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</o:shapelayout></xml><![endif]--></head><body bgcolor=white lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='color:#1F497D'>Thank for the input!! <o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>When I do, it fails and a SIP trace shows error: Failed to authenticate on INVITE to "Anonymous" <sip:anonymus@vazycomm.com> <o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>I'm using basically using the "residential" script with some hooks for Asterisk. Asterisk is doing some IVR, once a department / individual is selected I want asterisk to transfer the call to that extension.<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>Again, thanks for any help!!!<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>Juls<o:p></o:p></span></p><p class=MsoNormal><span style='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";color:windowtext'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'> users-bounces@lists.opensips.org [mailto:users-bounces@lists.opensips.org] <b>On Behalf Of </b>Stefano Pisani<br><b>Sent:</b> Thursday, May 07, 2015 4:24 AM<br><b>To:</b> OpenSIPS users mailling list<br><b>Subject:</b> Re: [OpenSIPS-Users] forwarding calls from Asterisk to OpenSIPs<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>Just define a peer in sip.conf<br><br>then Dial(SIP/peer_name/OpensipsAccount) on extensions.conf. Very simple.<br><br>regards,<br>s<br><br>Il 07/05/2015 08:32, Julian Kay ha scritto:<o:p></o:p></p></div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><p class=MsoNormal>Hello Everyone;<o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal>I want to be able to forward calls from Asterisk to a phone registered with OpenSIPs, can anyone give me some information on how this can be done? Or a link to the info I would need to do this.<o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal>Thanks for any help<o:p></o:p></p><p class=MsoNormal>Juls<o:p></o:p></p><p class=MsoNormal><span style='font-size:12.0pt;font-family:"Times New Roman","serif"'><br><br><br><o:p></o:p></span></p><pre>_______________________________________________<o:p></o:p></pre><pre>Users mailing list<o:p></o:p></pre><pre><a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><o:p></o:p></pre><pre><a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><o:p></o:p></pre></blockquote><p class=MsoNormal><span style='font-size:12.0pt;font-family:"Times New Roman","serif"'><o:p> </o:p></span></p></div></body></html>