[Asterisk-Users]Asterisk <-IP-> Siemens HiPath 3750
Nguyen
chinhnam.nguyen at gmail.com
Tue Jun 13 00:32:16 MST 2006
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
>
>
>
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