Hello Silviu,<br><br>Thank you very much for your reply. I will try that.<br><br><div><span class="gmail_quote">On 6/27/06, <b class="gmail_sendername">Herchi Silviu</b> <<a href="mailto:Silviu.Herchi@arcelor.com">Silviu.Herchi@arcelor.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;"><div>
<div>
<p><span lang="fr-ch"><font face="Arial" size="2">Hi Lito,</font></span>
</p>
<p><span lang="fr-ch"><font face="Arial" size="2">We have successfully integrated an existing Siemens HiPath 4500 PBX with two Asterisk servers.</font></span>
</p>
<p><span lang="fr-ch"><font face="Arial" size="2">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).</font></span></p>
<p><span lang="fr-ch"><font face="Arial" size="2">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.
</font></span></p>
<p><span lang="fr-ch"><font face="Arial" size="2">I hope this helps,</font></span>
</p>
<p><span lang="fr-ch"><font face="Arial" size="2">Silviu</font></span>
</p>
<br>
<p><span lang="fr-ch"><font face="Arial" size="2">---</font></span>
</p></div><div><span class="e" id="q_10c14d350cc7bcb9_1"><br><span lang="en-gb"><font face="Arial">Hello all,<br>
<br>
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?
<br>
<br>
thanks.<br>
<br>
Lito </font></span>
</span></div><div><p></p>
</div>
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