<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"><head><META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii"><meta name=Generator content="Microsoft Word 14 (filtered medium)"><style><!--
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</o:shapelayout></xml><![endif]--></head><body lang=DE link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Hi all,<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>can somebody tell me what i must write into the extension.conf that every call comming from the sipgate account „go“ or answerd by the repeater ?<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>if it is possible that I can only route spezial numbers to the repeater it will be great<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>tnx and greetings tom dh6mbt<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><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"'> app_rpt-users-bounces@ohnosec.org [mailto:app_rpt-users-bounces@ohnosec.org] <b>Im Auftrag von </b>Chris Andrist<br><b>Gesendet:</b> Mittwoch, 4. September 2013 19:20<br><b>An:</b> app_rpt-users@ohnosec.org<br><b>Betreff:</b> Re: [App_rpt-users] USING own SIP for AUTO PATCH<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>Bradley,<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>I actually wrote something up and put it online. You can see it here: <span style='font-size:11.5pt;font-family:".HelveticaNeueUI","serif"'><a href="http://www.allstarnode.com/viewtopic.php?f=6&t=108">http://www.allstarnode.com/viewtopic.php?f=6&t=108</a></span><o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Regards,<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Chris Andrist, KC7WSU<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal style='margin-bottom:12.0pt'><br>On Sep 4, 2013, at 10:00 AM, "<a href="mailto:app_rpt-users-request@ohnosec.org">app_rpt-users-request@ohnosec.org</a>" <<a href="mailto:app_rpt-users-request@ohnosec.org">app_rpt-users-request@ohnosec.org</a>> wrote:<o:p></o:p></p></div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><p class=MsoNormal>2. Re: USING own SIP for AUTO PATCH (Chuck Henderson)<br><br><br>----------------------------------------------------------------------<br><br>Message: 1<br>Date: Tue, 3 Sep 2013 22:05:03 -0500<br>From: Bradley Haney <<a href="mailto:hammin75@gmail.com">hammin75@gmail.com</a>><br>To: <a href="mailto:app_rpt-users@ohnosec.org">app_rpt-users@ohnosec.org</a><br>Subject: [App_rpt-users] USING own SIP for AUTO PATCH<br>Message-ID:<br> <<a href="mailto:CAGjOR2rVeXtN0h9Yw2p0JccZ=YEhxvRSUM9VfZgRYKBGjiPxkQ@mail.gmail.com">CAGjOR2rVeXtN0h9Yw2p0JccZ=YEhxvRSUM9VfZgRYKBGjiPxkQ@mail.gmail.com</a>><br>Content-Type: text/plain; charset="iso-8859-1"<br><br>Hello all. Having an issue with ACID trying to use my own SIP.. It<br>appears when i try to enter in the correct info for the peer and registry<br>statement on the sip.conf and do a astres.sh it bombs out where for what<br>ever reason for the next 10 or so when i log in and do a asterisk -r and<br>do help there are no SIP option on the command line.. then after some time<br>it some how starts up. I thought something was wrong with my system so i<br>put the install disc in again and re downloaded all the software and<br>started over. Everything works fine until i try to add a peer into the<br>SIP.conf I tried the settings for asterisk 1.2, 1.4 and even 1.8 VOIP<br>povider i use is callcentric. Any thoughts? I have been using callcentric<br>for about 2 years on my regular PBX and no issues. ON a side note does<br>anyone have a good inbound extension.conf part that woudl include incoming<br>from my own SIP to be able to have the asterisk answer and let phone<br>control work on the repeater. I would easily pay the 5.00 month for the<br>service, but our members would like to have a local number for our repeater.<br><br>Thanks in advance.<br>Bradley kc9gqr<o:p></o:p></p></blockquote></div></body></html>