-- SIP/1001-085dd138 is making progress passing it to SIP/3024-085d91d0 -- Executing [7775@incoming:1] Goto("SIP/1005-085c1578", "answer|1") in new stack -- Goto (incoming,answer,1) -- Executing [answer@incoming:1] Answer("SIP/1005-085c1578", "") in new stack -- Executing [answer@incoming:2] Wait("SIP/1005-085c1578", "1") in new stack -- Executing [answer@incoming:3] NoOp("SIP/1005-085c1578", "1005") in new stack -- Executing [answer@incoming:4] NoOp("SIP/1005-085c1578", "") in new stack -- Executing [answer@incoming:5] NoOp("SIP/1005-085c1578", "") in new stack -- Executing [answer@incoming:6] Set("SIP/1005-085c1578", "TIMEOUT(digit)=7") in new stack -- Digit timeout set to 7 -- Executing [answer@incoming:7] BackGround("SIP/1005-085c1578", "11") in new stack -- Playing '11' (language 'ru') -- SIP/1001-085dd138 answered SIP/3024-085d91d0 -- Executing [answer@incoming:8] SetMusicOnHold("SIP/1005-085c1578", "default") in new stack -- Executing [answer@incoming:9] WaitExten("SIP/1005-085c1578", "7") in new stack asterisk*CLI> -- Timeout on SIP/1005-085c1578, going to 't' -- Executing [t@incoming:1] Goto("SIP/1005-085c1578", "in_call|s|1") in new stack -- Goto (in_call,s,1) -- Executing [s@in_call:1] Dial("SIP/1005-085c1578", "SIP/3000|15|tr") in new stack Audio is at 192.168.4.2 port 14634 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.0.150:5060: INVITE sip:3000@192.168.0.150:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK4deb49e9;rport From: "1005" ;tag=as2c5c71d7 To: Contact: Call-ID: 10b26a620096b56756fa788339a7a150@192.168.4.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 22 Jun 2007 10:21:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 233 v=0 o=root 2666 2666 IN IP4 192.168.4.2 s=session c=IN IP4 192.168.4.2 t=0 0 m=audio 14634 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 3000 asterisk*CLI> <--- SIP read from 192.168.0.150:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK4deb49e9;rport From: "1005" ;tag=as2c5c71d7 To: Call-ID: 10b26a620096b56756fa788339a7a150@192.168.4.2 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.0.16 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk*CLI> <--- SIP read from 192.168.0.150:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK4deb49e9;rport From: "1005" ;tag=as2c5c71d7 To: ;tag=6c573764c0b95fc9 Call-ID: 10b26a620096b56756fa788339a7a150@192.168.4.2 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.0.16 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/3000-085d7c50 is ringing asterisk*CLI> <--- SIP read from 192.168.0.150:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK4deb49e9;rport From: "1005" ;tag=as2c5c71d7 To: ;tag=6c573764c0b95fc9 Call-ID: 10b26a620096b56756fa788339a7a150@192.168.4.2 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.0.16 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Supported: replaces, timer Content-Length: 210 v=0 o=3000 8000 8000 IN IP4 192.168.0.150 s=SIP Call c=IN IP4 192.168.0.150 t=0 0 m=audio 5004 RTP/AVP 0 96 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-11 <-------------> --- (12 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 96 Peer audio RTP is at port 192.168.0.150:5004 Found description format PCMU for ID 0 Found description format telephone-event for ID 96 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=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 192.168.0.150:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.150, port 5060 Transmitting (no NAT) to 192.168.0.150:5060: ACK sip:3000@192.168.0.150:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK33e41028;rport From: "1005" ;tag=as2c5c71d7 To: ;tag=6c573764c0b95fc9 Contact: Call-ID: 10b26a620096b56756fa788339a7a150@192.168.4.2 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/3000-085d7c50 answered SIP/1005-085c1578 -- SIP/3007-085f2c18 answered SIP/3012-085c54e0 == Spawn extension (macro-office_dial, s, 1) exited non-zero on 'SIP/3012-085c54e0' in macro 'office_dial' == Spawn extension (macro-office_dial, s, 1) exited non-zero on 'SIP/3012-085c54e0' -- Executing [3007@default:1] Macro("SIP/3012-085bfff8", "office_dial|3007|20") in new stack -- Executing [s@macro-office_dial:1] Dial("SIP/3012-085bfff8", "SIP/3007|20|tr") in new stack -- Called 3007 -- SIP/3007-085e6490 is ringing -- Started music on hold, class 'default', on SIP/1005-085c1578 -- Playing 'pbx-transfer' (language 'ru') -- Executing [600@default:1] Playback("Local/600@default-fd27,2", "demo-echotest") in new stack -- Playing 'demo-echotest' (language 'en') [Jun 22 13:21:28] WARNING[3478]: cdr.c:767 ast_cdr_init: CDR already initialized on 'Local/600@default-fd27,1' -- Executing [980296922041@default:1] Macro("SIP/3007-085f50e8", "velcom_dial|6922041") in new stack -- Executing [s@macro-velcom_dial:1] Dial("SIP/3007-085f50e8", "Sip/1001/80296922041|60|t") in new stack -- Called 1001/80296922041 asterisk*CLI> quit <--- SIP read from 192.168.0.150:5060 ---> BYE sip:1005@192.168.4.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bKcc9952e5e5ae3857 From: ;tag=6c573764c0b95fc9 To: "1005" ;tag=as2c5c71d7 Call-ID: 10b26a620096b56756fa788339a7a150@192.168.4.2 CSeq: 22251 BYE User-Agent: Grandstream GXP2000 1.1.0.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 192.168.0.150 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.0.150:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bKcc9952e5e5ae3857;received=192.168.0.150 From: ;tag=6c573764c0b95fc9 To: "1005" ;tag=as2c5c71d7 Call-ID: 10b26a620096b56756fa788339a7a150@192.168.4.2 CSeq: 22251 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Stopped music on hold on SIP/1005-085c1578 -- Playing 'beep' (language 'en') == Spawn extension (in_call, s, 1) exited non-zero on 'Transfered/SIP/1005-085c1578' Really destroying SIP dialog '10b26a620096b56756fa788339a7a150@192.168.4.2' Method: BYE -- SIP/1001-085f2c18 is making progress passing it to SIP/3007-085f50e8 asterisk*CLI> quit [root@asterisk ~]#