pbx1*CLI> == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found unknown media description format X-NSE for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.1.7:18930 Looking for 2155555757 in inbound (domain 192.168.1.10) list_route: hop: --- Transmitting (no NAT) to 192.168.1.7:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK191EB4;received=192.168.1.7 From: ;tag=47DFBD-E9 To: Call-ID: 718DAAAF-E76911E1-803FE0FD-8B13B9A1@192.168.1.7 CSeq: 101 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [2155555757@inbound:1] Set("SIP/vgw1-00000012", "DNIS=2155555757") in new stack -- Executing [2155555757@inbound:2] Set("SIP/vgw1-00000012", "CALLERID(num)=9910") in new stack -- Executing [2155555757@inbound:3] GotoIf("SIP/vgw1-00000012", "0?override") in new stack -- Executing [2155555757@inbound:4] Set("SIP/vgw1-00000012", "CALLERID(name)=") in new stack -- Executing [2155555757@inbound:5] Goto("SIP/vgw1-00000012", "start") in new stack -- Goto (inbound,2155555757,7) -- Executing [2155555757@inbound:7] Goto("SIP/vgw1-00000012", "attendant,s,1") in new stack -- Goto (attendant,s,1) -- Executing [s@attendant:1] Answer("SIP/vgw1-00000012", "") in new stack Audio is at 28136 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.1.7:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK191EB4;received=192.168.1.7 From: ;tag=47DFBD-E9 To: ;tag=as6a531390 Call-ID: 718DAAAF-E76911E1-803FE0FD-8B13B9A1@192.168.1.7 CSeq: 101 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Type: application/sdp Content-Length: 233 v=0 o=root 49062644 49062644 IN IP4 192.168.1.10 s=Asterisk PBX 1.8.15.0 c=IN IP4 192.168.1.10 t=0 0 m=audio 28136 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> -- Executing [s@attendant:2] Set("SIP/vgw1-00000012", "TIMEOUT(absolute)=14400") in new stack Channel will hangup at 2012-08-17 06:13:08.045 EDT. -- Executing [s@attendant:3] GotoIf("SIP/vgw1-00000012", "0?blacklist,1") in new stack -- Executing [s@attendant:4] Playback("SIP/vgw1-00000012", "local/silence") in new stack -- Playing 'local/silence.gsm' (language 'en') -- Executing [s@attendant:5] GotoIf("SIP/vgw1-00000012", "0?open,1") in new stack -- Executing [s@attendant:6] Set("SIP/vgw1-00000012", "HOLIDAY=") in new stack -- Executing [s@attendant:7] GotoIf("SIP/vgw1-00000012", "0?holiday,1") in new stack -- Executing [s@attendant:8] GotoIfTime("SIP/vgw1-00000012", "09:00-17:30,mon-wed,*,*?open,1") in new stack -- Executing [s@attendant:9] GotoIfTime("SIP/vgw1-00000012", "09:00-20:00,thu-fri,*,*?open,1") in new stack -- Executing [s@attendant:10] GotoIfTime("SIP/vgw1-00000012", "10:00-16:00,sat,*,*?open,1") in new stack -- Executing [s@attendant:11] Set("SIP/vgw1-00000012", "MSG=open") in new stack -- Executing [s@attendant:12] Goto("SIP/vgw1-00000012", "main,1") in new stack -- Goto (attendant,main,1) -- Executing [main@attendant:1] BackGround("SIP/vgw1-00000012", "local/open") in new stack -- Playing 'local/open.ulaw' (language 'en') [Aug 17 02:13:10] DTMF[27804]: channel.c:4090 __ast_read: DTMF begin '1' received on SIP/vgw1-00000012 [Aug 17 02:13:10] DTMF[27804]: channel.c:4094 __ast_read: DTMF begin ignored '1' on SIP/vgw1-00000012 [Aug 17 02:13:10] DTMF[27804]: channel.c:4005 __ast_read: DTMF end '1' received on SIP/vgw1-00000012, duration 58 ms [Aug 17 02:13:10] DTMF[27804]: channel.c:4074 __ast_read: DTMF end passthrough '1' on SIP/vgw1-00000012 == CDR updated on SIP/vgw1-00000012 -- Executing [1@attendant:1] GotoIf("SIP/vgw1-00000012", "0?vm") in new stack -- Executing [1@attendant:2] Dial("SIP/vgw1-00000012", "SIP/111&SIP/112&SIP/113&SIP/114,11,mk") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/111 == Using SIP RTP CoS mark 5 -- Called SIP/112 == Using SIP RTP CoS mark 5 -- Called SIP/113 == Using SIP RTP CoS mark 5 -- Called SIP/114 -- Started music on hold, class 'default', on SIP/vgw1-00000012 -- SIP/111-00000013 connected line has changed. Saving it until answer for SIP/vgw1-00000012 -- SIP/112-00000014 connected line has changed. Saving it until answer for SIP/vgw1-00000012 -- SIP/113-00000015 connected line has changed. Saving it until answer for SIP/vgw1-00000012 -- SIP/114-00000016 connected line has changed. Saving it until answer for SIP/vgw1-00000012 -- SIP/112-00000014 is ringing -- SIP/111-00000013 is ringing -- SIP/114-00000016 is ringing -- SIP/113-00000015 is ringing pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> == Using SIP RTP CoS mark 5 [Aug 17 02:13:24] NOTICE[27808]: features.c:7270 ast_pickup_call: pickup SIP/112-00000014 attempt by SIP/139-00000017 -- SIP/139-00000017 answered SIP/vgw1-00000012 -- Stopped music on hold on SIP/vgw1-00000012 -- Executing [h@attendant:1] NoOp("SIP/vgw1-00000012", """ <9910> hung up") in new stack == Spawn extension (attendant, 1, 2) exited non-zero on 'SIP/vgw1-00000012' Scheduling destruction of SIP dialog '718DAAAF-E76911E1-803FE0FD-8B13B9A1@192.168.1.7' in 6400 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.7:5060 Reliably Transmitting (no NAT) to 192.168.1.7:5060: BYE sip:9910@192.168.1.7:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK2c020d08 Max-Forwards: 70 From: ;tag=as6a531390 To: ;tag=47DFBD-E9 Call-ID: 718DAAAF-E76911E1-803FE0FD-8B13B9A1@192.168.1.7 CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.15.0 X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 --- <--- 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=z9hG4bK2c020d08 From: ;tag=as6a531390 To: ;tag=47DFBD-E9 Date: Fri, 17 Aug 2012 06:13:25 GMT Call-ID: 718DAAAF-E76911E1-803FE0FD-8B13B9A1@192.168.1.7 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 102 BYE <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '718DAAAF-E76911E1-803FE0FD-8B13B9A1@192.168.1.7' Method: ACK pbx1*CLI>