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<DIV dir=ltr align=left><SPAN class=703533014-27062006><FONT face=Arial
color=#0000ff size=2>Hello,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=703533014-27062006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=703533014-27062006><FONT face=Arial
color=#0000ff size=2>The main differences I can see:</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=703533014-27062006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=703533014-27062006><FONT face=Arial
color=#0000ff size=2>- in zaptel.conf</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=703533014-27062006><FONT face=Arial
color=#0000ff 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 class=703533014-27062006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=703533014-27062006><FONT face=Arial
color=#0000ff size=2>- in zapata.conf</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=703533014-27062006><FONT face=Arial
color=#0000ff size=2>I have switchtype=EuroISDN. Generally speaking, try using
other switchtypes.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=703533014-27062006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=703533014-27062006><FONT face=Arial
color=#0000ff size=2>Regards,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=703533014-27062006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=703533014-27062006><FONT face=Arial
color=#0000ff size=2>Silviu</FONT></SPAN></DIV>
<DIV dir=ltr align=left>
<HR tabIndex=-1>
</DIV>
<DIV dir=ltr align=left><FONT face=Tahoma size=2><B>From:</B>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <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>
<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">
<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 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>
<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>
</P></DIV></DIV><BR>_______________________________________________<BR>--Bandwidth
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