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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>The “Ring” is Asterisk ringing local extensions to indicate an incoming call (DAHDI-1 gets call, rings SIP/1000 and SIP/2000 to let you know you have a call). When you do the *72 forwarding, change the SIP/1000 and SIP/2000 ring to ring local/1.<o:p></o:p></span></p><p class=MsoNormal><span 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 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>motty.cruz<br><b>Sent:</b> Wednesday, May 16, 2012 10:35 AM<br><b>To:</b> 'Asterisk Users Mailing List - Non-Commercial Discussion'<br><b>Subject:</b> Re: [asterisk-users] Asterisk forward call<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Here is the issue, I have Asterisk Server with Digium TDM400P card four ports four anolog lines come throught this server, but we forward all this lines to a voip number (to our main Asterisk Server), for various reason call ID is one of them. </span><o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>When we forward the number to our voip Asterisk Server we get two incoming calls one from our anolog lines because of the partial ring and one from our main Asterisk Server, that is the reason i want to elimitate that partial ring. At this moment when our Asterisk Server with anolog lines get a call it wait 8 seconds before forward that call on. </span><o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Hope i was specific and clear. </span><o:p></o:p></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Thanks, </span><o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><div class=MsoNormal align=center style='text-align:center'><hr size=2 width="100%" align=center></div><p class=MsoNormal style='margin-bottom:12.0pt'><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> <a href="mailto:[mailto:asterisk-users-bounces@lists.digium.com]">[mailto:asterisk-users-bounces@lists.digium.com]</a> <b>On Behalf Of </b>Carlos Alvarez<br><b>Sent:</b> Wednesday, May 16, 2012 7:15 AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Asterisk forward call</span><o:p></o:p></p><p class=MsoNormal>So Asterisk is playing no role during the times you have the calls forwarded, and you just don't want it to ring? Why not just make it go off-hook during those times? <o:p></o:p></p><div><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p><div><p class=MsoNormal>On Tue, May 15, 2012 at 3:42 PM, motty.cruz <<a href="mailto:motty.cruz@gmail.com" target="_blank">motty.cruz@gmail.com</a>> wrote:<o:p></o:p></p><div><p class=MsoNormal><span style='font-family:"Arial","sans-serif";color:blue'>Hello Carlos, </span><o:p></o:p></p><p class=MsoNormal><span style='font-family:"Arial","sans-serif";color:blue'>I'm already using the provider to forward #72 throught my provider, but by default it rings once. at this moment i'm ignoring the first 8 seconds when someone call my number otherwise my phone ring twice. </span><o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal><span style='font-family:"Arial","sans-serif";color:blue'>Thanks, </span><o:p></o:p></p><div class=MsoNormal align=center style='text-align:center'><hr size=2 width="100%" align=center></div><p class=MsoNormal><b><span style='font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-family:"Tahoma","sans-serif"'> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Carlos Alvarez<br><b>Sent:</b> Tuesday, May 15, 2012 2:18 PM <o:p></o:p></span></p><div><p class=MsoNormal><span style='font-family:"Tahoma","sans-serif"'><br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Asterisk forward call<o:p></o:p></span></p></div><p class=MsoNormal><o:p> </o:p></p><div><div><div><p class=MsoNormal>On Tue, May 15, 2012 at 2:00 PM, motty.cruz <<a href="mailto:motty.cruz@gmail.com" target="_blank">motty.cruz@gmail.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal>Any suggestions how to stop the Asterisk from rining once before forward to<br>another number?<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Since analog lines can only signal using ANALOG methods, it will always have a partial ring. Otherwise there's no way for Asterisk to know it is ringing. Why don't you just forward it using the phone company's call forwarding service?<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div></div><p class=MsoNormal>-- <o:p></o:p></p><div><p class=MsoNormal>Carlos Alvarez<o:p></o:p></p></div><div><p class=MsoNormal>TelEvolve<o:p></o:p></p></div><div><p class=MsoNormal><a href="tel:602-889-3003" target="_blank">602-889-3003</a><o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div><div class=MsoNormal align=center style='text-align:center'><hr size=1 width="100%" noshade style='color:#A0A0A0' align=center></div><div><p>No virus found in this message.<br>Checked by AVG - <a href="http://www.avg.com" target="_blank">www.avg.com</a><br>Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12<o:p></o:p></p></div></div><p class=MsoNormal><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p></div><p class=MsoNormal><br><br clear=all><o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal>-- <o:p></o:p></p><div><p class=MsoNormal>Carlos Alvarez<o:p></o:p></p></div><div><p class=MsoNormal>TelEvolve<o:p></o:p></p></div><div><p class=MsoNormal>602-889-3003<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div><div class=MsoNormal align=center style='text-align:center'><hr size=1 width="100%" noshade style='color:#A0A0A0' align=center></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>No virus found in this message.<br>Checked by AVG - <a href="http://www.avg.com">www.avg.com</a><br>Version: 2012.0.2176 / Virus Database: 2425/5002 - Release Date: 05/16/12<o:p></o:p></p></div></body></html>