTE>Thanks for the advice on this.<BR>The Hicom can be set in
Point-to-Point or Point-to-Multipoint mode (amongst others), I assume one is
NT and the other is TE mode? If not, I cannot find any specific option to
set NT and TE.</P>
<P class=RTE>Anyway, I have a crossover cable, however, the article here:
<A
href="http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom">http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom</A> suggests
that I need an NT1 (NTBA) as well - is this correct? Do I also need
termination?</P>
<P class=RTE>I have set asterisk as bri_net_ptmp.</P>
<P class=RTE>Many thanks for your help people,</P>
<P class=RTE>Simon</P>
<P class=RTE> </P>
<DIV></DIV>
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<DIV></DIV>From: <I>"Sander"
<crombeen@rommelweb.nl></I><BR>Reply-To: <I>Asterisk Users
Mailing List - Non-Commercial
Discussion<asterisk-users@lists.digium.com></I><BR>To: <I>"'Asterisk
Users Mailing List - Non-Commercial
Discussion'"<asterisk-users@lists.digium.com></I><BR>Subject: <I>RE:
[Asterisk-Users] Integration between Asterisk and SiemensHiCom150e over
ISDN</I><BR>Date: <I>Tue, 13 Sep 2005 17:04:57
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(HELO pc1) (84.245.20.233)by gollum.cambrium.nl with SMTP; 13 Sep 2005
15:05:00 -0000</I><BR>>Just setup the stls4 card to work in NT mode and
connect the siemens pbx to<BR>>the asterisk with a crossover cable. Then
you will be able to make calls to<BR>>from the hicom to the asterisk
machine you do not need to have an nt box to<BR>>make the connection.
With the nt box you can power an ISDN phone if the<BR>>phone needs
power.<BR>><BR>><BR>><BR>>-----Oorspronkelijk
bericht-----<BR>>Van:
asterisk-users-bounces@lists.digium.com<BR>>[mailto:asterisk-users-bounces@lists.digium.com]
Namens Simon D<BR>>Verzonden: dinsdag 13 september 2005 15:14<BR>>Aan:
asterisk-users@lists.digium.com<BR>>Onderwerp: [Asterisk-Users]
Integration between Asterisk and Siemens<BR>>HiCom150e over
ISDN<BR>><BR>>Hi,<BR>><BR>>I am looking to integrate Asterisk
with a Siemens HiCom 150e via BRI and<BR>>wondered if anyone is able to
offer any advice.<BR>><BR>>In simplistic terms, my goal is to pass
calls from the HiCom to the Asterisk<BR>>box. e.g: HiCom user dials
access code and can call Asterisk extension or<BR>>establish SIP call
over Internet. Likewise, I'd like Asterisk to be able
to<BR>>present a call to the Hicom, either Asterisk extension calling
HiCom<BR>>extension, or an incoming Sipgate call presented to the HiCom,
for example.<BR>><BR>>My hardware:<BR>><BR>>-
Asterisk:<BR>><BR>>I have an Asterisk box configured with 1x Sitecom
DC105 PCI ISDN Card (HFC<BR>>chipset, TE/NT capable). [and 2x X100P
Analogue FXOs, but that's not<BR>>relevant here]<BR>><BR>>My
understanding is that I should configure the ISDN card in NT mode
and<BR>>power the bus with an NT1, or will a crossover cable in to the
HiCom<BR>>suffice? Reference to NT1's and line power
here:<BR>>http://isdn.jolly.de/download/v3.0/PBX4Linux-3.0.pdf<BR>><BR>><BR>>-
Siemens HiCom 150e:<BR>><BR>>I currently have a HiCom 150e switch with
digital and analogue stations and<BR>>an analogue trunk card
(TLA4).<BR>>I also have an STLS4 card. Initially, I thought
this would be the answer to<BR>>my prayers but now am not so
sure...<BR>>According to http://www.webco.com/siemens/interfaces.html, my
STLS4 is<BR>>defined as the following:<BR>><BR>>"Connects up to 4
ISDN S0 terminals (8 channels) for data equipment and<BR>>video
conferencing applications to the OfficePoint or OfficeCom.
Connected<BR>>devices must provide their own power. Some special wiring
for the ISDN S0<BR>>device connection is required. Also requires
connection of ISDN BRI trunks<BR>>(TMQ4) for network
access."<BR>><BR>>Documentation on ISDN
BRI:<BR>>http://web2.tac.siemenscom.com/pub/150e/Config/Note020.pdf<BR>>Documentation
on ISDN S0 Device
Install:<BR>>http://web2.tac.siemenscom.com/pub/150e/Config/Note009.pdf<BR>><BR>>I
can't figure out if the SLTS4 is the correct card for my requirements,
or<BR>>do I need a TMQ4 *as well* or a TMQ4
*instead*?..<BR>><BR>>Trawling through previous posts, I've found
reference to the STLS4
here:<BR>>http://lists.digium.com/pipermail/asterisk-users/2005-January/081449.html<BR>><BR>><BR>><BR>>Is
anyone able to help?<BR>><BR>><BR>><BR>>Many thanks in
advance,<BR>><BR>>Simon<BR>>England,
UK<BR>><BR>><BR>>_______________________________________________<BR>>--Bandwidth
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