<div> Hi Nguyen.</div>
<div>
<div style="PADDING-RIGHT: 10px; PADDING-LEFT: 10px; PADDING-BOTTOM: 10px; PADDING-TOP: 10px">The Asterisk with configured 50 SIP Phones is integrated with HiPath 3750, through a board TMS2. For access the PSTN I created a context in asterisk so that asterisk has access HiPath 3750 and uses the LCR's that I configured in the HiPath. The CDR's do asterisk is registered no billing do HiPath that is the Siemens Call Report. The system of voicemail for all the set (HiPath 3750 and Asterisk) I use the Asterisk with access to the Internet, where all the 120 users receive its messages perfectly and also they receive copy from the message for email. The interconnection that we effect is ETSI or EuroISDN.
<br>I wait to have helped.<br>If to need plus some thing is to inform.</div>
<div style="PADDING-RIGHT: 10px; PADDING-LEFT: 10px; PADDING-BOTTOM: 10px; PADDING-TOP: 10px">Best Regards</div> Josué<br><br> </div>
<div><span class="gmail_quote">2006/6/13, Viktor Tatianin <<a href="mailto:vtatian@druzhba.lviv.ua">vtatian@druzhba.lviv.ua</a>>:</span>
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<div>
<div><font face="Arial" color="#0000ff" size="2"><span><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 size="+0"><span>Asterisk I use for voice mail, ivr and gateway for voice over ip </span></font></div>
<div><font size="+0"><span>I try connect Asterisk to PSTN with EDSS1 signaling it work fine</span></font></div>
<div><font size="+0"><span>at PSTN side statioon type 5ESS</span></font></div>
<div><font size="+0"><span></span></font> </div>
<div><font face="Arial" color="#0000ff" size="2"><span><font face="Times New Roman" color="#000000" size="3">What problem you have ?</font><br></span></font></div>
<blockquote>
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<div><span class="q">-----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></span>
</div>
<div><span class="e" id="q_10bcd447c97f2e41_2"><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></span></div>
<div></div></blockquote></div>
<div><span class="e" id="q_10bcd447c97f2e41_4"><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 onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:vtatian@druzhba.lviv.ua" target="_blank">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">
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<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>
<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>
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<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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