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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'>KA packets is one (perhaps the stated) function. But, in my experience, you can “run” a “dead” SIP line with qualify=no.<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><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>NaJIm<br><b>Sent:</b> Wednesday, September 14, 2011 4:18 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Confusion with the status of SIP Trunk<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>Oh.. thank you. That could be the reason. Let me try that. <o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>But In fact, I thought qualify = yes is used to send some thing like <i>keep alive</i> packets in an already connected trunk to make sure the trunk is still alive.<o:p></o:p></p></div><div><p class=MsoNormal>In my case the trunk was completely down, and then it was showing status OK as soon as the Internet came up. <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 style='margin-bottom:12.0pt'>Najim<o:p></o:p></p><div><p class=MsoNormal>On Thu, Sep 15, 2011 at 2:02 AM, Danny Nicholas <<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>> wrote:<o:p></o:p></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;color:#1F497D'>If you use qualify=yes, you should only get OK when the line is functional.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span style='font-size:10.0pt'>From:</span></b><span style='font-size:10.0pt'> <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>NaJIm<br><b>Sent:</b> Wednesday, September 14, 2011 3:29 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> [asterisk-users] Confusion with the status of SIP Trunk</span><o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Hi,<o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>I just wanted to clear a doubt I had. In a SIP trunk, will it show "OK" status even if only one side of the SIP trunk is configured when we do " sip show peers " ?? <o:p></o:p></p></div></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>If yes, is there any other way to make sure that the trunk is ready for making calls?? <img border=0 width=14 height=14 id="_x0000_i1025" src="cid:image001.png@01CC72FA.2C9EB210" alt="Description: cid:gtalk.323@goomoji.gmail"><o:p></o:p></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Last day we had a situation here at my Office. There was a fiber cut around our area and one of our leased lines went down. We changed to our secondary line and had to inform our SIP Trunk provider to make the corresponding changes in IP at their end.<o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>But even before they did that my trunk status was showing OK and this caused us a lot of confusion. <o:p></o:p></p></div></div></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Thank You. <img border=0 width=14 height=14 id="_x0000_i1026" src="cid:image002.png@01CC72FA.2C9EB210" alt="Description: cid:gtalk.330@goomoji.gmail"><o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Najim<o:p></o:p></p></div></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><o:p> </o:p></p></div></div></body></html>