HERE IS answer !@!<br><br><br>had same problem..<br><br>make the settings for 90 volt.. not 70 volt ringer.. <br><br>make it trapezoidal not sinusoisal<br><br>make it 900 ohm not 600 impedence..<br><br>that worked for pap2's
<br><br>seem siemens are made for europe style ring voltage not north american.<br><br><br><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>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2">Hello,</font></span></div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2"></font></span> </div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2">The main differences I can see:</font></span></div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2"></font></span> </div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2">- in zaptel.conf</font></span></div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2">you have span=1,0,0,ccs,hdb3, which means you ask Asterisk
to serve as a timer for the PBX - on my setup the PBX is the master clock and
Asterisk is the secondary one, so I have span=1,1,0,ccs,hdb3 (in fact, as I use
CRC4 error correction, my setup is span=1,1,0,ccs,hdb3,crc4)</font></span></div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2"></font></span> </div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2">- in zapata.conf</font></span></div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2">I have switchtype=EuroISDN. Generally speaking, try using
other switchtypes.</font></span></div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2"></font></span> </div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2">Regards,</font></span></div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2"></font></span> </div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2">Silviu</font></span></div>
<div dir="ltr" align="left">
<hr>
</div>
<div dir="ltr" align="left"><font face="Tahoma" size="2"><b>From:</b>
<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Josué
Conti<br><b>Sent:</b> 27 June 2006 14:41<br><b>To:</b> Asterisk Users Mailing
List - Non-Commercial Discussion<br><b>Subject:</b> Re: [Asterisk-Users] Re:
Asterisk x Siemens HiPath 4000<br></font><br></div></div><div><span class="e" id="q_10c15fdc5440da15_1">
<div></div>
<div style="padding: 10px;">Silviu,
thank's will be this attention. Below my configurations of zapata.conf and
zaptel.conf</div>
<div>#zapte.conf</div>
<div>span=1,0,0,ccs,hdb3<br>bchan=1-15<br>dchan=16<br>bchan=17-31<br>loadzone=us<br>defaultzone=us<br> </div>
<div>#zapata.conf</div>
<p>[trunkgroups]</p>
<p>[channels]<br>language=pt_BR<br>context=default<br>switchtype=qsig<br>pridialplan=private<br>prilocaldialplan=private<br>facilityenable
=
yes<br>signalling=pri_cpe<br>;rxwink=300<br>usecallerid=yes<br>hidecallerid=no<br>callwaiting=yes<br>usecallingpres=yes<br>restrictcid=no<br>callwaitingcallerid=yes<br>threewaycalling=yes<br>transfer=yes<br>canpark=yes<br>
cancallforward=yes<br>callreturn=yes<br>echocancel=yes<br>echocancelwhenbridged=yes
<br>rxgain=0.0<br>txgain=0.0<br>group=1<br>callgroup=1<br>immediate=no<br>callerid=asreceived<br>musiconhold=default<br>group=1<br>channel=>1-15<br>channel=>17-31<br></p>
<p> </p>
<div>Best Regards</div>
<div> </div>
<div>Josué</div>
<div><br><br> </div>
<div><span class="gmail_quote">2006/6/27, Herchi Silviu <<a href="mailto:Silviu.Herchi@arcelor.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">Silviu.Herchi@arcelor.com</a>>:</span>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">
<div>
<div>
<p><font face="Arial" size="2">Hi,</font> </p>
<p><font face="Arial" size="2">Could you post your /etc/zaptel.conf and
zapata.conf?</font> </p>
<p><font face="Arial" size="2">Also, is everything OK the other way round (i.e.,
from the SIP phones to the PBX)?</font> </p>
<p><font face="Arial" size="2">Silviu</font> </p>
<p><font face="Arial" size="2">----</font> </p></div>
<div><span><br><font face="Arial" size="2">Hello
all.</font> <br><font face="Arial" size="2">I have installed and functioning
asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9 , is
interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG,
the one that is happening he is that the calls originated for PABX Siemens and
destined to SIP phones asterisk are being without audio, nor Ring, is dumb.
They could help in this case me? </font></span></div>
<div>
<p></p>
<p><font face="Arial" size="2">Best Regards</font> <br><font face="Arial" size="2"> </font> <br><font face="Arial" size="2">Josué</font>
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