<div>Following working config might help you.</div><div><br></div><div><div>Asterisk 1.6.2.10</div></div><div>dahdi-linux-complete-2.3.0.1+2.3.0</div><div>libss7-1.0.2</div><div><br></div><div>Latest version of asterisk and dahdi should work.</div>
<div><br></div><div>/etc/dahdi/system.conf</div><div>#</div><div><div># Global data</div><div><br></div><div>loadzone = us</div><div>defaultzone = us</div><div><br></div><div># Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS </div>
<div>span=1,1,0,ccs,hdb3</div><div>bchan=1-15,17-31</div><div>mtp2=16</div><div>echocanceller=mg2,1-15,17-31</div><div><br></div><div># Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" HDB3/CCS </div><div>span=2,2,0,ccs,hdb3</div>
<div>bchan=32-46,48-62</div><div>echocanceller=mg2,32-46,48-62</div></div><div><br></div><div>/etc/asterisk/chan_dahdi.conf</div><div><div>;Dahdi Channels Configurations </div><div><br></div><div>[channels]</div><div>context=default</div>
<div>usecallerid=yes</div><div>hidecallerid=no</div><div>callwaiting=yes</div><div>usecallingpres=yes</div><div>callwaitingcallerid=yes</div><div>threewaycalling=yes</div><div>transfer=yes</div><div>canpark=yes</div><div>
cancallforward=yes</div><div>callreturn=yes</div><div>echocancel=yes</div><div>echocancelwhenbridged=yes</div><div>relaxdtmf=yes</div><div>rxgain=0.0</div><div>txgain=0.0</div><div>immediate=no</div><div>signalling=ss7</div>
</div><div><div>ss7type=itu</div><div>ss7_called_nai=national</div><div>ss7_calling_nai=national</div><div>networkindicator=national</div><div>ss7_internationalprefix=00</div><div>ss7_nationalprefix=0</div><div><br></div>
<div>linkset=1</div><div>pointcode=1234 ;OPC - Our side</div><div>defaultdpc=5678 ;DPC - MSC end</div><div>adjpointcode=9012 ;ACP - MSC end </div><div><br></div><div>group=0</div><div><div><br></div>
<div>;port 1</div><div>sigchan=16</div><div>cicbeginswith=1</div><div>channel=1-15</div><div>cicbeginswith=17</div><div>channel=17-31</div><div><br></div><div>;port 2</div><div>sigchan=47</div><div>cicbeginswith=33</div>
<div>
channel=32-46</div><div>cicbeginswith=49</div><div>channel=48-62</div><div><br></div></div></div><br><br><div class="gmail_quote">On Mon, Mar 14, 2011 at 4:20 PM, tanveer khan <span dir="ltr"><<a href="mailto:tanveerhk@hotmail.com">tanveerhk@hotmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div>
Dear Basit,<br><br>thanks for your reply. i dont want to use chan_ss7, i tries almost every version of asterisk, dahdi, zaptel.<br><br>i am using the following card<br><br>pci:0000:05:00.0 wcte11xp+ e159:0001 Digium Wildcard TE110P T1/E1 Board<br>
<br><br>i you have some running system kindly share your version details and configurations.<br><br><br>reinstalling asterisk dahdi and libss7.<br><br><br>regards,<br><font color="#888888"><br>Tanveer<br></font><div class="hm">
<br><hr>From: <a href="mailto:basit.engg@gmail.com" target="_blank">basit.engg@gmail.com</a><br>Date: Mon, 14 Mar 2011 15:19:10 +0500<br>Subject: Re: [asterisk-ss7] facing problem with libss7<br>To: <a href="mailto:asterisk-ss7@lists.digium.com" target="_blank">asterisk-ss7@lists.digium.com</a><br>
CC: <a href="mailto:tanveerhk@hotmail.com" target="_blank">tanveerhk@hotmail.com</a></div><div><div></div><div class="h5"><br><br>Welcome to ss7. <div>Libss7 is good and work able with asterisk. You can use chan_ss7 as well.</div>
<div><br></div><div>What are asterisk, dahdi and libss7 versions? What cards you are using for interconnect?</div><div>Have you done any media verification test? </div>
<div><br></div><div>Also share your configs. We may help you identifying the issue.</div><div><br></div><div><br></div><div><br><br><div>On Mon, Mar 14, 2011 at 3:10 PM, tanveer khan <span dir="ltr"><<a href="mailto:tanveerhk@hotmail.com" target="_blank">tanveerhk@hotmail.com</a>></span> wrote:<br>
<blockquote style="border-left:1px solid rgb(204, 204, 204);padding-left:1ex">
<div>
Dear All,<br><br>i am new to libss7 and having many problem. i was able to install different version of libss7, dahid and asterisk. but when i test it. each contain some problem i.e<br><br>1. continuous rbt on ss7 side even i answer the call on sip side.<br>
2. one sides voice( no voice to user on ss7 side).<br><br><br>could some one recommend me the working version for asterisk, zaptel, dahdi etc?<br><br>is it safe to use libss7 based interconnect in production environment?<br>
<br><br>thanks in advance every one.<br><br><br><br>Regards,<br><br><br>Tanveer<br><br><br><br> </div>
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