[May 23 18:42:07] DEBUG[2655] chan_sip.c: Header 0: (0) [May 23 18:42:07] DEBUG[2655] chan_sip.c: Line: (0) [May 23 18:42:08] VERBOSE[2655] logger.c: <--- SIP read from 192.168.10.10:5061 ---> INVITE sip:6177@192.168.130.116:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.10:5061;branch=z9hG4bK33ce679c From: "3002" ;tag=17327731 To: Date: Fri, 23 May 2008 17:41:52 GMT Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 Supported: timer Min-SE: 1800 User-Agent: Cisco-CCM4.1 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK CSeq: 101 INVITE Max-Forwards: 70 Remote-Party-ID: "3002" ;party=calling;screen=no;privacy =off Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 229 v=0 o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 192.168.10.10 s=SIP Call c=IN IP4 192.168.10.10 t=0 0 m=audio 29560 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 0: INVITE sip:6177@192.168.130. 116:5060 SIP/2.0 (44) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10 .10:5061;branch=z9hG4bK33ce679c (59) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 2: From: "3002" ;tag=17327731 (50) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 3: To: (30) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 4: Date: Fri, 23 May 2008 17:41 :52 GMT (35) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 5: Call-ID: 85e37580-1de146f2-8 4971-a0aa8c0@192.168.10.10 (54) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 6: Supported: timer (16) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 7: Min-SE: 1800 (13) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 8: User-Agent: Cisco-CCM4.1 (24 ) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 9: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK (47) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 10: CSeq: 101 INVITE (16) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 11: Max-Forwards: 70 (16) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 12: Remote-Party-ID: "3002" ;party=calling;screen=no;privacy=off (84) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 13: Contact: (38) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 14: Expires: 180 (12) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 15: Allow-Events: telephone-eve nt (29) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 16: Content-Type: application/s dp (29) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 17: Content-Length: 229 (19) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 18: (0) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Line: v=0 (3) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Line: o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 192.168.10.10 (52) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Line: s=SIP Call (10) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Line: c=IN IP4 192.168.10.10 (22) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Line: t=0 0 (5) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Line: m=audio 29560 RTP/AVP 0 101 (27) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Line: a=sendrecv (10) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Line: a=ptime:20 (10) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Line: a=rtpmap:101 telephone-event/800 0 (33) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Line: a=fmtp:101 0-15 (15) [May 23 18:42:08] VERBOSE[2655] logger.c: --- (18 headers 11 lines) --- [May 23 18:42:08] DEBUG[2655] chan_sip.c: = No match Their Call ID: Nzc4YWQyNTZl YjkxZTUxMmJlNmZjYzlkOTAzNTMwNDE. Their Tag 3a114229 Our tag: as4c1a4ff5 [May 23 18:42:08] DEBUG[2655] chan_sip.c: = No match Their Call ID: Mzk5NWI0OTcz NGE5M2NkMDBjOTMzMDU2ZjAxOWNjMTE. Their Tag 9c18b943 Our tag: as1f285696 [May 23 18:42:08] DEBUG[2655] chan_sip.c: = No match Their Call ID: OTUwNzUwMDMw N2Y0ZjJiNWM1NzE1ZGJlYzY0ODEzM2M. Their Tag 24784362 Our tag: as1248e371 [May 23 18:42:08] DEBUG[2655] chan_sip.c: = No match Their Call ID: YmYxNzUxNjJm NGMwODY1ODk5YmNiNmJmZGE5NzkzODQ. Their Tag d93a1a48 Our tag: as48a89b8f [May 23 18:42:08] DEBUG[2655] chan_sip.c: Setting NAT on RTP to Off [May 23 18:42:08] DEBUG[2655] chan_sip.c: Allocating new SIP dialog for 85e37580 -1de146f2-84971-a0aa8c0@192.168.10.10 - INVITE (With RTP) [May 23 18:42:08] DEBUG[2655] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [May 23 18:42:08] DEBUG[2655] chan_sip.c: Begin: parsing SIP "Supported: timer" [May 23 18:42:08] DEBUG[2655] chan_sip.c: Found SIP option: -timer- [May 23 18:42:08] DEBUG[2655] chan_sip.c: Matched SIP option: timer [May 23 18:42:08] VERBOSE[2655] logger.c: Sending to 192.168.10.10 : 5061 (no NA T) [May 23 18:42:08] VERBOSE[2655] logger.c: Using INVITE request as basis request - 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 [May 23 18:42:08] VERBOSE[2655] logger.c: Found no matching peer or user for '19 2.168.10.10:5061' [May 23 18:42:08] VERBOSE[2655] logger.c: Found RTP audio format 0 [May 23 18:42:08] VERBOSE[2655] logger.c: Found RTP audio format 101 [May 23 18:42:08] VERBOSE[2655] logger.c: Peer audio RTP is at port 192.168.10.1 0:29560 [May 23 18:42:08] VERBOSE[2655] logger.c: Found audio description format PCMU fo r ID 0 [May 23 18:42:08] VERBOSE[2655] logger.c: Found audio description format telepho ne-event for ID 101 [May 23 18:42:08] VERBOSE[2655] logger.c: Got unsupported a:fmtp in SDP offer [May 23 18:42:08] DEBUG[2655] chan_sip.c: T38 state changed to 0 on channel [May 23 18:42:08] VERBOSE[2655] logger.c: Capabilities: us - 0x8000e (gsm|ulaw|a law|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [May 23 18:42:08] VERBOSE[2655] logger.c: Non-codec capabilities (dtmf): us - 0x 1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-eve nt) [May 23 18:42:08] VERBOSE[2655] logger.c: Peer audio RTP is at port 192.168.10.1 0:29560 [May 23 18:42:08] DEBUG[2655] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Checking SIP call limits for device [May 23 18:42:08] DEBUG[2655] chan_sip.c: Updating call counter for incoming cal l [May 23 18:42:08] VERBOSE[2655] logger.c: Looking for 6177 in default (domain 19 2.168.130.116) [May 23 18:42:08] DEBUG[2655] chan_sip.c: *** Our native formats are 0x4 (ulaw) [May 23 18:42:08] DEBUG[2655] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [May 23 18:42:08] DEBUG[2655] chan_sip.c: *** Our capabilities are 0x8000e (gsm| ulaw|alaw|h263) [May 23 18:42:08] DEBUG[2655] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 ( ulaw) [May 23 18:42:08] DEBUG[2655] chan_sip.c: This channel will not be able to handl e video. [May 23 18:42:08] DEBUG[2655] chan_sip.c: build_route: Contact hop: [May 23 18:42:08] VERBOSE[2655] logger.c: list_route: hop: [May 23 18:42:08] DEBUG[2655] chan_sip.c: SIP/192.168.10.10-091cda88: New call i s still down.... Trying... [May 23 18:42:08] VERBOSE[2655] logger.c: <--- Transmitting (no NAT) to 192.168.10.10:5061 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.10:5061;branch=z9hG4bK33ce679c;received=192.168.10. 10 From: "3002" ;tag=17327731 To: Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10 .10:5061;branch=z9hG4bK33ce679c;received=192.168.10.10 (82) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 2: From: "3002" ;tag=17327731 (50) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 3: To: (30) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 4: Call-ID: 85e37580-1de146f2-8 4971-a0aa8c0@192.168.10.10 (54) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 5: CSeq: 101 INVITE (16) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24 ) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 8: Supported: replaces (19) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 9: Contact: (35) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 10: Content-Length: 0 (17) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 11: (0) [May 23 18:42:08] DEBUG[2591] chan_sip.c: Checking device state for peer 192.168 .10.10-091cda88 [May 23 18:42:08] VERBOSE[15260] logger.c: -- Executing [6177@default:1] Ans wer("SIP/192.168.10.10-091cda88", "") in new stack [May 23 18:42:08] DEBUG[15260] chan_sip.c: SIP answering channel: SIP/192.168.10 .10-091cda88 [May 23 18:42:08] DEBUG[15260] chan_sip.c: Setting framing from config on incomi ng call [May 23 18:42:08] DEBUG[15260] chan_sip.c: ** Our capability: 0x4 (ulaw) Video f lag: True [May 23 18:42:08] DEBUG[15260] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [May 23 18:42:08] VERBOSE[15260] logger.c: Audio is at 192.168.130.116 port 1922 2 [May 23 18:42:08] VERBOSE[15260] logger.c: Adding codec 0x4 (ulaw) to SDP [May 23 18:42:08] VERBOSE[15260] logger.c: Adding non-codec 0x1 (telephone-event ) to SDP [May 23 18:42:08] DEBUG[15260] chan_sip.c: -- Done with adding codecs to SDP [May 23 18:42:08] DEBUG[15260] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 23 18:42:08] VERBOSE[15260] logger.c: <--- Reliably Transmitting (no NAT) to 192.168.10.10:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.10:5061;branch=z9hG4bK33ce679c;received=192.168.10. 10 From: "3002" ;tag=17327731 To: ;tag=as7e8cf3ad Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 217 v=0 o=root 2552 2552 IN IP4 192.168.130.116 s=session c=IN IP4 192.168.130.116 t=0 0 m=audio 19222 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 <------------> [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1 0.10:5061;branch=z9hG4bK33ce679c;received=192.168.10.10 (82) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 2: From: "3002" ;tag=17327731 (50) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 3: To: ;tag=as7e8cf3ad (45) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 4: Call-ID: 85e37580-1de146f2- 84971-a0aa8c0@192.168.10.10 (54) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 5: CSeq: 101 INVITE (16) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 6: User-Agent: Asterisk PBX (2 4) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 8: Supported: replaces (19) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 9: Contact: (35) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 10: Content-Type: application/ sdp (29) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 11: Content-Length: 217 (19) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 12: (0) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Line: v=0 (3) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Line: o=root 2552 2552 IN IP4 192.168 .130.116 (39) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Line: s=session (9) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Line: c=IN IP4 192.168.130.116 (24) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Line: t=0 0 (5) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Line: m=audio 19222 RTP/AVP 0 101 (27 ) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Line: a=rtpmap:101 telephone-event/80 00 (33) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Line: a=fmtp:101 0-16 (15) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Line: a=ptime:20 (10) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Line: a=sendrecv (10) [May 23 18:42:08] DEBUG[15260] chan_sip.c: *** SIP TIMER: Initializing retransmi t timer on packet: Id #-1 [May 23 18:42:08] VERBOSE[15260] logger.c: -- Executing [6177@default:2] Got o("SIP/192.168.10.10-091cda88", "6176|1") in new stack [May 23 18:42:08] VERBOSE[15260] logger.c: -- Goto (default,6176,1) [May 23 18:42:08] DEBUG[2591] chan_sip.c: Checking device state for peer 192.168 .10.10 [May 23 18:42:08] DEBUG[2591] chan_sip.c: Checking device state for peer 192.168 .10.10-091cda88 [May 23 18:42:08] VERBOSE[2655] logger.c: <--- SIP read from 192.168.10.10:5061 ---> ACK sip:6177@192.168.130.116:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.10:5061;branch=z9hG4bKd6a1ad4 From: "3002" ;tag=17327731 To: ;tag=as7e8cf3ad Date: Fri, 23 May 2008 17:41:52 GMT Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 <-------------> [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 0: ACK sip:6177@192.168.130.116 :5060 SIP/2.0 (41) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10 .10:5061;branch=z9hG4bKd6a1ad4 (58) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 2: From: "3002" ;tag=17327731 (50) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 3: To: ;tag=as7e8cf3ad (45) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 4: Date: Fri, 23 May 2008 17:41 :52 GMT (35) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 5: Call-ID: 85e37580-1de146f2-8 4971-a0aa8c0@192.168.10.10 (54) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 6: Max-Forwards: 70 (16) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 7: CSeq: 101 ACK (13) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 8: Content-Length: 0 (17) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 9: (0) [May 23 18:42:08] VERBOSE[2655] logger.c: --- (9 headers 0 lines) --- [May 23 18:42:08] DEBUG[2655] chan_sip.c: = Found Their Call ID: 85e37580-1de146 f2-84971-a0aa8c0@192.168.10.10 Their Tag 17327731 Our tag: as7e8cf3ad [May 23 18:42:08] DEBUG[2655] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [May 23 18:42:08] DEBUG[2655] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1653 [May 23 18:42:08] DEBUG[2655] chan_sip.c: Stopping retransmission on '85e37580-1 de146f2-84971-a0aa8c0@192.168.10.10' of Response 101: Match Found [May 23 18:42:08] VERBOSE[15260] logger.c: -- Executing [6176@default:1] Tra nsfer("SIP/192.168.10.10-091cda88", "SIP/6903@192.168.10.10") in new stack [May 23 18:42:08] DEBUG[15260] chan_sip.c: SIP transfer of 85e37580-1de146f2-849 71-a0aa8c0@192.168.10.10 to 6903@192.168.10.10 [May 23 18:42:08] DEBUG[2591] chan_sip.c: Checking device state for peer 192.168 .10.10 [May 23 18:42:08] DEBUG[15260] chan_sip.c: Strict routing enforced for session 8 5e37580-1de146f2-84971-a0aa8c0@192.168.10.10 [May 23 18:42:08] VERBOSE[15260] logger.c: set_destination: Parsing for address/port to send to [May 23 18:42:08] VERBOSE[15260] logger.c: set_destination: set destination to 1 92.168.10.10, port 5061 [May 23 18:42:08] VERBOSE[15260] logger.c: Reliably Transmitting (no NAT) to 192 .168.10.10:5061: REFER sip:3002@192.168.10.10:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.116:5060;branch=z9hG4bK2f17865b;rport From: ;tag=as7e8cf3ad To: "3002" ;tag=17327731 Contact: Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 CSeq: 102 REFER User-Agent: Asterisk PBX Max-Forwards: 70 Refer-To: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Referred-By: --- [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 0: REFER sip:3002@192.168.10.1 0:5061 SIP/2.0 (41) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.13 0.116:5060;branch=z9hG4bK2f17865b;rport (66) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 2: From: ;tag=as7e8cf3ad (47) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 3: To: "3002" ;tag=17327731 (48) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 4: Contact: (35) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 5: Call-ID: 85e37580-1de146f2- 84971-a0aa8c0@192.168.10.10 (54) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 6: CSeq: 102 REFER (15) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 7: User-Agent: Asterisk PBX (2 4) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 8: Max-Forwards: 70 (16) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 9: Refer-To: (34) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL , OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 11: Supported: replaces (19) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 12: Referred-By: (39) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 13: (0) [May 23 18:42:08] DEBUG[15260] chan_sip.c: *** SIP TIMER: Initializing retransmi t timer on packet: Id #-1 [May 23 18:42:08] VERBOSE[15260] logger.c: -- Executing [6176@default:2] NoO p("SIP/192.168.10.10-091cda88", "SUCCESS") in new stack [May 23 18:42:08] VERBOSE[15260] logger.c: == Auto fallthrough, channel 'SIP/1 92.168.10.10-091cda88' status is 'UNKNOWN' [May 23 18:42:08] DEBUG[15260] chan_sip.c: Hangup call SIP/192.168.10.10-091cda8 8, SIP callid 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10) [May 23 18:42:08] VERBOSE[15260] logger.c: Scheduling destruction of SIP dialog '85e37580-1de146f2-84971-a0aa8c0@192.168.10.10' in 32000 ms (Method: ACK) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Strict routing enforced for session 8 5e37580-1de146f2-84971-a0aa8c0@192.168.10.10 [May 23 18:42:08] VERBOSE[15260] logger.c: set_destination: Parsing for address/port to send to [May 23 18:42:08] VERBOSE[15260] logger.c: set_destination: set destination to 1 92.168.10.10, port 5061 [May 23 18:42:08] VERBOSE[15260] logger.c: Reliably Transmitting (no NAT) to 192 .168.10.10:5061: BYE sip:3002@192.168.10.10:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.116:5060;branch=z9hG4bK25d28e2d;rport From: ;tag=as7e8cf3ad To: "3002" ;tag=17327731 Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 0: BYE sip:3002@192.168.10.10: 5061 SIP/2.0 (39) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.13 0.116:5060;branch=z9hG4bK25d28e2d;rport (66) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 2: From: ;tag=as7e8cf3ad (47) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 3: To: "3002" ;tag=17327731 (48) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 4: Call-ID: 85e37580-1de146f2- 84971-a0aa8c0@192.168.10.10 (54) [May 23 18:42:08] VERBOSE[2655] logger.c: <--- SIP read from 192.168.10.10:5060 ---> SIP/2.0 405 Method Not Allowed Via: SIP/2.0/UDP 192.168.130.116:5060;branch=z9hG4bK2f17865b;rport From: ;tag=as7e8cf3ad To: "3002" ;tag=17327731 Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 CSeq: 102 REFER Content-Length: 0 <-------------> [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 0: SIP/2.0 405 Method Not Allow ed (30) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.130 .116:5060;branch=z9hG4bK2f17865b;rport (66) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 2: From: ;tag=as7e8cf3ad (47) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 3: To: "3002" ;tag=17327731 (48) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 4: Call-ID: 85e37580-1de146f2-8 4971-a0aa8c0@192.168.10.10 (54) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 5: CSeq: 102 REFER (15) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 6: Content-Length: 0 (17) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 7: (0) [May 23 18:42:08] VERBOSE[2655] logger.c: --- (7 headers 0 lines) --- [May 23 18:42:08] DEBUG[2655] chan_sip.c: = Found Their Call ID: 85e37580-1de146 f2-84971-a0aa8c0@192.168.10.10 Their Tag 17327731 Our tag: as7e8cf3ad [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 5: CSeq: 103 BYE (13) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 6: User-Agent: Asterisk PBX (2 4) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 7: Max-Forwards: 70 (16) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 8: Content-Length: 0 (17) [May 23 18:42:08] DEBUG[15260] chan_sip.c: Header 9: (0) [May 23 18:42:08] DEBUG[15260] chan_sip.c: *** SIP TIMER: Initializing retransmi t timer on packet: Id #-1 [May 23 18:42:08] DEBUG[2591] chan_sip.c: Checking device state for peer 192.168 .10.10-091cda88 [May 23 18:42:08] DEBUG[2591] chan_sip.c: Checking device state for peer 192.168 .10.10 [May 23 18:42:08] VERBOSE[2655] logger.c: <--- SIP read from 192.168.10.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.130.116:5060;branch=z9hG4bK25d28e2d;rport From: ;tag=as7e8cf3ad To: "3002" ;tag=17327731 Date: Fri, 23 May 2008 17:41:52 GMT Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 Content-Length: 0 CSeq: 103 BYE <-------------> [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.130 .116:5060;branch=z9hG4bK25d28e2d;rport (66) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 2: From: ;tag=as7e8cf3ad (47) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 3: To: "3002" ;tag=17327731 (48) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 4: Date: Fri, 23 May 2008 17:41 :52 GMT (35) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 5: Call-ID: 85e37580-1de146f2-8 4971-a0aa8c0@192.168.10.10 (54) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 6: Content-Length: 0 (17) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 7: CSeq: 103 BYE (13) [May 23 18:42:08] DEBUG[2655] chan_sip.c: Header 8: (0) [May 23 18:42:08] VERBOSE[2655] logger.c: --- (8 headers 0 lines) --- [May 23 18:42:08] DEBUG[2655] chan_sip.c: = Found Their Call ID: 85e37580-1de146 f2-84971-a0aa8c0@192.168.10.10 Their Tag 17327731 Our tag: as7e8cf3ad [May 23 18:42:08] DEBUG[2655] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1656 [May 23 18:42:08] DEBUG[2655] chan_sip.c: Stopping retransmission on '85e37580-1 de146f2-84971-a0aa8c0@192.168.10.10' of Request 103: Match Found [May 23 18:42:08] VERBOSE[2655] logger.c: SIP Response message for INCOMING dial og BYE arrived [May 23 18:42:09] DEBUG[2655] chan_sip.c: SIP TIMER: Rescheduling retransmission #1654 (1) REFER - 9 [May 23 18:42:09] DEBUG[2655] chan_sip.c: ** SIP timers: Rescheduling retransmis sion 2 to 1000 ms (t1 500 ms (Retrans id #1654)) [May 23 18:42:09] VERBOSE[2655] logger.c: Retransmitting #1 (no NAT) to 192.168. 10.10:5061: REFER sip:3002@192.168.10.10:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.116:5060;branch=z9hG4bK2f17865b;rport From: ;tag=as7e8cf3ad To: "3002" ;tag=17327731 Contact: Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 CSeq: 102 REFER User-Agent: Asterisk PBX Max-Forwards: 70 Refer-To: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Referred-By: --- [May 23 18:42:09] VERBOSE[2655] logger.c: <--- SIP read from 192.168.10.10:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.130.116:5060;branch=z9hG4bK2f17865b;rport From: ;tag=as7e8cf3ad To: "3002" ;tag=17327731 Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 CSeq: 102 REFER Content-Length: 0 <-------------> [May 23 18:42:09] DEBUG[2655] chan_sip.c: Header 0: SIP/2.0 481 Call Leg/Transac tion Does Not Exist (47) [May 23 18:42:09] DEBUG[2655] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.130 .116:5060;branch=z9hG4bK2f17865b;rport (66) [May 23 18:42:09] DEBUG[2655] chan_sip.c: Header 2: From: ;tag=as7e8cf3ad (47) [May 23 18:42:09] DEBUG[2655] chan_sip.c: Header 3: To: "3002" ;tag=17327731 (48) [May 23 18:42:09] DEBUG[2655] chan_sip.c: Header 4: Call-ID: 85e37580-1de146f2-8 4971-a0aa8c0@192.168.10.10 (54) [May 23 18:42:09] DEBUG[2655] chan_sip.c: Header 5: CSeq: 102 REFER (15) [May 23 18:42:09] DEBUG[2655] chan_sip.c: Header 6: Content-Length: 0 (17) [May 23 18:42:09] DEBUG[2655] chan_sip.c: Header 7: (0) [May 23 18:42:09] VERBOSE[2655] logger.c: --- (7 headers 0 lines) --- [May 23 18:42:09] DEBUG[2655] chan_sip.c: = Found Their Call ID: 85e37580-1de146 f2-84971-a0aa8c0@192.168.10.10 Their Tag 17327731 Our tag: as7e8cf3ad [May 23 18:42:10] DEBUG[2655] chan_sip.c: SIP TIMER: Rescheduling retransmission #1654 (2) REFER - 9 [May 23 18:42:10] DEBUG[2655] chan_sip.c: ** SIP timers: Rescheduling retransmis sion 3 to 2000 ms (t1 500 ms (Retrans id #1654)) [May 23 18:42:10] VERBOSE[2655] logger.c: Retransmitting #2 (no NAT) to 192.168. 10.10:5061: REFER sip:3002@192.168.10.10:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.116:5060;branch=z9hG4bK2f17865b;rport From: ;tag=as7e8cf3ad To: "3002" ;tag=17327731 Contact: Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 CSeq: 102 REFER User-Agent: Asterisk PBX Max-Forwards: 70 Refer-To: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Referred-By: --- [May 23 18:42:10] VERBOSE[2655] logger.c: <--- SIP read from 192.168.10.10:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.130.116:5060;branch=z9hG4bK2f17865b;rport From: ;tag=as7e8cf3ad To: "3002" ;tag=17327731 Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 CSeq: 102 REFER Content-Length: 0 <-------------> [May 23 18:42:10] DEBUG[2655] chan_sip.c: Header 0: SIP/2.0 481 Call Leg/Transac tion Does Not Exist (47) [May 23 18:42:10] DEBUG[2655] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.130 .116:5060;branch=z9hG4bK2f17865b;rport (66) [May 23 18:42:10] DEBUG[2655] chan_sip.c: Header 2: From: ;tag=as7e8cf3ad (47) [May 23 18:42:10] DEBUG[2655] chan_sip.c: Header 3: To: "3002" ;tag=17327731 (48) [May 23 18:42:10] DEBUG[2655] chan_sip.c: Header 4: Call-ID: 85e37580-1de146f2-8 4971-a0aa8c0@192.168.10.10 (54) [May 23 18:42:10] DEBUG[2655] chan_sip.c: Header 5: CSeq: 102 REFER (15) [May 23 18:42:10] DEBUG[2655] chan_sip.c: Header 6: Content-Length: 0 (17) [May 23 18:42:10] DEBUG[2655] chan_sip.c: Header 7: (0) [May 23 18:42:10] VERBOSE[2655] logger.c: --- (7 headers 0 lines) --- [May 23 18:42:10] DEBUG[2655] chan_sip.c: = Found Their Call ID: 85e37580-1de146 f2-84971-a0aa8c0@192.168.10.10 Their Tag 17327731 Our tag: as7e8cf3ad [May 23 18:42:12] DEBUG[2655] chan_sip.c: SIP TIMER: Rescheduling retransmission #1654 (3) REFER - 9 [May 23 18:42:12] DEBUG[2655] chan_sip.c: ** SIP timers: Rescheduling retransmis sion 4 to 4000 ms (t1 500 ms (Retrans id #1654)) [May 23 18:42:12] VERBOSE[2655] logger.c: Retransmitting #3 (no NAT) to 192.168. 10.10:5061: REFER sip:3002@192.168.10.10:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.116:5060;branch=z9hG4bK2f17865b;rport From: ;tag=as7e8cf3ad To: "3002" ;tag=17327731 Contact: Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 CSeq: 102 REFER User-Agent: Asterisk PBX Max-Forwards: 70 Refer-To: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Referred-By: --- [May 23 18:42:12] VERBOSE[2655] logger.c: <--- SIP read from 192.168.10.10:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.130.116:5060;branch=z9hG4bK2f17865b;rport From: ;tag=as7e8cf3ad To: "3002" ;tag=17327731 Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 CSeq: 102 REFER Content-Length: 0 <-------------> [May 23 18:42:12] DEBUG[2655] chan_sip.c: Header 0: SIP/2.0 481 Call Leg/Transac tion Does Not Exist (47) [May 23 18:42:12] DEBUG[2655] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.130 .116:5060;branch=z9hG4bK2f17865b;rport (66) [May 23 18:42:12] DEBUG[2655] chan_sip.c: Header 2: From: ;tag=as7e8cf3ad (47) [May 23 18:42:12] DEBUG[2655] chan_sip.c: Header 3: To: "3002" ;tag=17327731 (48) [May 23 18:42:12] DEBUG[2655] chan_sip.c: Header 4: Call-ID: 85e37580-1de146f2-8 4971-a0aa8c0@192.168.10.10 (54) [May 23 18:42:12] DEBUG[2655] chan_sip.c: Header 5: CSeq: 102 REFER (15) [May 23 18:42:12] DEBUG[2655] chan_sip.c: Header 6: Content-Length: 0 (17) [May 23 18:42:12] DEBUG[2655] chan_sip.c: Header 7: (0) [May 23 18:42:12] VERBOSE[2655] logger.c: --- (7 headers 0 lines) --- [May 23 18:42:12] DEBUG[2655] chan_sip.c: = Found Their Call ID: 85e37580-1de146 f2-84971-a0aa8c0@192.168.10.10 Their Tag 17327731 Our tag: as7e8cf3ad [May 23 18:42:16] DEBUG[2655] chan_sip.c: SIP TIMER: Rescheduling retransmission #1654 (4) REFER - 9 [May 23 18:42:16] DEBUG[2655] chan_sip.c: ** SIP timers: Rescheduling retransmis sion 5 to 4000 ms (t1 500 ms (Retrans id #1654)) [May 23 18:42:16] VERBOSE[2655] logger.c: Retransmitting #4 (no NAT) to 192.168. 10.10:5061: REFER sip:3002@192.168.10.10:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.116:5060;branch=z9hG4bK2f17865b;rport From: ;tag=as7e8cf3ad To: "3002" ;tag=17327731 Contact: Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 CSeq: 102 REFER User-Agent: Asterisk PBX Max-Forwards: 70 Refer-To: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Referred-By: --- [May 23 18:42:16] VERBOSE[2655] logger.c: <--- SIP read from 192.168.10.10:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.130.116:5060;branch=z9hG4bK2f17865b;rport From: ;tag=as7e8cf3ad To: "3002" ;tag=17327731 Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 CSeq: 102 REFER Content-Length: 0 <-------------> [May 23 18:42:16] DEBUG[2655] chan_sip.c: Header 0: SIP/2.0 481 Call Leg/Transac tion Does Not Exist (47) [May 23 18:42:16] DEBUG[2655] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.130 .116:5060;branch=z9hG4bK2f17865b;rport (66) [May 23 18:42:16] DEBUG[2655] chan_sip.c: Header 2: From: ;tag=as7e8cf3ad (47) [May 23 18:42:16] DEBUG[2655] chan_sip.c: Header 3: To: "3002" ;tag=17327731 (48) [May 23 18:42:16] DEBUG[2655] chan_sip.c: Header 4: Call-ID: 85e37580-1de146f2-8 4971-a0aa8c0@192.168.10.10 (54) [May 23 18:42:16] DEBUG[2655] chan_sip.c: Header 5: CSeq: 102 REFER (15) [May 23 18:42:16] DEBUG[2655] chan_sip.c: Header 6: Content-Length: 0 (17) [May 23 18:42:16] DEBUG[2655] chan_sip.c: Header 7: (0) [May 23 18:42:16] VERBOSE[2655] logger.c: --- (7 headers 0 lines) --- [May 23 18:42:16] DEBUG[2655] chan_sip.c: = Found Their Call ID: 85e37580-1de146 f2-84971-a0aa8c0@192.168.10.10 Their Tag 17327731 Our tag: as7e8cf3ad [May 23 18:42:20] DEBUG[2655] chan_sip.c: SIP TIMER: Rescheduling retransmission #1654 (5) REFER - 9 [May 23 18:42:20] DEBUG[2655] chan_sip.c: ** SIP timers: Rescheduling retransmis sion 6 to 4000 ms (t1 500 ms (Retrans id #1654)) [May 23 18:42:20] VERBOSE[2655] logger.c: Retransmitting #5 (no NAT) to 192.168. 10.10:5061: REFER sip:3002@192.168.10.10:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.116:5060;branch=z9hG4bK2f17865b;rport From: ;tag=as7e8cf3ad To: "3002" ;tag=17327731 Contact: Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 CSeq: 102 REFER User-Agent: Asterisk PBX Max-Forwards: 70 Refer-To: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Referred-By: --- [May 23 18:42:20] VERBOSE[2655] logger.c: <--- SIP read from 192.168.10.10:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.130.116:5060;branch=z9hG4bK2f17865b;rport From: ;tag=as7e8cf3ad To: "3002" ;tag=17327731 Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 CSeq: 102 REFER Content-Length: 0 <-------------> [May 23 18:42:20] DEBUG[2655] chan_sip.c: Header 0: SIP/2.0 481 Call Leg/Transac tion Does Not Exist (47) [May 23 18:42:20] DEBUG[2655] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.130 .116:5060;branch=z9hG4bK2f17865b;rport (66) [May 23 18:42:20] DEBUG[2655] chan_sip.c: Header 2: From: ;tag=as7e8cf3ad (47) [May 23 18:42:20] DEBUG[2655] chan_sip.c: Header 3: To: "3002" ;tag=17327731 (48) [May 23 18:42:20] DEBUG[2655] chan_sip.c: Header 4: Call-ID: 85e37580-1de146f2-8 4971-a0aa8c0@192.168.10.10 (54) [May 23 18:42:20] DEBUG[2655] chan_sip.c: Header 5: CSeq: 102 REFER (15) [May 23 18:42:20] DEBUG[2655] chan_sip.c: Header 6: Content-Length: 0 (17) [May 23 18:42:20] DEBUG[2655] chan_sip.c: Header 7: (0) [May 23 18:42:20] VERBOSE[2655] logger.c: --- (7 headers 0 lines) --- [May 23 18:42:20] DEBUG[2655] chan_sip.c: = Found Their Call ID: 85e37580-1de146 f2-84971-a0aa8c0@192.168.10.10 Their Tag 17327731 Our tag: as7e8cf3ad [May 23 18:42:24] DEBUG[2655] chan_sip.c: SIP TIMER: Rescheduling retransmission #1654 (6) REFER - 9 [May 23 18:42:24] DEBUG[2655] chan_sip.c: ** SIP timers: Rescheduling retransmis sion 7 to 4000 ms (t1 500 ms (Retrans id #1654)) [May 23 18:42:24] VERBOSE[2655] logger.c: Retransmitting #6 (no NAT) to 192.168. 10.10:5061: REFER sip:3002@192.168.10.10:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.130.116:5060;branch=z9hG4bK2f17865b;rport From: ;tag=as7e8cf3ad To: "3002" ;tag=17327731 Contact: Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 CSeq: 102 REFER User-Agent: Asterisk PBX Max-Forwards: 70 Refer-To: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Referred-By: --- [May 23 18:42:24] VERBOSE[2655] logger.c: <--- SIP read from 192.168.10.10:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.130.116:5060;branch=z9hG4bK2f17865b;rport From: ;tag=as7e8cf3ad To: "3002" ;tag=17327731 Call-ID: 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 CSeq: 102 REFER Content-Length: 0 <-------------> [May 23 18:42:24] DEBUG[2655] chan_sip.c: Header 0: SIP/2.0 481 Call Leg/Transac tion Does Not Exist (47) [May 23 18:42:24] DEBUG[2655] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.130 .116:5060;branch=z9hG4bK2f17865b;rport (66) [May 23 18:42:24] DEBUG[2655] chan_sip.c: Header 2: From: ;tag=as7e8cf3ad (47) [May 23 18:42:24] DEBUG[2655] chan_sip.c: Header 3: To: "3002" ;tag=17327731 (48) [May 23 18:42:24] DEBUG[2655] chan_sip.c: Header 4: Call-ID: 85e37580-1de146f2-8 4971-a0aa8c0@192.168.10.10 (54) [May 23 18:42:24] DEBUG[2655] chan_sip.c: Header 5: CSeq: 102 REFER (15) [May 23 18:42:24] DEBUG[2655] chan_sip.c: Header 6: Content-Length: 0 (17) [May 23 18:42:24] DEBUG[2655] chan_sip.c: Header 7: (0) [May 23 18:42:24] VERBOSE[2655] logger.c: --- (7 headers 0 lines) --- [May 23 18:42:24] DEBUG[2655] chan_sip.c: = Found Their Call ID: 85e37580-1de146 f2-84971-a0aa8c0@192.168.10.10 Their Tag 17327731 Our tag: as7e8cf3ad [May 23 18:42:28] WARNING[2655] chan_sip.c: Maximum retries exceeded on transmis sion 85e37580-1de146f2-84971-a0aa8c0@192.168.10.10 for seqno 102 (Non-critical R equest) [May 23 18:42:28] VERBOSE[2655] logger.c: Really destroying SIP dialog '85e37580 -1de146f2-84971-a0aa8c0@192.168.10.10' Method: ACK [May 23 18:42:37] DEBUG[2655] chan_sip.c: Header 0: (0) [May 23 18:42:37] DEBUG[2655] chan_sip.c: Line: (0)