Hi, I was trying to use 2 of asterisk servers and interconnected, one of them as a peer to other sever (configured in sip.conf), so all the calls to server 1 will just be passed to server 2 (has PRI Card, TE 412P, only one PRI connected), i was sending calls to server 1 and that would send to server 2 and then dial out using Dahdi, but the problem that i got was the hangup cause codes, i was not able to pass the appropriate ones to the server 0 (Test server) that sent a call to server 1. For example, when the user at server 0 (test server) made a call to server 1, that sends it to server 2 and connects to the appropriate destination, but in the mean while if we just cancel the call, we need to see the SIP error code as 487 - Request terminated, but I was only able to see the ISDN core in PRI debug on server 2, but was not able to see '487' in sip debug, even though if i am handling the error code conditions........Is there any way of handling the error codes properly....? Asterisk version: 1.4.22.1 Libpri: 1.4.10.1 dahdi: 2.2.0.2 are the versions that I am using. The way I was handling the codes for the server 2: [macro-result] exten => s,1,Wait(1) exten => s,2,ResetCDR(w) exten => s,3,NoCDR() exten => s,4,GotoIf($[${ISNULL(${ARG1})}]?7:5) exten => s,5,Set(RC=${ARG1}) exten => s,6,Goto(s|9) exten => s,7,GotoIf($[${ISNULL(${DIALSTATUS})}]?8:rc-${DIALSTATUS}|1) exten => s,8,Set(RC=${IF($[${ISNULL(${HANGUPCAUSE})}]?0:${HANGUPCAUSE})}) exten => s,9,Goto(rc-${RC}|1) exten => s,10,Hangup(${RC}) exten => i,1,Set(RC=0) exten => i,2,Goto(s|9) exten => rc-ANSWER,1,Set(RC=16) exten => rc-ANSWER,2,Goto(s|9) exten => rc-BUSY,1,Set(RC=17) exten => rc-BUSY,2,Goto(s|9) exten => rc-CANCEL,1,Set(RC=16) exten => rc-CANCEL,2,Goto(s|9) exten => rc-CHANUNAVAIL,1,Set(RC=44) exten => rc-CHANUNAVAIL,2,Goto(s|9) exten => rc-CONGESTION ,1,Set(RC=19) exten => rc-CONGESTION ,2,Goto(s|9) ;exten => rc-NOANSWER,1,Set(RC=19) ;exten => rc-NOANSWER,2,Goto(s|9) exten => rc-0,1,NoOp(NOTDEFINED) exten => rc-0,n,Goto(s|10) exten => rc-1,1,NoOp(UNALLOCATED) exten => rc-1,n,Goto(s|10) exten => rc-2,1,NoOp(NO_ROUTE_TRANSIT_NET) exten => rc-2,n,Goto(s|10) exten => rc-3,1,NoOp(NO_ROUTE_DESTINATION) exten => rc-3,n,Goto(s|10) exten => rc-6,1,NoOp(CHANNEL_UNACCEPTABLE) exten => rc-6,n,Goto(s|10) exten => rc-7,1,NoOp(CALL_AWARDED_DELIVERED) exten => rc-7,n,Goto(s|10) exten => rc-16,1,NoOp(NORMAL_CLEARING) exten => rc-16,n,Goto(s|10) exten => rc-17,1,NoOp(USER_BUSY) ;exten => rc-17,n,Busy() exten => rc-17,n,Goto(s|10) exten => rc-18,1,NoOp(NO_USER_RESPONSE) exten => rc-18,n,Goto(s|10) exten => rc-19,1,NoOp(NO_ANSWER) exten => rc-19,n,Goto(s|10) exten => rc-21,1,NoOp(CALL_REJECTED) exten => rc-21,n,Goto(s|10) exten => rc-28,1,NoOp(INVALID_NUMBER_FORMAT) exten => rc-28,n,Goto(s|10) exten => rc-29,1,NoOp(FACILITY_REJECTED) exten => rc-29,n,Goto(s|10) exten => rc-30,1,NoOp(RESPONSE_TO_STATUS_ENQUIRY) exten => rc-30,n,Goto(s|10) exten => rc-31,1,NoOp(NORMAL_UNSPECIFIED) exten => rc-31,n,Goto(s|10) exten => rc-34,1,NoOp(NORMAL_CIRCUIT_CONGESTION) exten => rc-34,n,Congestion() exten => rc-34,n,Goto(s|10) Thank you for your help. Regards Sandesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100122/ee87ae54/attachment.htm