<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto">If I’m understanding what you’re trying to achieve it would likely be easiest with a call file. Create one and drop it in /var/spool/asterisk/outgoing <div><br></div><div>Next option would be to use the Asterisk Manager interface with the originate command. <br><br><div id="AppleMailSignature"><span style="font-family: UICTFontTextStyleBody; -webkit-text-size-adjust: auto;">Kind regards,</span><div><span style="font-family: UICTFontTextStyleBody; -webkit-text-size-adjust: auto;"><br></span></div><div><span style="font-family: UICTFontTextStyleBody; -webkit-text-size-adjust: auto;">Matt</span></div></div><div><br>On Nov 22, 2017, at 08:56, Joshua Colp <<a href="mailto:jcolp@digium.com">jcolp@digium.com</a>> wrote:<br><br></div><blockquote type="cite"><div><span>On Wed, Nov 22, 2017, at 09:48 AM, vijay sukumaran nair wrote:</span><br><blockquote type="cite"><span>Dear Joshua,</span><br></blockquote><blockquote type="cite"><span>Thank you for the reply.</span><br></blockquote><blockquote type="cite"><span>As you understand, I do not have SIP request to generate sip_pvt</span><br></blockquote><blockquote type="cite"><span>structure & a SIP channel.</span><br></blockquote><blockquote type="cite"><span>1. Is it possible to trigger dial plan (extensions.conf - default</span><br></blockquote><blockquote type="cite"><span>context) and create SIP request (INVITE) on the new channel I created?</span><br></blockquote><blockquote type="cite"><span>Would this create a SIP channel ? - If yes, how can I do that? -</span><br></blockquote><blockquote type="cite"><span>Is it possible to bind the new channel and the SIP channel created?</span><br></blockquote><blockquote type="cite"><span>2. What are the asterisk API's available to originate a SIP invite</span><br></blockquote><blockquote type="cite"><span>outside of SIP channel?</span><br></blockquote><blockquote type="cite"><span>Thanks, Vijay. </span><br></blockquote><span></span><br><span>Anything in Asterisk can originate an outgoing channel and call a</span><br><span>device. You can use the AMI originate or CLI command as a basis to</span><br><span>understanding that.</span><br><span></span><br><span>What you can't do is pretend a SIP INVITE arrived into Asterisk somehow</span><br><span>and create a SIP channel and send it into the dialplan. You can only</span><br><span>place an outgoing call to a device that once answered goes where you</span><br><span>want.</span><br><span></span><br><span>If this isn't what you need to do then there's nothing really built in</span><br><span>to do what you want precisely.</span><br><span></span><br><span>-- </span><br><span>Joshua Colp</span><br><span>Digium, Inc. | Senior Software Developer</span><br><span>445 Jan Davis Drive NW - Huntsville, AL 35806 - US</span><br><span>Check us out at: <a href="http://www.digium.com">www.digium.com</a> & <a href="http://www.asterisk.org">www.asterisk.org</a></span><br><span></span><br><span>_______________________________________________</span><br><span>asterisk-app-dev mailing list</span><br><span><a href="mailto:asterisk-app-dev@lists.digium.com">asterisk-app-dev@lists.digium.com</a></span><br><span><a href="http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev">http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev</a></span><br></div></blockquote></div></body></html>