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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'>Run Progress before the playtones. This is documented in https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application<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><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>--Eric<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><p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;color:#1F497D'>---<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;color:#1F497D'>Frequently Asked Questions<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;color:#1F497D'>Q1: How do a transfer a call using a Polycom phone?<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;color:#1F497D'>A1: While on a call press the Transfer button on the phone, wait for dialtone, dial the number you want to transfer to, wait for the person answer, tell them you are transferring a call, then press the Transfer button the phone a second time.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;color:#1F497D'>Q2: I don't want to wait for the person answer when transferring a call.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;color:#1F497D'>A2: Press the Transfer buton on the phone, the press the Blind softkey, the dial the extension you want to transfer the call to. The transfer should complete automatically. If it does not, you may need to press the Send softkey<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;color:#1F497D'>Q3: Where can I find more information on using Polycom phones?<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;color:#1F497D'>A3: Go to <a href="http://help.nyigc.net/"><span style='color:blue'>http://help.nyigc.net/</span></a> for documentation for Polycom phones.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;color:#1F497D'>Q4: What is the best kept secret on the Internet? <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;color:#1F497D'>A4: That would be the InterGlobe Help Site, at <a href="http://help.nyigc.net/"><span style='color:blue'>http://help.nyigc.net/</span></a> <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;color:#1F497D'>Q5: When calling my VMAX fax line I always get a busy signal.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;color:#1F497D'>A5: You must call VMAX fax lines either from another VMAX fax line or from a non-VMAX voice line. If you call a VMAX fax line from a VMAX voice line you will always receive a busy signal.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.5pt;font-family:Consolas;color:#1F497D'><o:p> </o:p></span></p></div><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-dev-bounces@lists.digium.com [mailto:asterisk-dev-bounces@lists.digium.com] <b>On Behalf Of </b>Justin Killen<br><b>Sent:</b> Monday, August 04, 2014 11:25 AM<br><b>To:</b> Asterisk Developers Mailing List<br><b>Subject:</b> [asterisk-dev] How to diagnose early media on a PRI<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>I asked this on the users list a week and a half ago but haven’t gotten any response. I’m hoping someone here with PRI/ISDN experience can help guide me in the right direction. <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p> </o:p></span></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>I have a dialplan (freepbx) that plays a busy signal in-band when an extension is busy (before an Answer). Stripped down, it looks like this:<o:p></o:p></span></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:10.0pt;font-family:"Courier New"'>exten => 1005,n,PlayTones(busy)<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New"'>exten => 1005,n,Busy(20)<o:p></o:p></span></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>Note that there is no Answer() prior to this. Our trunk is a PRI.<o:p></o:p></span></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>When I call into this extension from outside, I get about 25 seconds of ringing, followed by a hangup. Looking at the asterisk logs, 20 seconds of that delay is AFTER the PlayTones() function is invoked. I talked with our Telco about this, and they want to refer to in-band tones prior to answer as a media cut-through. The tech said that it is enabled on their end, and he did some test calls and got some ISDN trap logs. He is saying that the PBX is playing the ring-back tone instead of the busy tone, but I don’t think that’s the case (If I add an Answer() to the dialplan, I do in fact hear the busy tone).<o:p></o:p></span></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>Is there anybody out there who has experience with reading/analyzing IDSN trap logs (Q931) that can help me narrow down where the issue is and how to fix it?<o:p></o:p></span></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p> </o:p></span></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>Thanks,<o:p></o:p></span></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>-Justin<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p> </o:p></span></p></div></div></body></html>