<--- SIP read from UDP:192.168.1.5:12592 ---> INVITE sip:299@192.168.1.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:12592;branch=z9hG4bK-d8754z-2e5158118100bf63-1---d8754z-;rport Max-Forwards: 70 Contact: To: "299" From: ;tag=11788805 Call-ID: MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 214 v=0 o=- 1 2 IN IP4 192.168.1.5 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.5 t=0 0 m=audio 26162 RTP/AVP 107 0 8 3 101 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (12 headers 10 lines) --- == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 Sending to 192.168.1.5 : 12592 (no NAT) Using INVITE request as basis request - MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM. Found peer '201' for '201' from 192.168.1.5:12592 <--- Reliably Transmitting (NAT) to 192.168.1.5:12592 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.5:12592;branch=z9hG4bK-d8754z-2e5158118100bf63-1---d8754z-;received=192.168.1.5;rport=12592 From: ;tag=11788805 To: "299";tag=as5637521e Call-ID: MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM. CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7998902f" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM.' in 6656 ms (Method: INVITE) <--- SIP read from UDP:192.168.1.5:12592 ---> ACK sip:299@192.168.1.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:12592;branch=z9hG4bK-d8754z-2e5158118100bf63-1---d8754z-;rport To: "299";tag=as5637521e From: ;tag=11788805 Call-ID: MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM. CSeq: 1 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:192.168.1.5:12592 ---> INVITE sip:299@192.168.1.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:12592;branch=z9hG4bK-d8754z-1134f10557266c2e-1---d8754z-;rport Max-Forwards: 70 Contact: To: "299" From: ;tag=11788805 Call-ID: MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Authorization: Digest username="201",realm="asterisk",nonce="7998902f",uri="sip:299@192.168.1.111",response="5f04108bc1f9ac781bb4a15f210781f7",algorithm=MD5 Content-Length: 214 v=0 o=- 1 2 IN IP4 192.168.1.5 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.5 t=0 0 m=audio 26162 RTP/AVP 107 0 8 3 101 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (13 headers 10 lines) --- Sending to 192.168.1.5 : 12592 (NAT) Using INVITE request as basis request - MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM. Found peer '201' for '201' from 192.168.1.5:12592 Found RTP audio format 107 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Found audio description format BV32 for ID 107 Found audio description format telephone-event for ID 101 Capabilities: us - 0x100a (gsm|alaw|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xa (gsm|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.5:26162 Looking for 299 in from-internal (domain 192.168.1.111) list_route: hop: <--- Transmitting (NAT) to 192.168.1.5:12592 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.5:12592;branch=z9hG4bK-d8754z-1134f10557266c2e-1---d8754z-;received=192.168.1.5;rport=12592 From: ;tag=11788805 To: "299" Call-ID: MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM. CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [299@from-internal:1] Macro("SIP/201-0000000e", "user-callerid,") in new stack -- Executing [s@macro-user-callerid:1] Set("SIP/201-0000000e", "AMPUSER=201") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("SIP/201-0000000e", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("SIP/201-0000000e", "1?Set(REALCALLERIDNUM=201)") in new stack -- Executing [s@macro-user-callerid:4] Set("SIP/201-0000000e", "AMPUSER=201") in new stack -- Executing [s@macro-user-callerid:5] Set("SIP/201-0000000e", "AMPUSERCIDNAME=201") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("SIP/201-0000000e", "0?report") in new stack -- Executing [s@macro-user-callerid:7] Set("SIP/201-0000000e", "AMPUSERCID=201") in new stack -- Executing [s@macro-user-callerid:8] Set("SIP/201-0000000e", "CALLERID(all)="201" <201>") in new stack -- Executing [s@macro-user-callerid:9] ExecIf("SIP/201-0000000e", "0?Set(CHANNEL(language)=)") in new stack -- Executing [s@macro-user-callerid:10] GotoIf("SIP/201-0000000e", "0?continue") in new stack -- Executing [s@macro-user-callerid:11] Set("SIP/201-0000000e", "__TTL=64") in new stack -- Executing [s@macro-user-callerid:12] GotoIf("SIP/201-0000000e", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] NoOp("SIP/201-0000000e", "Using CallerID "201" <201>") in new stack -- Executing [299@from-internal:2] Answer("SIP/201-0000000e", "") in new stack Audio is at 192.168.1.111 port 10320 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.1.5:12592 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.5:12592;branch=z9hG4bK-d8754z-1134f10557266c2e-1---d8754z-;received=192.168.1.5;rport=12592 From: ;tag=11788805 To: "299";tag=as303ba4df Call-ID: MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM. CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 260 v=0 o=root 658433285 658433285 IN IP4 192.168.1.111 s=Asterisk PBX 1.6.2.11 c=IN IP4 192.168.1.111 t=0 0 m=audio 10320 RTP/AVP 8 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> -- Executing [299@from-internal:3] Set("SIP/201-0000000e", "__BLKVM_OVERRIDE=BLKVM/299/SIP/201-0000000e") in new stack -- Executing [299@from-internal:4] Set("SIP/201-0000000e", "__BLKVM_BASE=299") in new stack -- Executing [299@from-internal:5] Set("SIP/201-0000000e", "DB(BLKVM/299/SIP/201-0000000e)=TRUE") in new stack -- Executing [299@from-internal:6] ExecIf("SIP/201-0000000e", "1?Set(_DIAL_OPTIONS=tkM(auto-blkvm))") in new stack -- Executing [299@from-internal:7] Set("SIP/201-0000000e", "__NODEST=299") in new stack -- Executing [299@from-internal:8] Set("SIP/201-0000000e", "MONITOR_FILENAME=/var/spool/asterisk/monitor/q299-20100819-144755-1282222075.19") in new stack -- Executing [299@from-internal:9] Queue("SIP/201-0000000e", "299,t,,") in new stack -- Started music on hold, class 'default', on SIP/201-0000000e -- Executing [202@from-queue:1] Set("Local/202@from-queue-af93;2", "QAGENT=202") in new stack -- Executing [202@from-queue:2] Goto("Local/202@from-queue-af93;2", "299,1") in new stack -- Goto (from-queue,299,1) -- Executing [299@from-queue:1] Goto("Local/202@from-queue-af93;2", "from-internal,202,1") in new stack -- Goto (from-internal,202,1) -- Executing [202@from-internal:1] Macro("Local/202@from-queue-af93;2", "exten-vm,novm,202") in new stack -- Executing [s@macro-exten-vm:1] GotoIf("Local/202@from-queue-af93;2", "0?ext-intercom,nointercom202,1") in new stack -- Executing [s@macro-exten-vm:2] GotoIf("Local/202@from-queue-af93;2", "0?ext-intercom,nointercom202,1") in new stack -- Executing [s@macro-exten-vm:3] Macro("Local/202@from-queue-af93;2", "user-callerid") in new stack -- Executing [s@macro-user-callerid:1] Set("Local/202@from-queue-af93;2", "AMPUSER=201") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("Local/202@from-queue-af93;2", "1?report") in new stack -- Goto (macro-user-callerid,s,10) -- Executing [s@macro-user-callerid:10] GotoIf("Local/202@from-queue-af93;2", "0?continue") in new stack -- Executing [s@macro-user-callerid:11] Set("Local/202@from-queue-af93;2", "__TTL=63") in new stack -- Executing [s@macro-user-callerid:12] GotoIf("Local/202@from-queue-af93;2", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] NoOp("Local/202@from-queue-af93;2", "Using CallerID "201" <201>") in new stack -- Executing [s@macro-exten-vm:4] Set("Local/202@from-queue-af93;2", "RingGroupMethod=none") in new stack -- Executing [s@macro-exten-vm:5] Set("Local/202@from-queue-af93;2", "VMBOX=novm") in new stack -- Executing [s@macro-exten-vm:6] Set("Local/202@from-queue-af93;2", "EXTTOCALL=202") in new stack -- Executing [s@macro-exten-vm:7] Set("Local/202@from-queue-af93;2", "CFUEXT=") in new stack -- Executing [s@macro-exten-vm:8] Set("Local/202@from-queue-af93;2", "CFBEXT=") in new stack -- Executing [s@macro-exten-vm:9] Set("Local/202@from-queue-af93;2", "RT=""") in new stack -- Executing [s@macro-exten-vm:10] Macro("Local/202@from-queue-af93;2", "record-enable,202,IN") in new stack -- Executing [s@macro-record-enable:1] MacroExit("Local/202@from-queue-af93;2", "") in new stack -- Executing [s@macro-exten-vm:11] Macro("Local/202@from-queue-af93;2", "dial,,tkM(auto-blkvm),202") in new stack -- Executing [s@macro-dial:1] GotoIf("Local/202@from-queue-af93;2", "1?dial") in new stack -- Goto (macro-dial,s,3) -- Executing [s@macro-dial:3] AGI("Local/202@from-queue-af93;2", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- Executing [203@from-queue:1] Set("Local/203@from-queue-9f48;2", "QAGENT=203") in new stack -- Executing [203@from-queue:2] Goto("Local/203@from-queue-9f48;2", "299,1") in new stack -- Goto (from-queue,299,1) -- Executing [299@from-queue:1] Goto("Local/203@from-queue-9f48;2", "from-internal,203,1") in new stack -- Goto (from-internal,203,1) -- Executing [203@from-internal:1] Macro("Local/203@from-queue-9f48;2", "exten-vm,novm,203") in new stack <--- SIP read from UDP:192.168.1.5:12592 ---> ACK sip:299@192.168.1.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:12592;branch=z9hG4bK-d8754z-cf21da15e1698211-1---d8754z-;rport Max-Forwards: 70 Contact: To: "299";tag=as303ba4df From: ;tag=11788805 Call-ID: MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM. CSeq: 2 ACK User-Agent: X-Lite release 1104o stamp 56125 Authorization: Digest username="201",realm="asterisk",nonce="7998902f",uri="sip:299@192.168.1.111",response="5f04108bc1f9ac781bb4a15f210781f7",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- -- Executing [s@macro-exten-vm:1] GotoIf("Local/203@from-queue-9f48;2", "0?ext-intercom,nointercom203,1") in new stack -- Executing [s@macro-exten-vm:2] GotoIf("Local/203@from-queue-9f48;2", "0?ext-intercom,nointercom203,1") in new stack -- Executing [s@macro-exten-vm:3] Macro("Local/203@from-queue-9f48;2", "user-callerid") in new stack -- Executing [s@macro-user-callerid:1] Set("Local/203@from-queue-9f48;2", "AMPUSER=201") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("Local/203@from-queue-9f48;2", "1?report") in new stack -- Goto (macro-user-callerid,s,10) -- Executing [s@macro-user-callerid:10] GotoIf("Local/203@from-queue-9f48;2", "0?continue") in new stack -- Executing [s@macro-user-callerid:11] Set("Local/203@from-queue-9f48;2", "__TTL=63") in new stack -- Executing [s@macro-user-callerid:12] GotoIf("Local/203@from-queue-9f48;2", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] NoOp("Local/203@from-queue-9f48;2", "Using CallerID "201" <201>") in new stack -- Executing [s@macro-exten-vm:4] Set("Local/203@from-queue-9f48;2", "RingGroupMethod=none") in new stack -- Executing [s@macro-exten-vm:5] Set("Local/203@from-queue-9f48;2", "VMBOX=novm") in new stack -- Executing [s@macro-exten-vm:6] Set("Local/203@from-queue-9f48;2", "EXTTOCALL=203") in new stack -- Executing [s@macro-exten-vm:7] Set("Local/203@from-queue-9f48;2", "CFUEXT=") in new stack -- Executing [s@macro-exten-vm:8] Set("Local/203@from-queue-9f48;2", "CFBEXT=") in new stack -- Executing [s@macro-exten-vm:9] Set("Local/203@from-queue-9f48;2", "RT=""") in new stack -- Executing [s@macro-exten-vm:10] Macro("Local/203@from-queue-9f48;2", "record-enable,203,IN") in new stack -- Executing [s@macro-record-enable:1] MacroExit("Local/203@from-queue-9f48;2", "") in new stack -- Executing [s@macro-exten-vm:11] Macro("Local/203@from-queue-9f48;2", "dial,,tkM(auto-blkvm),203") in new stack -- Executing [s@macro-dial:1] GotoIf("Local/203@from-queue-9f48;2", "1?dial") in new stack -- Goto (macro-dial,s,3) -- Executing [s@macro-dial:3] AGI("Local/203@from-queue-9f48;2", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi dialparties.agi: Caller ID name is '201' number is '201' dialparties.agi: Methodology of ring is 'none' -- dialparties.agi: Added extension 202 to extension map -- dialparties.agi: Extension 202 cf is disabled -- dialparties.agi: Extension 202 do not disturb is disabled dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE) dialparties.agi: Extension 202 has ExtensionState: 0 -- dialparties.agi: Checking CW and CFB status for extension 202 -- dialparties.agi: dbset CALLTRACE/202 to 201 -- dialparties.agi: Filtered ARG3: 202 -- AGI Script dialparties.agi completed, returning 0 dialparties.agi: Starting New Dialparties.agi -- Executing [s@macro-dial:7] Dial("Local/202@from-queue-af93;2", "SIP/202,,tkM(auto-blkvm)") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 Audio is at 192.168.1.111 port 10426 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x1000 (g722) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.1.52:5060: INVITE sip:202@192.168.1.52:5060;line=budunjf2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK3e5aa209;rport Max-Forwards: 70 From: "201" ;tag=as0ec9c5cf To: Contact: Call-ID: 429f11b251cb8f722c27bbfd215259ec@192.168.1.111 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.11 Date: Thu, 19 Aug 2010 12:47:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 286 v=0 o=root 1941640983 1941640983 IN IP4 192.168.1.111 s=Asterisk PBX 1.6.2.11 c=IN IP4 192.168.1.111 t=0 0 m=audio 10426 RTP/AVP 8 3 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called 202 dialparties.agi: Caller ID name is '201' number is '201' dialparties.agi: Methodology of ring is 'none' -- dialparties.agi: Added extension 203 to extension map -- dialparties.agi: Extension 203 cf is disabled -- dialparties.agi: Extension 203 do not disturb is disabled dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE) dialparties.agi: Extension 203 has ExtensionState: 0 -- dialparties.agi: Checking CW and CFB status for extension 203 -- dialparties.agi: dbset CALLTRACE/203 to 201 -- dialparties.agi: Filtered ARG3: 203 -- AGI Script dialparties.agi completed, returning 0 -- Executing [s@macro-dial:7] Dial("Local/203@from-queue-9f48;2", "SIP/203,,tkM(auto-blkvm)") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 Audio is at 192.168.1.111 port 10220 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x1000 (g722) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.1.213:5060: NVITE sip:203@192.168.1.213:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5e0e1565;rport Max-Forwards: 70 From: "201" ;tag=as2330a1d9 To: Contact: Call-ID: 3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.11 Date: Thu, 19 Aug 2010 12:47:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 286 v=0 o=root 1910113364 1910113364 IN IP4 192.168.1.111 s=Asterisk PBX 1.6.2.11 c=IN IP4 192.168.1.111 t=0 0 m=audio 10220 RTP/AVP 8 3 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called 203 <--- SIP read from UDP:192.168.1.52:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK3e5aa209;rport=5060 From: "201" ;tag=as0ec9c5cf To: ;tag=nvbtn81cu6 Call-ID: 429f11b251cb8f722c27bbfd215259ec@192.168.1.111 CSeq: 102 INVITE Contact: ;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/202-0000000f is ringing -- Local/202@from-queue-af93;1 is ringing <--- SIP read from UDP:192.168.1.213:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5e0e1565;rport From: "201" ;tag=as2330a1d9 To: Call-ID: 3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111 CSeq: 102 INVITE User-Agent: Grandstream BT200 1.2.2.19 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:192.168.1.213:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5e0e1565;rport From: "201" ;tag=as2330a1d9 To: ;tag=e516915c3d248474 Call-ID: 3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111 CSeq: 102 INVITE User-Agent: Grandstream BT200 1.2.2.19 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/203-00000010 is ringing -- Local/203@from-queue-9f48;1 is ringing <--- SIP read from UDP:192.168.1.52:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK3e5aa209;rport=5060 From: "201" ;tag=as0ec9c5cf To: ;tag=nvbtn81cu6 Call-ID: 429f11b251cb8f722c27bbfd215259ec@192.168.1.111 CSeq: 102 INVITE Contact: ;reg-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/202-0000000f is ringing <--- SIP read from UDP:192.168.1.52:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK3e5aa209;rport=5060 From: "201" ;tag=as0ec9c5cf To: ;tag=nvbtn81cu6 Call-ID: 429f11b251cb8f722c27bbfd215259ec@192.168.1.111 CSeq: 102 INVITE Contact: ;reg-id=1 User-Agent: snom300/7.3.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 267 v=0 o=root 1230806085 1230806086 IN IP4 192.168.1.52 s=call c=IN IP4 192.168.1.52 t=0 0 m=audio 51580 RTP/AVP 8 3 9 101 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:9 g722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (13 headers 13 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 9 Found RTP audio format 101 Found audio description format pcma for ID 8 Found audio description format gsm for ID 3 Found audio description format g722 for ID 9 Found audio description format telephone-event for ID 101 Capabilities: us - 0x100a (gsm|alaw|g722), peer - audio=0x100a (gsm|alaw|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100a (gsm|alaw|g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.52:51580 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.52, port 5060 Transmitting (NAT) to 192.168.1.52:5060: ACK sip:202@192.168.1.52:5060;line=budunjf2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK475078ee;rport Max-Forwards: 70 From: "201" ;tag=as0ec9c5cf To: ;tag=nvbtn81cu6 Contact: Call-ID: 429f11b251cb8f722c27bbfd215259ec@192.168.1.111 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.11 Content-Length: 0 --- -- SIP/202-0000000f answered Local/202@from-queue-af93;2 -- Executing [s@macro-auto-blkvm:1] Set("SIP/202-0000000f", "__MACRO_RESULT=") in new stack -- Executing [s@macro-auto-blkvm:2] NoOp("SIP/202-0000000f", "Deleting: BLKVM/299/SIP/201-0000000e TRUE") in new stack -- Local/202@from-queue-af93;1 answered SIP/201-0000000e Scheduling destruction of SIP dialog '3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 192.168.1.213:5060: CANCEL sip:203@192.168.1.213:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5e0e1565;rport Max-Forwards: 70 From: "201" ;tag=as2330a1d9 To: Call-ID: 3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111 CSeq: 102 CANCEL User-Agent: Asterisk PBX 1.6.2.11 Content-Length: 0 --- Scheduling destruction of SIP dialog '3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111' in 6400 ms (Method: INVITE) == Spawn extension (macro-dial, s, 7) exited non-zero on 'Local/203@from-queue-9f48;2' in macro 'dial' == Spawn extension (macro-exten-vm, s, 11) exited non-zero on 'Local/203@from-queue-9f48;2' in macro 'exten-vm' == Spawn extension (from-internal, 203, 1) exited non-zero on 'Local/203@from-queue-9f48;2' -- Executing [h@from-internal:1] Macro("Local/203@from-queue-9f48;2", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("Local/203@from-queue-9f48;2", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf("Local/203@from-queue-9f48;2", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf("Local/203@from-queue-9f48;2", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup("Local/203@from-queue-9f48;2", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'Local/203@from-queue-9f48;2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'Local/203@from-queue-9f48;2' -- Stopped music on hold on SIP/201-0000000e <--- SIP read from UDP:192.168.1.213:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5e0e1565;rport From: "201" ;tag=as2330a1d9 To: ;tag=e516915c3d248474 Call-ID: 3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111 CSeq: 102 CANCEL User-Agent: Grandstream BT200 1.2.2.19 Supported: replaces, timer Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.1.213:5060 ---> SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5e0e1565;rport From: "201" ;tag=as2330a1d9 To: ;tag=e516915c3d248474 Call-ID: 3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111 CSeq: 102 INVITE User-Agent: Grandstream BT200 1.2.2.19 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 192.168.1.213:5060: ACK sip:203@192.168.1.213:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5e0e1565;rport Max-Forwards: 70 From: "201" ;tag=as2330a1d9 To: ;tag=e516915c3d248474 Contact: Call-ID: 3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.11 Content-Length: 0 --- Really destroying SIP dialog '3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111' Method: INVITE Reliably Transmitting (NAT) to 192.168.1.213:5060: OPTIONS sip:203@192.168.1.213:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK77822428;rport Max-Forwards: 70 From: "Unknown" ;tag=as54fb82c2 To: Contact: Call-ID: 52f325c63a89535f425a461a45d8908a@192.168.1.111 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.11 Date: Thu, 19 Aug 2010 12:47:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.1.213:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK77822428;rport From: "Unknown" ;tag=as54fb82c2 To: ;tag=ac178e1d691de328 Call-ID: 52f325c63a89535f425a461a45d8908a@192.168.1.111 CSeq: 102 OPTIONS User-Agent: Grandstream BT200 1.2.2.19 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '52f325c63a89535f425a461a45d8908a@192.168.1.111' Method: OPTIONS <--- SIP read from UDP:192.168.1.52:5060 ---> BYE sip:201@192.168.1.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.52:5060;branch=z9hG4bK-jvjz0rd9x021;rport From: ;tag=nvbtn81cu6 To: "201" ;tag=as0ec9c5cf Call-ID: 429f11b251cb8f722c27bbfd215259ec@192.168.1.111 CSeq: 1 BYE Max-Forwards: 70 Contact: ;reg-id=1 User-Agent: snom300/7.3.30 RTP-RxStat: Total_Rx_Pkts=68,Rx_Pkts=67,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=67,Tx_Pkts=67,Remote_Tx_Pkts=0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.1.52 : 5060 (NAT) <--- Transmitting (NAT) to 192.168.1.52:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.52:5060;branch=z9hG4bK-jvjz0rd9x021;received=192.168.1.52;rport=5060 From: ;tag=nvbtn81cu6 To: "201" ;tag=as0ec9c5cf Call-ID: 429f11b251cb8f722c27bbfd215259ec@192.168.1.111 CSeq: 1 BYE Server: Asterisk PBX 1.6.2.11 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> -- Executing [h@macro-dial:1] Macro("Local/202@from-queue-af93;2", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("Local/202@from-queue-af93;2", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf("Local/202@from-queue-af93;2", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf("Local/202@from-queue-af93;2", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup("Local/202@from-queue-af93;2", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'Local/202@from-queue-af93;2' in macro 'hangupcall' == Spawn extension (macro-dial, h, 1) exited non-zero on 'Local/202@from-queue-af93;2' == Spawn extension (macro-dial, s, 7) exited non-zero on 'Local/202@from-queue-af93;2' in macro 'dial' == Spawn extension (macro-exten-vm, s, 11) exited non-zero on 'Local/202@from-queue-af93;2' in macro 'exten-vm' == Spawn extension (from-internal, 202, 1) exited non-zero on 'Local/202@from-queue-af93;2' -- Executing [h@from-internal:1] Macro("Local/202@from-queue-af93;2", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("Local/202@from-queue-af93;2", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf("Local/202@from-queue-af93;2", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf("Local/202@from-queue-af93;2", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup("Local/202@from-queue-af93;2", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'Local/202@from-queue-af93;2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'Local/202@from-queue-af93;2' -- Executing [h@from-internal:1] Macro("SIP/201-0000000e", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/201-0000000e", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf("SIP/201-0000000e", "0?skipblkvm") in new stack -- Executing [s@macro-hangupcall:5] NoOp("SIP/201-0000000e", "Cleaning Up Block VM Flag: BLKVM/299/SIP/201-0000000e") in new stack -- Executing [s@macro-hangupcall:6] NoOp("SIP/201-0000000e", "Deleting: BLKVM/299/SIP/201-0000000e ") in new stack -- Executing [s@macro-hangupcall:7] GotoIf("SIP/201-0000000e", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup("SIP/201-0000000e", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/201-0000000e' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/201-0000000e' == Spawn extension (from-internal, 299, 9) exited non-zero on 'SIP/201-0000000e' -- Executing [h@from-internal:1] Macro("SIP/201-0000000e", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/201-0000000e", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf("SIP/201-0000000e", "0?skipblkvm") in new stack -- Executing [s@macro-hangupcall:5] NoOp("SIP/201-0000000e", "Cleaning Up Block VM Flag: BLKVM/299/SIP/201-0000000e") in new stack -- Executing [s@macro-hangupcall:6] NoOp("SIP/201-0000000e", "Deleting: BLKVM/299/SIP/201-0000000e ") in new stack -- Executing [s@macro-hangupcall:7] GotoIf("SIP/201-0000000e", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup("SIP/201-0000000e", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/201-0000000e' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/201-0000000e' Scheduling destruction of SIP dialog 'MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM.' in 6656 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.5, port 12592 Reliably Transmitting (NAT) to 192.168.1.5:12592: BYE sip:201@192.168.1.5:12592 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5fd6ef35;rport Max-Forwards: 70 From: "299";tag=as303ba4df To: ;tag=11788805 Call-ID: MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM. CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.2.11 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:192.168.1.5:12592 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5fd6ef35;rport=5060 Contact: To: ;tag=11788805 From: "299";tag=as303ba4df Call-ID: MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM. CSeq: 102 BYE User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '429f11b251cb8f722c27bbfd215259ec@192.168.1.111' Method: BYE Really destroying SIP dialog 'MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM.' Method: ACK