Thanks in advance.
On 6/27/06,
richard Coco <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] ] On
> Behalf Of Michael Hamann
> Sent: 27 June 2006 14:58
> To: Asterisk Users Mailing List - Non-Commercial
> Discussion
> Cc: asterisk-users@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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