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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Henrik Westerberg<br><b>Sent:</b> Wednesday, January 02, 2013 8:02 AM</span><o:p></o:p></p><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>Hi,<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p> </o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>I am using asterisk via AGI and want to be able to record a call.<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>The scenario is:<o:p></o:p></span></p></div><ol start=1 type=1><li class=MsoNormal style='color:black;mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0 level1 lfo3'><span style='font-size:10.5pt;font-family:"Calibri","sans-serif"'>A call comes in<o:p></o:p></span></li><li class=MsoNormal style='color:black;mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0 level1 lfo3'><span style='font-size:10.5pt;font-family:"Calibri","sans-serif"'>The call is redirected to a mobile number via a local extension and ChannelRedirect<o:p></o:p></span></li><li class=MsoNormal style='color:black;mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0 level1 lfo3'><span style='font-size:10.5pt;font-family:"Calibri","sans-serif"'>The local extension looks like something this:<o:p></o:p></span></li></ol><div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>exten => _X.,1,Dial(SIP/${EXTEN},60,…)<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>exten => _X.,n,Agi(agi://localhost/aj.agi?action=……..)<o:p></o:p></span></p></div><div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p> </o:p></span></p></div></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>I have looked through all arguments of Dial but haven't found any way to continue having a connected call between the caller and the callee and have AGI control of it. Is there a way to do this or do I have to use G() and connect the both ends to AGI separately and then bridging them before recording the call?<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'><o:p> </o:p></span></p></div></div><div><p class=MsoNormal><span style='font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'>Thanks for help.<o:p></o:p></span></p></div><div><div><p class=MsoNormal><span style='font-size:13.5pt;font-family:Consolas;color:black'><o:p> </o:p></span></p></div><div><p class=MsoNormal><span style='font-size:13.5pt;font-family:Consolas;color:black'>Regards,<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:13.5pt;font-family:Consolas;color:black'><o:p> </o:p></span></p></div><div><p class=MsoNormal><span style='font-size:13.5pt;font-family:Consolas;color:black'>Henrik<o:p></o:p></span></p><p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Danny Nicholas<br><b>Sent:</b> Wednesday, January 02, 2013 8:18 AM</span><o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Put the AGI call in a macro context and add M(macro) to your Dial string.<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>Don Kelly<br><b>Sent:</b> Wednesday, January 02, 2013 9:32 AM<br><b>To:</b> 'Asterisk Users Mailing List - Non-Commercial Discussion'<br><b>Subject:</b> Re: [asterisk-users] Dialing out and recording<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I have the same requirement, but it’s important that the caller ID information from the original caller is presented to the destination and we announce the call before the “transfer” is complete. The carrier requires a diversion header if the ANI is not one of “our” DIDs. Does someone have experience with this working?<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'>Two suggestions for you, Don. #1 if the Dial is “Private” the “announcement” is taken care of. #2 I’m supposing that you could do a “SIP Header” command before the Dial to resolve the diversion header issue.<o:p></o:p></span></p></div></div></div></body></html>