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<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black">FreePBX is based on Asterisk.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black">Should your call go from Sipwise to asterisk and then to your phone. Or from asterisk to sipwise then to your phone?<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black">To have a correct register from asterisk to sipwise you need to set in sip.conf the from-domain argument as the domain you configured on sipwise.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black">Regards,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black">Max<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black"><o:p> </o:p></span></p>
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">Von:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> spce-user-bounces@lists.sipwise.com [mailto:spce-user-bounces@lists.sipwise.com]
<b>Im Auftrag von </b>Michael Johnson<br>
<b>Gesendet:</b> Donnerstag, 18. April 2013 01:46<br>
<b>An:</b> spce-user@lists.sipwise.com<br>
<b>Betreff:</b> Re: [Spce-user] Set up a peer trunk to a pbx in Sip:Wise CE?????????<o:p></o:p></span></p>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">I am trying to setup peer (trunk) connection from the Sipwise NGCP Administration Interface (subscribers) section to a FreePBX Server. Calls are being sent to the pbx via trunk but in the wrong way.... For example:<br>
<br>
<br>
[2013-04-17 18:15:23] WARNING[1747] chan_sip.c: username mismatch, have <talk-power_trk>, digest has <><br>
[2013-04-17 18:15:23] NOTICE[1747] chan_sip.c: Failed to authenticate device "18506120444" <<a href="mailto:sip%3A18506120444@208.64.8.13">sip:18506120444@208.64.8.13</a>>;tag=1B927343-516F2D0B000B53D9-13FFF700<br>
[2013-04-17 18:15:30] NOTICE[32201] manager.c: Seems to have passed...<br>
[2013-04-17 18:16:23] WARNING[1747] chan_sip.c: username mismatch, have <talk-power_trk>, digest has <><br>
[2013-04-17 18:16:23] NOTICE[1747] chan_sip.c: Failed to authenticate device "18506120444" <<a href="mailto:sip%3A18506120444@184.188.39.232">sip:18506120444@184.188.39.232</a>>;tag=77C89B1D-516F2D47000A4F13-13DFD700<br>
[2013-04-17 18:16:32] NOTICE[32206] manager.c: Seems to have passed...<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">850-612-0444 is my cell phone #: I need this to send like this:<br>
<br>
<<a href="mailto:sip%3A18506340577@184.188.39.232">sip:18506340577@184.188.39.232</a>><o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">Does any one have a guild or any info on how to do this?????????????????????????<o:p></o:p></p>
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<p class="MsoNormal">Thanks,<o:p></o:p></p>
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<p class="MsoNormal">Michael Johnson<o:p></o:p></p>
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