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<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=514145511-13062006><FONT
color=#000000>I use</FONT> <FONT face="Times New Roman" color=#000000
size=3>PSTN -> Hicom 350-> Asterisk </FONT></SPAN></FONT></DIV>
<DIV><FONT><SPAN class=514145511-13062006>Asterisk I use for voice mail, ivr and
gateway for voice over ip </SPAN></FONT></DIV>
<DIV><FONT><SPAN class=514145511-13062006>I try connect Asterisk to PSTN with
EDSS1 signaling it work fine</SPAN></FONT></DIV>
<DIV><FONT><SPAN class=514145511-13062006>at PSTN side statioon type
5ESS</SPAN></FONT></DIV>
<DIV><FONT><SPAN class=514145511-13062006></SPAN></FONT> </DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=514145511-13062006><FONT
face="Times New Roman" color=#000000 size=3>What problem you have
?</FONT><BR></DIV></SPAN></FONT>
<BLOCKQUOTE>
<DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma
size=2>-----Original Message-----<BR><B>From:</B>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]<B>On Behalf Of
</B>Nguyen<BR><B>Sent:</B> Tuesday, June 13, 2006 1:37 PM<BR><B>To:</B>
Asterisk Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> Re:
[Asterisk-Users]Asterisk <-IP-> Siemens HiPath
3750<BR><BR></FONT></DIV><BR>Hi Viktor,<BR>So where is the PSTN side on your
schema? PSTN -> Asterisk -> Hicom 350? Or PSTN -> Hicom 350->
Asterisk?<BR>Thanks<BR>Nguyen<BR>
<DIV><SPAN class=gmail_quote>On 6/13/06, <B class=gmail_sendername>Viktor
Tatianin</B> <<A
href="mailto:vtatian@druzhba.lviv.ua">vtatian@druzhba.lviv.ua</A>>
wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">
<DIV>
<DIV>
<DIV><SPAN><FONT face=Arial color=#0000ff size=2>Hi </FONT></SPAN></DIV>
<DIV><SPAN><FONT face=Arial color=#0000ff size=2>I have next working
sheme</FONT></SPAN></DIV>
<DIV><SPAN><FONT face=Arial color=#0000ff size=2>Hicom 350 - (Diun2
card)with DSS1- Asterisk with Quiad E1 </FONT></SPAN></DIV>
<DIV><SPAN><FONT face=Arial color=#0000ff size=2>This is work
fine</FONT></SPAN></DIV></DIV>
<DIV><SPAN class=e id=q_10bcc8c3b027c04e_1>
<BLOCKQUOTE>
<DIV dir=ltr align=left><FONT face=Tahoma size=2>-----Original
Message-----<BR><B>From:</B> <A
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:asterisk-users-bounces@lists.digium.com"
target=_blank>asterisk-users-bounces@lists.digium.com</A> [mailto:<A
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:asterisk-users-bounces@lists.digium.com"
target=_blank>asterisk-users-bounces@lists.digium.com</A>]<B>On Behalf Of
</B>Nguyen<BR><B>Sent:</B> Tuesday, June 13, 2006 10:32 AM<BR><B>To:</B>
Asterisk Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B>
Re: [Asterisk-Users]Asterisk <-IP-> Siemens HiPath
3750<BR><BR></FONT></DIV>Hi Josué,<BR><BR>I just got the
confirmation about integrating TE110P with TMS2 of Hipath 3750. Your
help will be much appreciated.<BR><BR>The configuration is as
follow:<BR><BR>PSTN -> HIPATH 3750 (14 analog trunk lines) -> TMS2
-> TE110P -> Asterisk <BR><BR>All extensions of Hipath 3750 are
analog (120 extensions)<BR><BR>I know that it's maybe easier if we do
other way, PSTN->ASterisk (2E-1) -> TMS2 -> Hipath 3750. But this
is not an option, due to some political debat :((( <BR><BR>I don't have
the tech manual of Hipath yet, but here is what I want to do:<BR><BR>1/
Calls from PSTN side arrived on Hipath 3750. Hipath somehow transfer that
call into Asterisk box, using TMS2. Asterisk, functioning as an voicemail,
feature server (voice log, conference, etc), after some menu
prompts, will transfer back the call to Hipath 3750, using the
same TMS2-TE110P connection, to one analog extension of Hipath 3750.
<BR><BR>2/User of exteniosns of Hipath 3750, when dial out, will be
transfered into Asterisk, using the same TMS2->TE110P.
Asterisk will do the check of user balance account, LCR, and
if approved , will transfer the call back to Hipath 3750, for
getting into Analog trunk line. <BR><BR>Since for the Hipath, TMS2 is a
trunk module, so I suspect that some DISA operation must be enabled on
Hipath, so we can enable the path from analog trunk port -> TMS2 ->
Asterisk and back?<BR><BR>Is above configuration working? <BR><BR>And TMS2
use CAS, so do we have to use MFC/R2 (chan_unicall?)<BR><BR>Very
interested in your working configuration, can you explain a
bit?<BR><BR>Thank you and best regards,<BR>Nguyen<BR><BR><BR>
<DIV><SPAN class=gmail_quote>On 5/26/06, <B class=gmail_sendername>Josué
Conti</B> <<A onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:josueconti@gmail.com"
target=_blank>josueconti@gmail.com</A>> wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">
<DIV>
<DIV
style="PADDING-RIGHT: 10px; PADDING-LEFT: 10px; PADDING-BOTTOM: 10px; PADDING-TOP: 10px">Hi
I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help
you, I do not have manuals technician to send, but if to want can help.
Already I established connection asterisk( 1.0.9) with Hipath 3750 with
a TE110P and a TMS2, functioned 100%. The equipment says between sim.The
asterisk uses HiPath 3750, for access the PSTN and when a linking is for
a telephone of asterisk, the Hipath directs the digits for asterisk.
</DIV>
<DIV
style="PADDING-RIGHT: 10px; PADDING-LEFT: 10px; PADDING-BOTTOM: 10px; PADDING-TOP: 10px">I
wait to have helped.</DIV>
<DIV
style="PADDING-RIGHT: 10px; PADDING-LEFT: 10px; PADDING-BOTTOM: 10px; PADDING-TOP: 10px">Greetings</DIV>
<DIV>Josué</DIV>
<DIV> </DIV>
<DIV><BR> </DIV>
<DIV><SPAN class=gmail_quote>2006/5/25, Benchev <<A
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:bbench@mail.bg"
target=_blank>bbench@mail.bg</A>>:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid"></BLOCKQUOTE></DIV>
<DIV><SPAN>Hi Nguyen ,<BR>I haven't got the opportunity to make my
project real due to business <BR>obstacles, but I still think that it
should work. <BR>All that follows is a theory, but there are guys on the
list that<BR>might help you with more practical advises.<BR>> I have
stuck with Hipath 3750 and Asterisk + TE110P. I don't have the<BR>>
manual of Hipath 3500 yet (have to buy from local vendor), so I was not
<BR>> sure are these thing possible<BR>><BR>> Scenario:
Asterisk|TE110P->TMS2|Hipath 3750 ->(16 CO lines) PSTN<BR>I had
the same idea because I wanted to save on the card side(single
span),<BR>and use the Hipath as a "channel bank"
:-) <BR><BR>> - Is this possible for Asterisk Users call out using CO
lines? Some of<BR>> Siemens guys told me that I need an DISA card for
this? Is this true?<BR>Most of the time the Siemens guys don't know what
is Asterisk. <BR>Basically TE110P *is* a DISA since it gives Direct
Inward System Access<BR>(if this is what they mean by DISA)<BR><BR>Below
is a threat I found with exactly the same scenario like yours:<BR><A
onclick="return top.js.OpenExtLink(window,event,this)"
href="http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.html"
target=_blank>http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.html
</A><BR>And this proves that the idea must work.<BR>> - When the call
arrived from PSTN through CO line, can it be forwarded to<BR>>
Asterisk? Again, they says that we require the DISA card. <BR><BR>As far
as anything gets into Asterisk then you are free to do whatever
you<BR>want. I don't know what DISA they are talking about? Do they mean
S2M<BR>or similar thing(but TMS2 is S2M)?<BR>Anyone?<BR><BR>Sorry for
not being able to help, but hope somebody else <BR>would do
it.<BR><BR>Benchev<BR><BR><BR></SPAN></DIV></DIV></BLOCKQUOTE></DIV><BR></BLOCKQUOTE></SPAN></DIV>
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