Do I still need an ATA adapter for my analog phones once I was able to connect my Siemens HiPath 3750 to Asterisk?<br><br>Thanks in advance.<br><br><br><br><div><span class="gmail_quote">On 6/27/06, <b class="gmail_sendername">
richard Coco</b> <<a href="mailto:coco_richard@yahoo.com">coco_richard@yahoo.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>hi all,<br><br>The HG3550 V1 and HG3550v1.1 only supports H.323 V.2.<br>I'am not sure but i thing that the feature "CallerID<br>Name" was introduced in version 3 of the H.323<br>standard. More informations about the owerviews at
<br><a href="http://www.packetizer.com/voip/h323/">http://www.packetizer.com/voip/h323/</a>.<br><br>->Concerning HiPathv3.0.<br>In version 3.0 the HiPath has a new board (the HG3540)<br>which supports SIP (for Endpoints) and SIPQ for
<br>SIP-trunking. You are now able to interconnect<br>Asterisk and HiPath using H.323, ISDN and/or SIPQ.<br><br>rich<br><br>--- Herchi Silviu <<a href="mailto:Silviu.Herchi@arcelor.com">Silviu.Herchi@arcelor.com</a>> wrote:
<br><br>> Hi,<br>><br>> As I wrote, the HiPath needs to be upgraded to<br>> version 3 (don't ask me any details, I'm not a<br>> Siemens expert) in order to have the CallerID name<br>> passed over the H.323
link. Earlier versions (my<br>> case) ony sends and accepts the CallerId number.<br>><br>> I have set up a workaround for calls coming to<br>> Asterisk: an AGI script sets the CallerID name<br>> according to their CallerID number by looking it up
<br>> in a database. This is done in real time for every<br>> incoming call. Obviously it doesn't work for calls<br>> going from Asterisk to the HiPath.<br>><br>> Regards,<br>><br>> Silviu<br>><br>
> -----Original Message-----<br>> From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com
</a>] On<br>> Behalf Of Michael Hamann<br>> Sent: 27 June 2006 14:58<br>> To: Asterisk Users Mailing List - Non-Commercial<br>> Discussion<br>> Cc: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com
</a><br>> Subject: Re: [Asterisk-Users] Re: siemens pbx and<br>> asterisk<br>><br>> Hi Silviu,<br>><br>> did you manage to get the callername to work? I have<br>> a comparable setup with a hipath System but I
<br>can�t<br>> get the callername to be displayed over the trunk.<br>> The callernumber works but not the name...<br>><br>> Any suggestion?<br>><br>> Thanks<br>> Michael<br>><br>><br>> > We have successfully integrated an existing
<br>> Siemens HiPath 4500 PBX<br>> > with two Asterisk servers.<br>> ><br>> > On the first one we use a H.323 trunk (it needs a<br>> card on the PBX, I<br>> > think it's called HG3550). It works pretty well,
<br>> except for one<br>> > missing feature - the callerid name is not<br>> transmitted over the link<br>> > (it is a limitation of the PBX that should<br>> disappear when it is<br>> > upgraded to the
<br>> > V3 version). The nice thing is it doesn't take any<br>> special hardware on<br>> > the Asterisk server - you just have to compile and<br>> setup an H.323<br>> > channel (asterisk-oh323 works best for us).
<br>> ><br>> > On the second one we have a Digium TE110P<br>> connected to the PBX using a<br>> > PRI. It works well too, you just need the PBX to<br>> have a trunk defined<br>> > and you're ready to go. We only use ten channels,
<br>> so I can't say if<br>> > the performance is better. In this case you need<br>> libpri and zaptel on<br>> > the Asterisk.<br>> ><br>> > I hope this helps,<br>> ><br>> > Silviu
<br>> ><br>> ><br>> > ---<br>> > Hello all,<br>> ><br>> > I'm new to asterisk. Our company wants to setup an<br>> asterisk server and<br>> > will eventually move to IP centric phones, but
<br>> they don't want to just<br>> > throw away the old Siemens PBX, so during the<br>> process we want to<br>> > integrate it with asterisk. Is it possible? and<br>> how?<br>> > thanks.<br>> > Lito
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