<p>I'm propecting a deal where the installation will be:</p><p>SIEMENS Switch <-------ss7------>ASTERISK<-------pri------>NexSpan<-------->INTERNAL USERS;<br></p><p> |------------------pri------->PSTN</p>
<p>to connect asterisk to the switch i have choice between ss7 and pri; </p><p>i want experiment ss7 here ;)</p><p>the purpose of this is to have a full control on all the calls.</p><p>btw how would you calibrate the asterisk server?</p>
<p>regards,</p><br><div><span class="gmail_quote">On 7/11/08, <b class="gmail_sendername">Rony Ron</b> <<a href="mailto:upcomingbiz@gmail.com">upcomingbiz@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="margin:0;margin-left:0.8ex;border-left:1px #ccc solid;padding-left:1ex">
<p>please find below copy of the config part that deals with the PRI ( i don't have the alcatel's config)</p><p>the schema of the installation was :</p><p>PSTN <--------pri-------->GSM-GW<-------pri------->***ASTERISK***<-----pri------>ALCATEL 4200</p>
<p>;*************** PRI-GSM-GW************<br>group=1<br>context=from-bluetower<br>switchtype=euroisdn<br>pridialplan=unknown<br>callgroup=1<br>pickupgroup=1<br>immediate=no<br>musiconhold=default<br>signalling=pri_cpe<br>
channel=>1-15,17-31<br>;<br>usercallerid=yes<br>hidecallerid=no<br>restrictcid=no<br>callerid=asreceived<br>usercallingpress=yes<br>callwaitingcallerid=yes<br>threewaycalling=yes<br>transfer=yes<br>cancallforward=yes<br>
callreturn=yes<br>;echocancell=yes<br>rxgain=0.0<br>txgain=0.0<br><br>;************ PRI-ALCATEL 4200 **************<br>group=2<br>context=from-alc4200<br>immediate=no<br>overlapdial=yes<br>signalling=pri_net<br>channel=>32-46,48-62<br>
<br></p><p>BR,</p><br><div><span class="q"><span class="gmail_quote">On 7/11/08, <b class="gmail_sendername">Krzysztof Drewicz</b> <<a href="mailto:krzysztofdrewicz@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">krzysztofdrewicz@gmail.com</a>> wrote:</span></span><blockquote class="gmail_quote" style="margin:0;margin-left:0.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><span class="e" id="q_11b12fc751cf16ef_3">
2008/7/11 Rony Ron <<a href="mailto:upcomingbiz@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">upcomingbiz@gmail.com</a>>:<span><br><div class="gmail_quote"><blockquote class="gmail_quote" style="border-left:1px solid rgb(204, 204, 204);margin:0pt 0pt 0pt 0.8ex;padding-left:1ex">
<p>I don't know if it will help i have used digium pri card as asterisk's interface to an alcatel 4400 (or 4200) </p><p>and call transfer is working fine (since more than 1 year).<br></p></blockquote></div><br></span>Kindlly please say if you have used a PRI with QSIG or DSS-1 protocol?<br>
What side was master (network) and what was a slave?<br><br>do you have a backup of mao from PABX and *.conf files from Asterisk? if so, Please forward it to me to my email, I whoul like to test it in my labo.<br><br>TIA.<div>
<span><br>
<br>-- <br>Krzysztof Drewicz<br>+48 504 17 55 77
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