Thanks for your reply. This all I have, am I missing something? Please help in this regard. Here is full output from CLI
-- Executing [...@default:1] Goto("SIP/501-00000137", "nineoneone,s,1") in new stack -- Goto (nineoneone,s,1) -- Executing [...@nineoneone:1] Set("SIP/501-00000137", "SET_EMERG_FLAG=0") in new stack -- Executing [...@nineoneone:2] ChanIsAvail("SIP/501-00000137", "DAHDI/g0") in new stack -- Executing [...@nineoneone:3] Set("SIP/501-00000137", "EMERGENCY=1,g") in new stack -- Executing [...@nineoneone:4] Set("SIP/501-00000137", "SET_EMERG_FLAG=1") in new stack -- Executing [...@nineoneone:5] Dial("SIP/501-00000137", "DAHDI/g0/91234567") in new stack [Mar 3 11:26:06] WARNING[28572]: app_dial.c:1547 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/501-00000137' status is 'CONGESTION' Regards Shahnawaz On Wed, Mar 3, 2010 at 10:54 AM, Steve Howes <steve-li...@geekinter.net> wrote: > > On 3 Mar 2010, at 17:21, mir shahnawaz wrote: >> [nineoneone] >> exten => s,1,Set(SET_EMERG_FLAG=0) >> exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) >> exten => s,n,Set(EMERGENCY=1,g) >> exten => s,n,Set(SET_EMERG_FLAG=1) >> exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM}) >> exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY}=1]?inprogress) >> exten => s,n,SoftHangup(${EMERGENCY_TRUNK}-1) >> exten => s,n,Wait(12) >> exten => s,n,Goto(checkavail) >> exten => s,s+2(inprogress),Congestion >> exten => s,checkavail+101(notavail),Goto(trunkbusy) >> exten => h,1,GotoIf($["${SET_EMERG_FLAG}"="1"]?3) >> exten => h,3,Set(EMERGENCY=0,g) >> >> If all lines connecting to PSTN are busy. I get busy tone upon dialing >> 911 and following message is generated by CLI. >> >> app_dial.c:1547 dial_exec_full: Unable to create channel of type >> 'DAHDI' (cause 34 - Circuit/channel congestion) > > Can you tell us the other lines too? i.e. the bit where it attempts to > actually do the hangup.. > > S > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users