<div style="PADDING-RIGHT: 10px; PADDING-LEFT: 10px; PADDING-BOTTOM: 10px; PADDING-TOP: 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">Silviu.Herchi@arcelor.com</a>>:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
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<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 class="e" id="q_10c14da980fe1819_1"><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>
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<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> </p></div></div><br>_______________________________________________<br>--Bandwidth and Colocation provided by
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