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: [email protected] > 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 > > > _______________________________________________ > --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 > _______________________________________________ > --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 > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ --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
