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

[Viktor Tatianin]

I use PSTN -> Hicom 350-> Asterisk
Asterisk I use for voice mail, ivr and gateway for voice over ip
I try connect Asterisk to PSTN with EDSS1 signaling it work fine
at PSTN side statioon type 5ESS

What problem you have ?

  -----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 1:37 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users]Asterisk <-IP-> Siemens HiPath 3750



  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.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.
        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/d26c35a6/attachment.htm