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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='color:#1F497D'>I’m suspecting the answer is you need a CUBE then CUBE to Asterisk<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"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> cisco-voip [mailto:cisco-voip-bounces@puck.nether.net] <b>On Behalf Of </b>Thierry De Vriendt<br><b>Sent:</b> Tuesday, May 07, 2013 9:50 AM<br><b>To:</b> cisco-voip@puck.nether.net<br><b>Subject:</b> [cisco-voip] CCM 8.6 - SIP Register and authentication<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'><span lang=FR-BE style='font-size:12.0pt;font-family:"Times New Roman","serif"'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-family:"Courier New"'>Dear All,<o:p></o:p></span></p><p class=MsoNormal><span style='font-family:"Courier New"'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-family:"Courier New"'>We would like to configure on our Cisco Call Manager 8.6 different SIP trunks to the same Asterisk server (same destination IP). The utility to have different SIP trunks is for billing purposes inside the Asterisk server (1 SIP trunk per customer). The username/password sends over the SIP messages will make the difference. <o:p></o:p></span></p><p class=MsoNormal><span style='font-family:"Courier New"'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-family:"Courier New"'>Is it possible that the Call Manager sends SIP register messages to the Asterisk server (for incoming calls from the Asterisk)?<o:p></o:p></span></p><p class=MsoNormal><span style='font-family:"Courier New"'>Is it possible that the Call Manager sends login/password within the SIP messages when sending a call to the Asterisk server?<o:p></o:p></span></p><p class=MsoNormal><span style='font-family:"Courier New"'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-family:"Courier New"'>I don’t really see in the configuration how to associate username/password with a SIP trunk.<o:p></o:p></span></p><p class=MsoNormal><span style='font-family:"Courier New"'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-family:"Courier New"'>Thanks for your help.<o:p></o:p></span></p><p class=MsoNormal><span style='font-family:"Courier New"'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-family:"Courier New"'>Thierry D.<o:p></o:p></span></p><p class=MsoNormal><span style='font-family:"Courier New"'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-family:"Courier New"'><o:p> </o:p></span></p><p class=MsoNormal><span lang=FR-BE style='font-size:12.0pt;font-family:"Times New Roman","serif"'><br><br><span style='color:white'>itevomcid</span> <o:p></o:p></span></p></div></body></html>