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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:10.0pt;font-family:"Verdana","sans-serif";color:#1F497D'>You need to use relay request in your SBC instead of forward.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","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>Pezhman Lali<br><b>Sent:</b> Tuesday, February 15, 2011 5:42 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] unregistered trunks and registered phones coming from the same IP<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'>please send your sip.conf, is any NAT procedure implemented in your network?<br><br><o:p></o:p></p><div><p class=MsoNormal>On Mon, Feb 14, 2011 at 10:16 PM, Ricardo Carvalho <<a href="mailto:rjcarvalho.lists@gmail.com">rjcarvalho.lists@gmail.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal>Hi,<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>I manage an SBC which stands between my company server farm and some SIP telco trunks. The system works fine, for inbound and outbound calls.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Now I've configured the SBC to also act as a registration proxy, forwarding SIP registrations coming from the Internet to my asterisk servers.<o:p></o:p></p></div><div><p class=MsoNormal>It all seems fine, but it doesn't work well, because by the time at least one phone registers through the SBC to some asterisk server (lets say, server_A), future incoming calls coming from my SIP telco trunks to my server_A got refused by the asterisk running on that server, with "401 Unauthorized" messages back to the SBC.<o:p></o:p></p></div><div><p class=MsoNormal>Seems like that since the moment asterisk binds some contact to the IP of the SBC, because it registered through it, from that moment, asterisk only accepts calls from that IP if those INVITEs carry correct registration to my server (even if those calls came from my SBC, a trusted trunk, not registered in asterisk).<o:p></o:p></p></div><div><p class=MsoNormal>My phones are configured with type=friend. I've also tried type=peer and type=user, but it doesn't solve the problem.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Any ideas to fix this?<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Best regards,<o:p></o:p></p></div><div><p class=MsoNormal>Ricardo.<o:p></o:p></p></div><p class=MsoNormal><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></body></html>