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</o:shapelayout></xml><![endif]--></head><body lang=EN-AU link=blue vlink=purple><div class=WordSection1><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>I was trying to do a SIP call between two Asterisk servers (1.4.21.2)<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'><o:p> </o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>1) On the caller server, I coded the following in extensions.conf<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>Dial(SIP/1166:password@asterisk-callee);<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'><o:p> </o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>2) On the callee server, I coded the following in sip.conf<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>[1166]<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>type=friend ; Friends place calls and receive calls<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>context=incoming ; Context for incoming calls from this user<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>host=dynamic ; This peer register with us<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>qualify=yes ; Monitor latency between Asterisk server and phone<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>call-limit=99<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>username=1166 ; Username to use in INVITE until peer registers<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>secret=password ; Normally you do NOT need to set this parameter<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>mailbox=1166@default ; mailbox 5100 in voicemail context .default.<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>callgroup=1<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>pickupgroup=1<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'><o:p> </o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>The call was unsuccessful as follows.<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'> <o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>1) On the caller machine, this is what we got from the console<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'> -- Executing [1166@incoming:1] Dial("SIP/1166-09d81668", "SIP/1166:password@asterisk-callee") in new stack<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'> -- Called 1166:password@asterisk-callee<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'> -- SIP/asterisk-callee is circuit-busy<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'> == Everyone is busy/congested at this time (1:0/1/0)<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'><o:p> </o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>2) On the callee machine, this is what we got from the console,<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>[Sep 14 14:34:12] NOTICE[11991]: chan_sip.c:14035 handle_request_invite: Call from '2765' to extension '1166:password' rejected because extension not found.<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'><o:p> </o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>However, I found out that if I remove “secret=..” from the SIP entry and call without the password, then I will be able to call.<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'><o:p> </o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'>Any thoughts?<o:p></o:p></span></p><p class=MsoPlainText><span style='font-size:10.0pt;font-family:"Courier New"'><o:p> </o:p></span></p><p class=MsoNormal><o:p> </o:p></p></div></body></html>