[Asterisk-Users] Re: siemens pbx and asterisk

[Lito Lampitoc]

Do I still need an ATA adapter for my analog phones once I was able to
connect my Siemens HiPath 3750 to Asterisk?

Thanks in advance.



On 6/27/06, richard Coco <coco_richard at yahoo.com> wrote:
>
>
> hi all,
>
> The HG3550 V1 and HG3550v1.1 only supports H.323 V.2.
> I'am not sure but i thing that the feature "CallerID
> Name" was introduced in version 3 of the H.323
> standard. More informations about the owerviews at
> http://www.packetizer.com/voip/h323/.
>
> ->Concerning HiPathv3.0.
> In version 3.0 the HiPath has a new board (the HG3540)
> which supports SIP (for Endpoints) and SIPQ for
> SIP-trunking. You are now able to interconnect
> Asterisk and HiPath using H.323, ISDN and/or SIPQ.
>
> rich
>
> --- Herchi Silviu <Silviu.Herchi at arcelor.com> wrote:
>
> > Hi,
> >
> > As I wrote, the HiPath needs to be upgraded to
> > version 3 (don't ask me any details, I'm not a
> > Siemens expert) in order to have the CallerID name
> > passed over the H.323 link. Earlier versions (my
> > case) ony sends and accepts the CallerId number.
> >
> > I have set up a workaround for calls coming to
> > Asterisk: an AGI script sets the CallerID name
> > according to their CallerID number by looking it up
> > in a database. This is done in real time for every
> > incoming call. Obviously it doesn't work for calls
> > going from Asterisk to the HiPath.
> >
> > Regards,
> >
> > Silviu
> >
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On
> > Behalf Of Michael Hamann
> > Sent: 27 June 2006 14:58
> > To: Asterisk Users Mailing List - Non-Commercial
> > Discussion
> > Cc: asterisk-users at lists.digium.com
> > Subject: Re: [Asterisk-Users] Re: siemens pbx and
> > asterisk
> >
> > Hi Silviu,
> >
> > did you manage to get the callername to work? I have
> > a comparable setup with a hipath System but I
> can�t
> > get the callername to be displayed over the trunk.
> > The callernumber works but not the name...
> >
> > Any suggestion?
> >
> > Thanks
> > Michael
> >
> >
> > > We have successfully integrated an existing
> > Siemens HiPath 4500 PBX
> > > with two Asterisk servers.
> > >
> > > On the first one we use a H.323 trunk (it needs a
> > card on the PBX, I
> > > think it's called HG3550). It works pretty well,
> > except for one
> > > missing feature - the callerid name is not
> > transmitted over the link
> > > (it is a limitation of the PBX that should
> > disappear when it is
> > > upgraded to the
> > > V3 version). The nice thing is it doesn't take any
> > special hardware on
> > > the Asterisk server - you just have to compile and
> > setup an H.323
> > > channel (asterisk-oh323 works best for us).
> > >
> > > On the second one we have a Digium TE110P
> > connected to the PBX using a
> > > PRI. It works well too, you just need the PBX to
> > have a trunk defined
> > > and you're ready to go. We only use ten channels,
> > so I can't say if
> > > the performance is better. In this case you need
> > libpri and zaptel on
> > > the Asterisk.
> > >
> > > I hope this helps,
> > >
> > > Silviu
> > >
> > >
> > > ---
> > > Hello all,
> > >
> > > I'm new to asterisk. Our company wants to setup an
> > asterisk server and
> > > will eventually move to IP centric phones, but
> > they don't want to just
> > > throw away the old Siemens PBX, so during the
> > process we want to
> > > integrate it with asterisk. Is it possible? and
> > how?
> > > thanks.
> > > Lito
> >
> >
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