<div>Jun 26 12:43:16 WARNING[31148]: chan_zap.c:8386 pri_dchannel: Ring requested on unconfigured channel 0/16 span 1<br>I noticed this message in the CLI, when I tried to effect one call of HiPath 4000 for asterisk. Ring occurred, however when the voicemail of asterisk took care of call it was dumb, without no sound. I thank the attention
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<div>Regards<br><br>Josué</div>
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<div><span class="gmail_quote">2006/6/26, Josué Conti <<a href="mailto:josueconti@gmail.com">josueconti@gmail.com</a>>:</span>
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<div style="PADDING-RIGHT: 10px; PADDING-LEFT: 10px; PADDING-BOTTOM: 10px; PADDING-TOP: 10px">Hi Richard.</div>
<div style="PADDING-RIGHT: 10px; PADDING-LEFT: 10px; PADDING-BOTTOM: 10px; PADDING-TOP: 10px">Thank you very much for its attention. In the reality what is occurring is that in some originated calls of the HiPath with destination to the Asterisk they are being without the dumb and rings. I do not have this parameter in my HiPath 4000, what I have seemed in the COT is TR6T (1tr6 isdn tie link) would be this parameter? Best Regards
</div>Josué<br><br>
<div><span class="gmail_quote">2006/6/26, richard Coco <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:coco_richard@yahoo.com" target="_blank">coco_richard@yahoo.com</a>>:</span> </div>
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<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid"><br>Hi Josué<br><br>if the Siemens phone calls Asterisk, it didn't get a<br>dial tone from Asterisk? Is it correct?
<br><br>if yes, this is depending of Asterisk which didn't<br>generates a ringback messages as it expexts dial ton<br>generation localy. So try this workaround for HiPath<br>local dial ton generation:<br>-> Add option TR6Q(TRGT) to the class of trunk (COT)
<br>parameters<br><br>hope it will help...<br><br>rich<br><br><br><br><br><br>--- Josué Conti <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:josueconti@gmail.com" target="_blank">josueconti@gmail.com
</a>> wrote:<br><br>> Hello all.<br>> I have installed and functioning asterisk-1.2.9.1<br>> where I effected one<br>> upgrade in asterisk-1.0.9, is interconnected with a<br>> PABX Siemens HiPath 4000
<br>> in ISDN PRI with protocol QSIG, the one that is<br>> happening he is that the <br>> calls originated for PABX Siemens and destined to<br>> SIP phones asterisk are<br>> being without audio, nor Ring, is dumb. They could
<br>> help in this case me?<br>> Best Regards<br>><br>> Josué <br>> > _______________________________________________<br>> --Bandwidth and Colocation provided by <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://easynews.com/" target="_blank">
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