[Asterisk-Users]Asterisk <-IP-> Siemens HiPath 3750

[Nguyen]

Hi Viktor,
So where is the PSTN side on your schema? PSTN -> Asterisk -> Hicom 350? Or
PSTN -> Hicom 350-> Asterisk?
Thanks
Nguyen
On 6/13/06, Viktor Tatianin <vtatian at druzhba.lviv.ua> wrote:
>
>  Hi
> I have next working sheme
> Hicom 350 - (Diun2 card)with DSS1- Asterisk with Quiad E1
> This is work fine
>
> -----Original Message-----
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com]*On Behalf Of *Nguyen
> *Sent:* Tuesday, June 13, 2006 10:32 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [Asterisk-Users]Asterisk <-IP-> Siemens HiPath 3750
>
> Hi Josué,
>
> I just got the confirmation  about integrating TE110P with TMS2 of Hipath
> 3750. Your help will be much appreciated.
>
> The configuration is as follow:
>
> PSTN -> HIPATH 3750 (14 analog trunk lines) -> TMS2 -> TE110P -> Asterisk
>
> All extensions of Hipath 3750 are analog (120 extensions)
>
> 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
> :(((
>
> I don't have the tech manual of Hipath yet, but here is what I want to do:
>
> 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.
>
> 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.
>
> 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?
>
> Is above configuration working?
>
> And TMS2 use CAS, so do we have to use MFC/R2 (chan_unicall?)
>
> Very interested in your working configuration, can you explain a bit?
>
> Thank you and best regards,
> Nguyen
>
>
> On 5/26/06, Josué Conti <josueconti at gmail.com> wrote:
> >
> >  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.Theasterisk uses HiPath 3750, for access the PSTN and when a linking is for a
> > telephone of asterisk, the Hipath directs the digits for asterisk.
> > I wait to have helped.
> > Greetings
> > Josué
> >
> >
> >
> > 2006/5/25, Benchev <bbench at mail.bg>:
> > >
> > > Hi Nguyen ,
> > I haven't got the opportunity to make my project real due to business
> > obstacles, but I still think that it should work.
> > All that follows is a theory, but there are guys on the list that
> > might help you with more practical advises.
> > > I have stuck with Hipath 3750 and Asterisk + TE110P. I don't have the
> > > manual of Hipath 3500 yet (have to buy from local vendor), so I was
> > not
> > > sure are these thing possible
> > >
> > > Scenario: Asterisk|TE110P->TMS2|Hipath 3750 ->(16 CO lines) PSTN
> > I had the same idea because I wanted to save on the card side(single
> > span),
> > and use  the Hipath as a "channel  bank" :-)
> >
> > > - Is this possible for Asterisk Users call out using CO lines? Some of
> > > Siemens guys told me that I need an DISA card for this? Is this true?
> > Most of the time the Siemens guys don't know what is Asterisk.
> > Basically TE110P *is* a DISA since it gives Direct Inward System Access
> > (if this is what they mean by DISA)
> >
> > Below is a threat I found with exactly the same scenario like yours:
> > http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.html
> > And this proves that the idea must work.
> > > - When the call arrived from PSTN through CO line, can it be forwarded
> > to
> > > Asterisk? Again, they says that we require the DISA card.
> >
> > As far as anything gets into Asterisk then you are free to do whatever
> > you
> > want. I don't know what DISA they are talking about? Do they mean S2M
> > or similar thing(but TMS2 is S2M)?
> > Anyone?
> >
> > Sorry for not being able to help, but hope somebody else
> > would do it.
> >
> > Benchev
> >
> >
> >
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060613/85135e5f/attachment.htm