I'm trying to add a second extension to my setup. The second device is able to successfully connect to the Asterisk server. I am unable to contact extension 101 from 102 and vise-versa. Also are my context setup logically or is there a better fashion to organize them? My error is at the bottom.
Here is the extension.conf [default] ; ; By default we include the demo. In a production system, you ; probably don't want to have the demo there. ; ;include => demo exten => 101,1,Dial(SIP/101&SIP/[EMAIL PROTECTED],30) exten => 101,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?lbl_default_1:) exten => 101,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?lbl_default_1:) exten => 101,n(lbl_default_0),Hangup() exten => 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED],30) exten => 101,n,Goto(lbl_default_0) exten => 102,1,Dial(SIP/102,20) exten => 102,n,Hangup ;This automatically calls the right mailbox using the ${CALLERIDNUM} variable in the current context (var ${CONTEXT}). exten=>*98,1,VoiceMailMain([EMAIL PROTECTED]) include => inbound include => outgoing [inbound] exten => 9045622082,1,Goto(default,101,1) [outgoing] ; The following gives an Unknown Caller ID ;exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=XXXXXXXXXX) ;exten => _1NXXNXXXXXX,2,Set(CALLERID(name)=XXXXXXXXXX) exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=9045622082) exten => _1NXXNXXXXXX,n,Set(CALLERID(name)="Stephen Reese") exten => _1NXXNXXXXXX,n,Dial(SIP/[EMAIL PROTECTED]) exten => _NXXXXXX,1,Set(CALLERID(num)=9045622082) exten => _NXXXXXX,n,Set(CALLERID(name)="Stephen Reese") exten => _NXXXXXX,n,Dial(SIP/[EMAIL PROTECTED]) exten => _NXXNXXXXXX,1,Set(CALLERID(num)=9045622082) exten => _NXXNXXXXXX,n,Set(CALLERID(name)="Stephen Reese") exten => _NXXNXXXXXX,n,Dial(SIP/[EMAIL PROTECTED]) exten => _011.,1,Set(CALLERID(num)=9045622082) exten => _011.,n,Set(CALLERID(name)="Stephen Reese") exten => _011.,n,Dial(SIP/[EMAIL PROTECTED]) exten => _911,1,Set(CALLERID(num)=9045622082) exten => _911,n,Set(CALLERID(name)="Stephen Reese") exten => _911,n,Dial(SIP/[EMAIL PROTECTED]) This is a call from extension 101 to 102 that fails with a busy signal. -- Executing [EMAIL PROTECTED]:1] Dial("SIP/101-08266f60", "'SIP/102',20") in new stack [Oct 19 15:28:28] WARNING[26596]: channel.c:3470 ast_request: No channel type registered for ''SIP' [Oct 19 15:28:28] WARNING[26596]: app_dial.c:1450 dial_exec_full: Unable to create channel of type ''SIP' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:2] Hangup("SIP/101-08266f60", "") in new stack == Spawn extension (default, 102, 2) exited non-zero on 'SIP/101-08266f60' _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users