pbx1*CLI> sip set debug peer 139 SIP Debugging Enabled for IP: 192.168.1.139 pbx1*CLI> sip set debug peer vgw1 SIP Debugging Enabled for IP: 192.168.1.7 pbx1*CLI> core set verbose 10 Verbosity was 3 and is now 10 == Using SIP RTP CoS mark 5 -- Executing [2155555757@inbound:1] Set("SIP/vgw1-00000014", "DNIS=2155555757") in new stack -- Executing [2155555757@inbound:2] Set("SIP/vgw1-00000014", "CALLERID(num)=6105551111") in new stack -- Executing [2155555757@inbound:3] GotoIf("SIP/vgw1-00000014", "1?override") in new stack -- Goto (inbound,2155555757,12) -- Executing [2155555757@inbound:12] Set("SIP/vgw1-00000014", "CALLERID(name)=Jeremy Cell") in new stack -- Executing [2155555757@inbound:13] Goto("SIP/vgw1-00000014", "attendant,s,1") in new stack -- Goto (attendant,s,1) -- Executing [s@attendant:1] Answer("SIP/vgw1-00000014", "") in new stack -- Executing [s@attendant:2] Dial("SIP/vgw1-00000014", "SIP/111&SIP/112") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/111 == Using SIP RTP CoS mark 5 -- Called SIP/112 -- SIP/111-00000015 connected line has changed. Saving it until answer for SIP/vgw1-00000014 -- SIP/112-00000016 connected line has changed. Saving it until answer for SIP/vgw1-00000014 -- SIP/111-00000015 is ringing -- SIP/112-00000016 is ringing == Using SIP RTP CoS mark 5 [Oct 31 03:14:27] NOTICE[7969]: features.c:7270 ast_pickup_call: pickup SIP/111-00000015 attempt by SIP/139-00000017 -- SIP/139-00000017 answered SIP/vgw1-00000014 <--- SIP read from UDP:192.168.1.7:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK2c5585fa;rport From: ;tag=as0f6253e7 To: ;tag=11977EFF-185A Date: Wed, 31 Oct 2012 07:16:02 GMT Call-ID: A6B3E684-226111E2-BB8695E0-779A0B0@192.168.1.7 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:192.168.1.7:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK2c5585fa;rport From: ;tag=as0f6253e7 To: ;tag=11977EFF-185A Date: Wed, 31 Oct 2012 07:16:02 GMT Call-ID: A6B3E684-226111E2-BB8695E0-779A0B0@192.168.1.7 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Supported: replaces Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Contact: Content-Type: application/sdp Content-Length: 234 v=0 o=CiscoSystemsSIP-GW-UserAgent 6934 543 IN IP4 192.168.1.7 s=SIP Call c=IN IP4 192.168.1.7 t=0 0 m=audio 17486 RTP/AVP 0 101 c=IN IP4 192.168.1.7 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (15 headers 11 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.7:5060 Transmitting (NAT) to 192.168.1.7:5060: ACK sip:9910@192.168.1.7:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK328ff56b;rport Max-Forwards: 70 From: ;tag=as0f6253e7 To: ;tag=11977EFF-185A Contact: Call-ID: A6B3E684-226111E2-BB8695E0-779A0B0@192.168.1.7 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 --- Retransmitting #1 (NAT) to 192.168.1.7:5060: UPDATE sip:9910@192.168.1.7:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5e05bf7c;rport Max-Forwards: 70 From: ;tag=as0f6253e7 To: ;tag=11977EFF-185A Contact: Call-ID: A6B3E684-226111E2-BB8695E0-779A0B0@192.168.1.7 CSeq: 103 UPDATE User-Agent: Asterisk PBX 1.8.15.0 Remote-Party-ID: "testing" ;party=calling;privacy=off;screen=no X-Asterisk-rpid-update: Yes Content-Length: 0 --- <--- SIP read from UDP:192.168.1.7:5060 ---> SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5e05bf7c;rport From: ;tag=as0f6253e7 To: ;tag=11977EFF-185A Call-ID: A6B3E684-226111E2-BB8695E0-779A0B0@192.168.1.7 CSeq: 103 UPDATE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- -- Got SIP response 500 "Internal Server Error" back from 192.168.1.7:5060 == Spawn extension (attendant, s, 2) exited non-zero on 'SIP/vgw1-00000014' Really destroying SIP dialog 'A6B3E684-226111E2-BB8695E0-779A0B0@192.168.1.7' Method: ACK pbx1*CLI> core set verbose 0 Verbosity is now OFF pbx1*CLI> sip set debug off SIP Debugging Disabled