*CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> <--- SIP read from 217.113.77.13:56909 ---> INVITE sip:3271492041@217.113.77.17:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK10123CE From: ;tag=6758DAB2-1B1F To: Date: Fri, 18 Apr 2003 19:47:16 GMT Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 1640384452-1896681943-2196747696-809182324 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Remote-Party-ID: ;party=calling;screen=no;privacy=off Timestamp: 1050695236 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 230 v=0 o=CiscoSystemsSIP-GW-UserAgent 4104 4292 IN IP4 217.113.77.13 s=SIP Call c=IN IP4 217.113.77.13 t=0 0 m=audio 18468 RTP/AVP 18 19 c=IN IP4 217.113.77.13 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:19 CN/8000 <-------------> [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 0: INVITE sip:3271492041@217.113.77.17:5060 SIP/2.0 (48) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK10123CE (58) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 2: From: ;tag=6758DAB2-1B1F (48) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 3: To: (34) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 4: Date: Fri, 18 Apr 2003 19:47:16 GMT (35) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 5: Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 (58) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 6: Supported: 100rel,timer (23) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 7: Min-SE: 1800 (13) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 8: Cisco-Guid: 1640384452-1896681943-2196747696-809182324 (54) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 9: User-Agent: Cisco-SIPGateway/IOS-12.x (37) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 10: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER (104) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 11: CSeq: 101 INVITE (16) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 12: Max-Forwards: 70 (16) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 13: Remote-Party-ID: ;party=calling;screen=no;privacy=off (77) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 14: Timestamp: 1050695236 (21) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 15: Contact: (38) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 16: Expires: 180 (12) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 17: Allow-Events: telephone-event (29) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 18: Content-Type: application/sdp (29) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 19: Content-Length: 230 (19) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 20: (0) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: v=0 (3) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: o=CiscoSystemsSIP-GW-UserAgent 4104 4292 IN IP4 217.113.77.13 (61) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: s=SIP Call (10) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: c=IN IP4 217.113.77.13 (22) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: t=0 0 (5) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: m=audio 18468 RTP/AVP 18 19 (27) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: c=IN IP4 217.113.77.13 (22) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=fmtp:18 annexb=yes (20) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=rtpmap:19 CN/8000 (19) --- (20 headers 10 lines) --- [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:2520 do_setnat: Setting NAT on RTP to Off [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:2530 do_setnat: Setting NAT on UDPTL to Off [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4198 sip_alloc: Allocating new SIP dialog for 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 - INVITE (With RTP) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:14188 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:1633 parse_sip_options: Begin: parsing SIP "Supported: 100rel,timer" [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:1641 parse_sip_options: Found SIP option: -100rel- [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:1647 parse_sip_options: Matched SIP option: 100rel [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:1641 parse_sip_options: Found SIP option: -timer- [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:1647 parse_sip_options: Matched SIP option: timer Sending to 217.113.77.13 : 5060 (no NAT) Using INVITE request as basis request - 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 Found peer 'cisco' [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:2520 do_setnat: Setting NAT on RTP to Off [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:2530 do_setnat: Setting NAT on UDPTL to Off Found RTP audio format 18 Found RTP audio format 19 Peer audio RTP is at port 217.113.77.13:18468 [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4749 process_sdp: Peer doesn't provide T.38 UDPTL Found description format G729 for ID 18 Got unsupported a:fmtp in SDP offer Found description format CN for ID 19 [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4961 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x2 (CN), combined - 0x0 (nothing) Peer audio RTP is at port 217.113.77.13:18468 [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:5038 process_sdp: We're settling with these formats: 0x100 (g729) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:13000 handle_request_invite: Checking SIP call limits for device [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:2931 update_call_counter: Updating call counter for incoming call Looking for 3271492041 in internal (domain 217.113.77.17) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:3696 sip_new: *** Our native formats are 0x100 (g729) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:3697 sip_new: *** Joint capabilities are 0x100 (g729) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:3698 sip_new: *** Our capabilities are 0x100 (g729) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:3699 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:3722 sip_new: This channel will not be able to handle video. [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:7729 build_route: build_route: Contact hop: list_route: hop: [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:13074 handle_request_invite: SIP/217.113.77.13-081de210: New call is still down.... Trying... <--- Transmitting (no NAT) to 217.113.77.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK10123CE;received=217.113.77.13 From: ;tag=6758DAB2-1B1F To: Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 CSeq: 101 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Oct 17 15:02:59] DEBUG[14822]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/217.113.77.13-081de210 [Oct 17 15:02:59] DEBUG[14831]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' -- Executing [3271492041@internal:1] Dial("SIP/217.113.77.13-081de210", "SIP/3271492041@172.16.150.100||t") in new stack [Oct 17 15:02:59] DEBUG[14802]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 217.113.77.13 [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:14831 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Oct 17 15:02:59] DEBUG[14802]: chan_sip.c:14773 sip_devicestate: Checking device state for peer 217.113.77.13 [Oct 17 15:02:59] DEBUG[14802]: devicestate.c:287 do_state_change: Changing state for SIP/217.113.77.13 - state 2 (In use) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4198 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Oct 17 15:02:59] DEBUG[14832]: app_queue.c:546 changethread: Device 'SIP/217.113.77.13' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:3696 sip_new: *** Our native formats are 0x100 (g729) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:3697 sip_new: *** Joint capabilities are 0x0 (nothing) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:3698 sip_new: *** Our capabilities are 0x100 (g729) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:3699 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:3701 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:3722 sip_new: This channel will not be able to handle video. [Oct 17 15:02:59] DEBUG[14831]: channel.c:3145 ast_channel_inherit_variables: Not copying variable STACK-internal-3271492041-1. [Oct 17 15:02:59] DEBUG[14831]: channel.c:3145 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Oct 17 15:02:59] DEBUG[14831]: channel.c:3145 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Oct 17 15:02:59] DEBUG[14831]: channel.c:3145 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Oct 17 15:02:59] DEBUG[14831]: channel.c:3145 ast_channel_inherit_variables: Not copying variable SIPURI. [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:2771 sip_call: Outgoing Call for 3271492041 [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:2931 update_call_counter: Updating call counter for outgoing call [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:2779 sip_call: Our T38 capability (3856), joint T38 capability (3856) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:5948 add_sdp: ** Our capability: 0x100 (g729) Video flag: False [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:5949 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 10.100.20.12 port 10526 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:6100 add_sdp: -- Done with adding codecs to SDP [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:6145 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4425 parse_request: Header 0: INVITE sip:3271492041@172.16.150.100 SIP/2.0 (44) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4425 parse_request: Header 1: Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3916dd9b;rport (63) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4425 parse_request: Header 2: From: "7001" ;tag=as427cba2d (51) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4425 parse_request: Header 3: To: (35) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4425 parse_request: Header 4: Contact: (32) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4425 parse_request: Header 5: Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 (54) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4425 parse_request: Header 6: CSeq: 102 INVITE (16) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4425 parse_request: Header 7: User-Agent: gatewaycomms (24) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4425 parse_request: Header 8: Max-Forwards: 70 (16) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4425 parse_request: Header 9: Date: Tue, 17 Oct 2006 13:02:59 GMT (35) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4425 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4425 parse_request: Header 11: Supported: replaces (19) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4425 parse_request: Header 12: Content-Type: application/sdp (29) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4425 parse_request: Header 13: Content-Length: 251 (19) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4425 parse_request: Header 14: (0) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4457 parse_request: Line: v=0 (3) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4457 parse_request: Line: o=root 14799 14799 IN IP4 10.100.20.12 (38) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4457 parse_request: Line: s=session (9) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4457 parse_request: Line: c=IN IP4 10.100.20.12 (21) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4457 parse_request: Line: t=0 0 (5) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4457 parse_request: Line: m=audio 10526 RTP/AVP 18 101 (28) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4457 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4457 parse_request: Line: a=fmtp:18 annexb=no (19) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4457 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4457 parse_request: Line: a=fmtp:101 0-16 (15) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4457 parse_request: Line: a=silenceSupp:off - - - - (25) [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:4457 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 172.16.150.100:5060: INVITE sip:3271492041@172.16.150.100 SIP/2.0 Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3916dd9b;rport From: "7001" ;tag=as427cba2d To: Contact: Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 CSeq: 102 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 17 Oct 2006 13:02:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 251 v=0 o=root 14799 14799 IN IP4 10.100.20.12 s=session c=IN IP4 10.100.20.12 t=0 0 m=audio 10526 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- [Oct 17 15:02:59] DEBUG[14831]: chan_sip.c:1924 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #19 -- Called 3271492041@172.16.150.100 <--- SIP read from 172.16.150.100:5060 ---> SIP/2.0 100 Trying Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 From: "7001" ;tag=as427cba2d To: ;tag=27202 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3916dd9b;rport Supported: timer,100rel Content-Length: 0 <-------------> [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 0: SIP/2.0 100 Trying (18) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 1: Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 (54) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 2: From: "7001" ;tag=as427cba2d (51) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 3: To: ;tag=27202 (45) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 4: CSeq: 102 INVITE (16) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 5: Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3916dd9b;rport (63) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 6: Supported: timer,100rel (23) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 7: Content-Length: 0 (17) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:2066 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #19 - INVITE (got response) [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:2075 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12' Request 102: Found [Oct 17 15:02:59] DEBUG[14822]: chan_sip.c:11331 handle_response_invite: SIP response 100 to standard invite [Oct 17 15:03:00] DEBUG[14831]: chan_sip.c:6200 transmit_response_with_sdp: Setting framing from config on incoming call [Oct 17 15:03:00] DEBUG[14831]: chan_sip.c:5948 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Oct 17 15:03:00] DEBUG[14831]: chan_sip.c:5949 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 217.113.77.17 port 18050 Adding codec 0x100 (g729) to SDP [Oct 17 15:03:00] DEBUG[14831]: chan_sip.c:6100 add_sdp: -- Done with adding codecs to SDP [Oct 17 15:03:00] DEBUG[14831]: chan_sip.c:6145 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) <--- Transmitting (no NAT) to 217.113.77.13:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK10123CE;received=217.113.77.13 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 CSeq: 101 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 197 v=0 o=root 14799 14799 IN IP4 217.113.77.17 s=session c=IN IP4 217.113.77.17 t=0 0 m=audio 18050 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=sendrecv <------------> [Oct 17 15:03:00] DEBUG[14831]: rtp.c:2555 ast_rtp_write: Ooh, format changed from unknown to g729 [Oct 17 15:03:00] DEBUG[14831]: rtp.c:2572 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 [Oct 17 15:03:00] DEBUG[14831]: rtp.c:1129 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Oct 17 15:03:00] DEBUG[14831]: rtp.c:808 ast_rtcp_read: Got RTCP report of 48 bytes <--- SIP read from 172.16.150.100:5060 ---> SIP/2.0 183 Session Progress Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 From: "7001" ;tag=as427cba2d To: ;tag=27202 Content-Type: application/sdp CSeq: 102 INVITE Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3916dd9b;rport Supported: timer,100rel Content-Length: 195 v=0 o=MG4000|2.0 33 33 IN IP4 10.100.1.240 s=- c=IN IP4 10.100.1.240 t=0 0 m=audio 20064 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no a=ptime:20 <-------------> [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 1: Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 (54) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 2: From: "7001" ;tag=as427cba2d (51) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 3: To: ;tag=27202 (45) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 4: Content-Type: application/sdp (29) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 5: CSeq: 102 INVITE (16) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 6: Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3916dd9b;rport (63) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 7: Supported: timer,100rel (23) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 8: Content-Length: 195 (19) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 9: (0) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: v=0 (3) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: o=MG4000|2.0 33 33 IN IP4 10.100.1.240 (38) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: s=- (3) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: c=IN IP4 10.100.1.240 (21) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: t=0 0 (5) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: m=audio 20064 RTP/AVP 18 101 (28) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=fmtp:101 0-15 (15) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=fmtp:18 annexb=no (19) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=ptime:20 (10) --- (9 headers 10 lines) --- [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:2075 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12' Request 102: Found [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:11331 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.100.1.240:20064 [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4749 process_sdp: Peer doesn't provide T.38 UDPTL Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Got unsupported a:ptime in SDP offer [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4961 process_sdp: T38 state changed to 0 on channel SIP/172.16.150.100-081f0a40 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.100.1.240:20064 [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:5038 process_sdp: We're settling with these formats: 0x100 (g729) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:5045 process_sdp: We have an owner, now see if we need to change this call -- SIP/172.16.150.100-081f0a40 is making progress passing it to SIP/217.113.77.13-081de210 [Oct 17 15:03:02] DEBUG[14831]: rtp.c:2555 ast_rtp_write: Ooh, format changed from unknown to g729 [Oct 17 15:03:02] DEBUG[14831]: rtp.c:2572 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 <--- SIP read from 172.16.150.100:5060 ---> SIP/2.0 180 Ringing Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 From: "7001" ;tag=as427cba2d To: ;tag=27202 Content-Type: application/sdp CSeq: 102 INVITE Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3916dd9b;rport Supported: timer,100rel Content-Length: 195 v=0 o=MG4000|2.0 33 33 IN IP4 10.100.1.240 s=- c=IN IP4 10.100.1.240 t=0 0 m=audio 20064 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no a=ptime:20 <-------------> [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 1: Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 (54) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 2: From: "7001" ;tag=as427cba2d (51) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 3: To: ;tag=27202 (45) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 4: Content-Type: application/sdp (29) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 5: CSeq: 102 INVITE (16) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 6: Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3916dd9b;rport (63) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 7: Supported: timer,100rel (23) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 8: Content-Length: 195 (19) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 9: (0) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: v=0 (3) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: o=MG4000|2.0 33 33 IN IP4 10.100.1.240 (38) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: s=- (3) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: c=IN IP4 10.100.1.240 (21) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: t=0 0 (5) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: m=audio 20064 RTP/AVP 18 101 (28) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=fmtp:101 0-15 (15) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=fmtp:18 annexb=no (19) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=ptime:20 (10) --- (9 headers 10 lines) --- [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:2075 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12' Request 102: Found [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:11331 handle_response_invite: SIP response 180 to standard invite [Oct 17 15:03:02] DEBUG[14822]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/172.16.150.100-081f0a40 Found RTP audio format 18 Found RTP audio format 101 [Oct 17 15:03:02] DEBUG[14802]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 172.16.150.100 [Oct 17 15:03:02] DEBUG[14802]: chan_sip.c:14773 sip_devicestate: Checking device state for peer 172.16.150.100 [Oct 17 15:03:02] DEBUG[14802]: channel.c:895 channel_find_locked: Avoiding initial deadlock for channel '0x8202328' Peer audio RTP is at port 10.100.1.240:20064 [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4749 process_sdp: Peer doesn't provide T.38 UDPTL Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Got unsupported a:ptime in SDP offer [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:4961 process_sdp: T38 state changed to 0 on channel SIP/172.16.150.100-081f0a40 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.100.1.240:20064 [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:5038 process_sdp: We're settling with these formats: 0x100 (g729) [Oct 17 15:03:02] DEBUG[14822]: chan_sip.c:5045 process_sdp: We have an owner, now see if we need to change this call -- SIP/172.16.150.100-081f0a40 is ringing [Oct 17 15:03:02] DEBUG[14831]: channel.c:2279 ast_indicate_data: Driver for channel 'SIP/217.113.77.13-081de210' does not support indication 3, emulating it [Oct 17 15:03:02] DEBUG[14831]: channel.c:2428 ast_prod: Prodding channel 'SIP/217.113.77.13-081de210' [Oct 17 15:03:02] WARNING[14831]: channel.c:2651 set_format: Unable to find a codec translation path from g729 to slin [Oct 17 15:03:02] WARNING[14831]: indications.c:121 playtones_alloc: Unable to set 'SIP/217.113.77.13-081de210' to signed linear format (write) -- SIP/172.16.150.100-081f0a40 is making progress passing it to SIP/217.113.77.13-081de210 [Oct 17 15:03:02] DEBUG[14802]: devicestate.c:287 do_state_change: Changing state for SIP/172.16.150.100 - state 6 (Ringing) [Oct 17 15:03:02] DEBUG[14834]: app_queue.c:546 changethread: Device 'SIP/172.16.150.100' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Oct 17 15:03:03] DEBUG[14831]: rtp.c:808 ast_rtcp_read: Got RTCP report of 112 bytes <--- SIP read from 172.16.150.100:5060 ---> SIP/2.0 200 OK Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 From: "7001" ;tag=as427cba2d To: ;tag=27202 Content-Type: application/sdp CSeq: 102 INVITE Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3916dd9b;rport Contact: sip:3271492041@172.16.150.100:5060;user=phone Supported: timer,100rel Content-Length: 195 v=0 o=MG4000|2.0 33 33 IN IP4 10.100.1.240 s=- c=IN IP4 10.100.1.240 t=0 0 m=audio 20064 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no a=ptime:20 <-------------> [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 0: SIP/2.0 200 OK (14) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 1: Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 (54) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 2: From: "7001" ;tag=as427cba2d (51) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 3: To: ;tag=27202 (45) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 4: Content-Type: application/sdp (29) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 5: CSeq: 102 INVITE (16) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 6: Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3916dd9b;rport (63) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 7: Contact: sip:3271492041@172.16.150.100:5060;user=phone (54) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 8: Supported: timer,100rel (23) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 9: Content-Length: 195 (19) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 10: (0) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: v=0 (3) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: o=MG4000|2.0 33 33 IN IP4 10.100.1.240 (38) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: s=- (3) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: c=IN IP4 10.100.1.240 (21) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: t=0 0 (5) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: m=audio 20064 RTP/AVP 18 101 (28) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=fmtp:101 0-15 (15) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=fmtp:18 annexb=no (19) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=ptime:20 (10) --- (10 headers 10 lines) --- [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:2015 __sip_ack: Acked pending invite 102 [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:2033 __sip_ack: Stopping retransmission on '5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12' of Request 102: Match Not Found [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:11331 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.100.1.240:20064 [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4749 process_sdp: Peer doesn't provide T.38 UDPTL Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Got unsupported a:ptime in SDP offer [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4961 process_sdp: T38 state changed to 0 on channel SIP/172.16.150.100-081f0a40 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.100.1.240:20064 [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:5038 process_sdp: We're settling with these formats: 0x100 (g729) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:5045 process_sdp: We have an owner, now see if we need to change this call [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:2931 update_call_counter: Updating call counter for outgoing call [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:7729 build_route: build_route: Contact hop: sip:3271492041@172.16.150.100:5060;user=phone list_route: hop: [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:5469 reqprep: Strict routing enforced for session 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 set_destination: Parsing for address/port to send to set_destination: set destination to 172.16.150.100, port 5060 Transmitting (no NAT) to 172.16.150.100:5060: ACK sip:3271492041@172.16.150.100:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK2083cb2b;rport From: "7001" ;tag=as427cba2d To: ;tag=27202 Contact: Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Oct 17 15:03:03] DEBUG[14831]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/172.16.150.100-081f0a40 -- SIP/172.16.150.100-081f0a40 answered SIP/217.113.77.13-081de210 [Oct 17 15:03:03] DEBUG[14802]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 172.16.150.100 [Oct 17 15:03:03] DEBUG[14802]: chan_sip.c:14773 sip_devicestate: Checking device state for peer 172.16.150.100 [Oct 17 15:03:03] DEBUG[14802]: devicestate.c:287 do_state_change: Changing state for SIP/172.16.150.100 - state 2 (In use) [Oct 17 15:03:03] DEBUG[14835]: app_queue.c:546 changethread: Device 'SIP/172.16.150.100' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 17 15:03:03] DEBUG[14831]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/217.113.77.13-081de210 [Oct 17 15:03:03] DEBUG[14831]: chan_sip.c:3378 sip_answer: SIP answering channel: SIP/217.113.77.13-081de210 [Oct 17 15:03:03] DEBUG[14802]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 217.113.77.13 [Oct 17 15:03:03] DEBUG[14802]: chan_sip.c:14773 sip_devicestate: Checking device state for peer 217.113.77.13 [Oct 17 15:03:03] DEBUG[14831]: chan_sip.c:6200 transmit_response_with_sdp: Setting framing from config on incoming call [Oct 17 15:03:03] DEBUG[14802]: channel.c:895 channel_find_locked: Avoiding initial deadlock for channel '0x81efc70' [Oct 17 15:03:03] DEBUG[14831]: chan_sip.c:5948 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Oct 17 15:03:03] DEBUG[14831]: chan_sip.c:5949 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 217.113.77.17 port 18050 Adding codec 0x100 (g729) to SDP [Oct 17 15:03:03] DEBUG[14831]: chan_sip.c:6100 add_sdp: -- Done with adding codecs to SDP [Oct 17 15:03:03] DEBUG[14831]: chan_sip.c:6145 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) <--- Reliably Transmitting (no NAT) to 217.113.77.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK10123CE;received=217.113.77.13 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 CSeq: 101 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 197 v=0 o=root 14799 14800 IN IP4 217.113.77.17 s=session c=IN IP4 217.113.77.17 t=0 0 m=audio 18050 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=sendrecv <------------> [Oct 17 15:03:03] DEBUG[14831]: chan_sip.c:1924 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #23 [Oct 17 15:03:03] DEBUG[14831]: rtp.c:2572 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20 <--- SIP read from 217.113.77.13:56909 ---> ACK sip:3271492041@217.113.77.17:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK1021CF6 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Date: Fri, 18 Apr 2003 19:47:16 GMT Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 <-------------> [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 0: ACK sip:3271492041@217.113.77.17:5060 SIP/2.0 (45) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK1021CF6 (58) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 2: From: ;tag=6758DAB2-1B1F (48) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 3: To: ;tag=as5b009254 (49) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 4: Date: Fri, 18 Apr 2003 19:47:16 GMT (35) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 5: Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 (58) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 6: Max-Forwards: 70 (16) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 7: CSeq: 101 ACK (13) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 8: Content-Length: 0 (17) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:14188 handle_request: **** Received ACK (6) - Command in SIP ACK [Oct 17 15:03:03] DEBUG[14802]: devicestate.c:287 do_state_change: Changing state for SIP/217.113.77.13 - state 2 (In use) [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:2023 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #23 [Oct 17 15:03:03] DEBUG[14822]: chan_sip.c:2033 __sip_ack: Stopping retransmission on '64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13' of Response 101: Match Not Found [Oct 17 15:03:03] DEBUG[14836]: app_queue.c:546 changethread: Device 'SIP/217.113.77.13' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Oct 17 15:03:04] DEBUG[14831]: rtp.c:808 ast_rtcp_read: Got RTCP report of 72 bytes [Oct 17 15:03:05] DEBUG[14831]: rtp.c:808 ast_rtcp_read: Got RTCP report of 112 bytes *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> <--- SIP read from 172.16.150.100:5060 ---> INVITE sip:7001@10.100.20.12:5060;user=phone SIP/2.0 Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 From: ;tag=27202 To: "7001" ;tag=as427cba2d Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.150.100:5060;branch=z9hG4bK-8160000000000031-ac106464-34 Contact: sip:3271492041@172.16.150.100:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE Supported: timer,100rel Max-Forwards: 70 Content-Length: 144 v=0 o=MG4000|2.0 34 34 IN IP4 10.100.1.240 s=- c=IN IP4 10.100.1.240 t=0 0 m=audio 20064 RTP/AVP 0 a=silenceSupp:off - - - - a=ptime:10 <-------------> [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 0: INVITE sip:7001@10.100.20.12:5060;user=phone SIP/2.0 (52) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 1: Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 (54) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 2: From: ;tag=27202 (47) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 3: To: "7001" ;tag=as427cba2d (49) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 4: Content-Type: application/sdp (29) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 5: CSeq: 1 INVITE (14) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.150.100:5060;branch=z9hG4bK-8160000000000031-ac106464-34 (80) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 7: Contact: sip:3271492041@172.16.150.100:5060;user=phone (54) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 9: Supported: timer,100rel (23) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 10: Max-Forwards: 70 (16) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 11: Content-Length: 144 (19) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 12: (0) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: v=0 (3) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: o=MG4000|2.0 34 34 IN IP4 10.100.1.240 (38) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: s=- (3) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: c=IN IP4 10.100.1.240 (21) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: t=0 0 (5) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: m=audio 20064 RTP/AVP 0 (23) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=silenceSupp:off - - - - (25) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=ptime:10 (10) --- (12 headers 8 lines) --- [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:14188 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:1633 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:1641 parse_sip_options: Found SIP option: -timer- [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:1647 parse_sip_options: Matched SIP option: timer [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:1641 parse_sip_options: Found SIP option: -100rel- [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:1647 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.150.100 : 5060 (no NAT) Found RTP audio format 0 Peer audio RTP is at port 10.100.1.240:20064 [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4749 process_sdp: Peer doesn't provide T.38 UDPTL Got unsupported a:ptime in SDP offer [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4961 process_sdp: T38 state changed to 0 on channel SIP/172.16.150.100-081f0a40 Capabilities: us - 0x100 (g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4996 process_sdp: Have T.38 but no audio codecs, accepting offer anyway [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:13052 handle_request_invite: Got a SIP re-invite for call 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:13141 handle_request_invite: SIP/172.16.150.100-081f0a40: This call is UP.... [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:5948 add_sdp: ** Our capability: 0x100 (g729) Video flag: True [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:5949 add_sdp: ** Our prefcodec: 0x100 (g729) Audio is at 10.100.20.12 port 10526 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:6100 add_sdp: -- Done with adding codecs to SDP [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:6145 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729) <--- Reliably Transmitting (no NAT) to 172.16.150.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.150.100:5060;branch=z9hG4bK-8160000000000031-ac106464-34;received=172.16.150.100 From: ;tag=27202 To: "7001" ;tag=as427cba2d Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 251 v=0 o=root 14799 14800 IN IP4 10.100.20.12 s=session c=IN IP4 10.100.20.12 t=0 0 m=audio 10526 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv <------------> [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:1924 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #24 <--- SIP read from 172.16.150.100:5060 ---> ACK sip:7001@10.100.20.12:5060;user=phone SIP/2.0 Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 From: ;tag=27202 To: "7001" ;tag=as427cba2d CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.150.100:5060;branch=z9hG4bK-8160000000000031-ac106464-34 Max-Forwards: 70 Content-Length: 0 <-------------> [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 0: ACK sip:7001@10.100.20.12:5060;user=phone SIP/2.0 (49) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 1: Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 (54) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 2: From: ;tag=27202 (47) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 3: To: "7001" ;tag=as427cba2d (49) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 4: CSeq: 1 ACK (11) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.150.100:5060;branch=z9hG4bK-8160000000000031-ac106464-34 (80) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 6: Max-Forwards: 70 (16) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 7: Content-Length: 0 (17) [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:14188 handle_request: **** Received ACK (6) - Command in SIP ACK [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:2023 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #24 [Oct 17 15:03:09] DEBUG[14822]: chan_sip.c:2033 __sip_ack: Stopping retransmission on '5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12' of Response 1: Match Not Found [Oct 17 15:03:09] DEBUG[14831]: rtp.c:1129 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Oct 17 15:03:09] DEBUG[14831]: rtp.c:1129 ast_rtp_read: Forcing Marker bit, because SSRC has changed [Oct 17 15:03:10] DEBUG[14831]: rtp.c:808 ast_rtcp_read: Got RTCP report of 48 bytes [Oct 17 15:03:10] DEBUG[14831]: rtp.c:808 ast_rtcp_read: Got RTCP report of 112 bytes *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> [Oct 17 15:03:16] DEBUG[14831]: rtp.c:808 ast_rtcp_read: Got RTCP report of 88 bytes [Oct 17 15:03:22] DEBUG[14831]: rtp.c:808 ast_rtcp_read: Got RTCP report of 88 bytes [Oct 17 15:03:29] DEBUG[14831]: rtp.c:808 ast_rtcp_read: Got RTCP report of 88 bytes [Oct 17 15:03:32] DEBUG[14831]: rtp.c:808 ast_rtcp_read: Got RTCP report of 88 bytes [Oct 17 15:03:35] DEBUG[14831]: rtp.c:808 ast_rtcp_read: Got RTCP report of 88 bytes [Oct 17 15:03:41] DEBUG[14831]: rtp.c:808 ast_rtcp_read: Got RTCP report of 88 bytes <--- SIP read from 217.113.77.13:56909 ---> INVITE sip:3271492041@217.113.77.17:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK1031FE7 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Date: Fri, 18 Apr 2003 19:47:58 GMT Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 1640384452-1896681943-2196747696-809182324 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 102 INVITE Max-Forwards: 70 Remote-Party-ID: ;party=calling;screen=no;privacy=off Timestamp: 1050695278 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 398 v=0 o=CiscoSystemsSIP-GW-UserAgent 4104 4293 IN IP4 217.113.77.13 s=SIP Call c=IN IP4 217.113.77.13 t=0 0 m=image 18468 udptl t38 c=IN IP4 217.113.77.13 a=T38FaxVersion:0 a=T38MaxBitRate:7200 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy <-------------> [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 0: INVITE sip:3271492041@217.113.77.17:5060 SIP/2.0 (48) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK1031FE7 (58) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 2: From: ;tag=6758DAB2-1B1F (48) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 3: To: ;tag=as5b009254 (49) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 4: Date: Fri, 18 Apr 2003 19:47:58 GMT (35) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 5: Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 (58) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 6: Supported: 100rel,timer (23) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 7: Min-SE: 1800 (13) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 8: Cisco-Guid: 1640384452-1896681943-2196747696-809182324 (54) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 9: User-Agent: Cisco-SIPGateway/IOS-12.x (37) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 10: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER (104) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 11: CSeq: 102 INVITE (16) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 12: Max-Forwards: 70 (16) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 13: Remote-Party-ID: ;party=calling;screen=no;privacy=off (77) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 14: Timestamp: 1050695278 (21) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 15: Contact: (38) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 16: Expires: 180 (12) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 17: Allow-Events: telephone-event (29) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 18: Content-Type: application/sdp (29) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 19: Content-Length: 398 (19) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 20: (0) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: v=0 (3) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: o=CiscoSystemsSIP-GW-UserAgent 4104 4293 IN IP4 217.113.77.13 (61) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: s=SIP Call (10) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: c=IN IP4 217.113.77.13 (22) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: t=0 0 (5) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: m=image 18468 udptl t38 (23) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: c=IN IP4 217.113.77.13 (22) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxVersion:0 (17) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38MaxBitRate:7200 (20) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxFillBitRemoval:0 (24) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxTranscodingMMR:0 (24) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxTranscodingJBIG:0 (25) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxRateManagement:transferredTCF (37) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxMaxBuffer:200 (21) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxMaxDatagram:72 (22) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxUdpEC:t38UDPRedundancy (30) --- (20 headers 16 lines) --- [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:14188 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 217.113.77.13 : 5060 (no NAT) Got T.38 offer in SDP in dialog 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4679 process_sdp: T38 state changed to 4 on channel SIP/217.113.77.13-081de210 Peer doesn't provide audio [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4745 process_sdp: Peer T.38 UDPTL is at port 217.113.77.13:18468 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4893 process_sdp: FaxVersion: 0 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4869 process_sdp: T38MaxBitRate: 7200 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4907 process_sdp: FillBitRemoval: 0 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4913 process_sdp: Transcoding MMR: 0 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4920 process_sdp: Transcoding JBIG: 0 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4926 process_sdp: RateMangement: transferredTCF [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4865 process_sdp: MaxBufferSize:200 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4901 process_sdp: FaxMaxDatagram: 72 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4934 process_sdp: UDP EC: t38UDPRedundancy [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4954 process_sdp: Our T38 capability = (3856), peer T38 capability (1824), joint T38 capability (1824) Capabilities: us - 0x100 (g729), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4996 process_sdp: Have T.38 but no audio codecs, accepting offer anyway [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:13052 handle_request_invite: Got a SIP re-invite for call 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:13141 handle_request_invite: SIP/217.113.77.13-081de210: This call is UP.... [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:16261 sip_handle_t38_reinvite: Sending reinvite on SIP '5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12' - It's UDPTL soon redirected to us (IP 10.100.20.12) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:5469 reqprep: Strict routing enforced for session 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 set_destination: Parsing for address/port to send to set_destination: set destination to 172.16.150.100, port 5060 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:5801 add_t38_sdp: T.38 UDPTL is at 10.100.20.12 port 4598 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:5807 add_t38_sdp: Our T38 capability (3856), peer T38 capability (1824), joint capability (1824) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:5738 t38_get_rate: T38MaxFaxRate 7200 found [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:1582 initialize_initreq: Initializing already initialized SIP dialog 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 (presumably reinvite) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 0: INVITE sip:3271492041@172.16.150.100:5060;user=phone SIP/2.0 (60) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 1: Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK0ceec928;rport (63) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 2: From: "7001" ;tag=as427cba2d (51) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 3: To: ;tag=27202 (45) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 4: Contact: (32) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 5: Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 (54) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 6: CSeq: 103 INVITE (16) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 7: User-Agent: gatewaycomms (24) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 8: Max-Forwards: 70 (16) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 10: Supported: replaces (19) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 11: X-asterisk-info: SIP re-invite (T38 switchover) (47) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 12: Content-Type: application/sdp (29) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 13: Content-Length: 261 (19) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 14: (0) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: v=0 (3) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: o=root 14799 14801 IN IP4 10.100.20.12 (38) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: s=session (9) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: c=IN IP4 10.100.20.12 (21) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: t=0 0 (5) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: m=image 4598 udptl t38 (22) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxVersion:0 (17) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38MaxBitRate:7200 (20) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxFillBitRemoval:0 (24) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxTranscodingMMR:0 (24) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxTranscodingJBIG:0 (25) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxRateManagement:transferredTCF (37) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxMaxBuffer:72 (20) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxMaxDatagram:72 (22) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxUdpEC:t38UDPRedundancy (30) Reliably Transmitting (no NAT) to 172.16.150.100:5060: INVITE sip:3271492041@172.16.150.100:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK0ceec928;rport From: "7001" ;tag=as427cba2d To: ;tag=27202 Contact: Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 CSeq: 103 INVITE User-Agent: gatewaycomms Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-info: SIP re-invite (T38 switchover) Content-Type: application/sdp Content-Length: 261 v=0 o=root 14799 14801 IN IP4 10.100.20.12 s=session c=IN IP4 10.100.20.12 t=0 0 m=image 4598 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:7200 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy --- [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:1924 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #25 <--- SIP read from 172.16.150.100:5060 ---> SIP/2.0 200 OK Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 From: "7001" ;tag=as427cba2d To: ;tag=27202 Content-Type: application/sdp CSeq: 103 INVITE Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK0ceec928;rport Contact: sip:3271492041@172.16.150.100:5060;user=phone Supported: timer,100rel Content-Length: 105 v=0 o=MG4000|2.0 36 36 IN IP4 10.100.1.240 s=- c=IN IP4 10.100.1.240 t=0 0 m=image 20064 udptl t38 <-------------> [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 0: SIP/2.0 200 OK (14) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 1: Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 (54) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 2: From: "7001" ;tag=as427cba2d (51) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 3: To: ;tag=27202 (45) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 4: Content-Type: application/sdp (29) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 5: CSeq: 103 INVITE (16) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 6: Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK0ceec928;rport (63) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 7: Contact: sip:3271492041@172.16.150.100:5060;user=phone (54) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 8: Supported: timer,100rel (23) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 9: Content-Length: 105 (19) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 10: (0) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: v=0 (3) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: o=MG4000|2.0 36 36 IN IP4 10.100.1.240 (38) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: s=- (3) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: c=IN IP4 10.100.1.240 (21) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: t=0 0 (5) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: m=image 20064 udptl t38 (23) --- (10 headers 6 lines) --- [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:2015 __sip_ack: Acked pending invite 103 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:2023 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #25 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:2033 __sip_ack: Stopping retransmission on '5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12' of Request 103: Match Not Found [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:11329 handle_response_invite: SIP response 200 to RE-invite on outgoing call 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 Got T.38 offer in SDP in dialog 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4679 process_sdp: T38 state changed to 4 on channel SIP/172.16.150.100-081f0a40 Peer doesn't provide audio [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4745 process_sdp: Peer T.38 UDPTL is at port 10.100.1.240:20064 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4954 process_sdp: Our T38 capability = (3856), peer T38 capability (1824), joint T38 capability (1824) Capabilities: us - 0x100 (g729), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4996 process_sdp: Have T.38 but no audio codecs, accepting offer anyway [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:2931 update_call_counter: Updating call counter for outgoing call [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:7668 build_route: build_route: Retaining previous route: [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:16289 sip_handle_t38_reinvite: Responding 200 OK on SIP '64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13' - It's UDPTL soon redirected to us (IP 217.113.77.17) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:16294 sip_handle_t38_reinvite: T38 changed state to 5 on channel SIP/172.16.150.100-081f0a40 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:16295 sip_handle_t38_reinvite: T38 changed state to 5 on channel SIP/217.113.77.13-081de210 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:5801 add_t38_sdp: T.38 UDPTL is at 217.113.77.17 port 4906 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:5807 add_t38_sdp: Our T38 capability (3856), peer T38 capability (1824), joint capability (1824) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:5738 t38_get_rate: T38MaxFaxRate 7200 found <--- Reliably Transmitting (no NAT) to 217.113.77.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK10123CE;received=217.113.77.13 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 CSeq: 101 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 261 v=0 o=root 14799 14801 IN IP4 217.113.77.17 s=session c=IN IP4 217.113.77.17 t=0 0 m=image 4906 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:7200 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy <------------> [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:1924 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #26 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:5469 reqprep: Strict routing enforced for session 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 set_destination: Parsing for address/port to send to set_destination: set destination to 172.16.150.100, port 5060 Transmitting (no NAT) to 172.16.150.100:5060: ACK sip:3271492041@172.16.150.100:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK009560ac;rport From: "7001" ;tag=as427cba2d To: ;tag=27202 Contact: Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 CSeq: 103 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- <--- SIP read from 217.113.77.13:56909 ---> INVITE sip:3271492041@217.113.77.17:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK1031FE7 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Date: Fri, 18 Apr 2003 19:47:58 GMT Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 1640384452-1896681943-2196747696-809182324 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 102 INVITE Max-Forwards: 70 Remote-Party-ID: ;party=calling;screen=no;privacy=off Timestamp: 1050695278 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 398 v=0 o=CiscoSystemsSIP-GW-UserAgent 4104 4293 IN IP4 217.113.77.13 s=SIP Call c=IN IP4 217.113.77.13 t=0 0 m=image 18468 udptl t38 c=IN IP4 217.113.77.13 a=T38FaxVersion:0 a=T38MaxBitRate:7200 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy <-------------> [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 0: INVITE sip:3271492041@217.113.77.17:5060 SIP/2.0 (48) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK1031FE7 (58) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 2: From: ;tag=6758DAB2-1B1F (48) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 3: To: ;tag=as5b009254 (49) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 4: Date: Fri, 18 Apr 2003 19:47:58 GMT (35) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 5: Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 (58) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 6: Supported: 100rel,timer (23) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 7: Min-SE: 1800 (13) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 8: Cisco-Guid: 1640384452-1896681943-2196747696-809182324 (54) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 9: User-Agent: Cisco-SIPGateway/IOS-12.x (37) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 10: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER (104) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 11: CSeq: 102 INVITE (16) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 12: Max-Forwards: 70 (16) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 13: Remote-Party-ID: ;party=calling;screen=no;privacy=off (77) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 14: Timestamp: 1050695278 (21) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 15: Contact: (38) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 16: Expires: 180 (12) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 17: Allow-Events: telephone-event (29) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 18: Content-Type: application/sdp (29) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 19: Content-Length: 398 (19) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 20: (0) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: v=0 (3) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: o=CiscoSystemsSIP-GW-UserAgent 4104 4293 IN IP4 217.113.77.13 (61) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: s=SIP Call (10) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: c=IN IP4 217.113.77.13 (22) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: t=0 0 (5) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: m=image 18468 udptl t38 (23) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: c=IN IP4 217.113.77.13 (22) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxVersion:0 (17) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38MaxBitRate:7200 (20) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxFillBitRemoval:0 (24) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxTranscodingMMR:0 (24) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxTranscodingJBIG:0 (25) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxRateManagement:transferredTCF (37) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxMaxBuffer:200 (21) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxMaxDatagram:72 (22) [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxUdpEC:t38UDPRedundancy (30) --- (20 headers 16 lines) --- [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:14188 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:14207 handle_request: Ignoring SIP message because of retransmit (INVITE Seqno 102, ours 102) Ignoring this INVITE request [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:13054 handle_request_invite: Got a SIP re-transmit of INVITE for call 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 [Oct 17 15:03:41] DEBUG[14822]: chan_sip.c:13141 handle_request_invite: SIP/217.113.77.13-081de210: This call is UP.... [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:1819 retrans_pkt: SIP TIMER: Rescheduling retransmission #26 (1) SIP/2.0 - 1 [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:1833 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #26)) Retransmitting #1 (no NAT) to 217.113.77.13:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK10123CE;received=217.113.77.13 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 CSeq: 101 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 261 v=0 o=root 14799 14801 IN IP4 217.113.77.17 s=session c=IN IP4 217.113.77.17 t=0 0 m=image 4906 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:7200 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy --- <--- SIP read from 217.113.77.13:56909 ---> INVITE sip:3271492041@217.113.77.17:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK1031FE7 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Date: Fri, 18 Apr 2003 19:47:59 GMT Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 1640384452-1896681943-2196747696-809182324 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 102 INVITE Max-Forwards: 70 Remote-Party-ID: ;party=calling;screen=no;privacy=off Timestamp: 1050695279 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 398 v=0 o=CiscoSystemsSIP-GW-UserAgent 4104 4293 IN IP4 217.113.77.13 s=SIP Call c=IN IP4 217.113.77.13 t=0 0 m=image 18468 udptl t38 c=IN IP4 217.113.77.13 a=T38FaxVersion:0 a=T38MaxBitRate:7200 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy <-------------> [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 0: INVITE sip:3271492041@217.113.77.17:5060 SIP/2.0 (48) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK1031FE7 (58) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 2: From: ;tag=6758DAB2-1B1F (48) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 3: To: ;tag=as5b009254 (49) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 4: Date: Fri, 18 Apr 2003 19:47:59 GMT (35) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 5: Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 (58) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 6: Supported: 100rel,timer (23) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 7: Min-SE: 1800 (13) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 8: Cisco-Guid: 1640384452-1896681943-2196747696-809182324 (54) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 9: User-Agent: Cisco-SIPGateway/IOS-12.x (37) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 10: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER (104) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 11: CSeq: 102 INVITE (16) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 12: Max-Forwards: 70 (16) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 13: Remote-Party-ID: ;party=calling;screen=no;privacy=off (77) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 14: Timestamp: 1050695279 (21) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 15: Contact: (38) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 16: Expires: 180 (12) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 17: Allow-Events: telephone-event (29) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 18: Content-Type: application/sdp (29) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 19: Content-Length: 398 (19) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 20: (0) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: v=0 (3) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: o=CiscoSystemsSIP-GW-UserAgent 4104 4293 IN IP4 217.113.77.13 (61) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: s=SIP Call (10) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: c=IN IP4 217.113.77.13 (22) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: t=0 0 (5) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: m=image 18468 udptl t38 (23) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: c=IN IP4 217.113.77.13 (22) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxVersion:0 (17) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38MaxBitRate:7200 (20) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxFillBitRemoval:0 (24) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxTranscodingMMR:0 (24) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxTranscodingJBIG:0 (25) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxRateManagement:transferredTCF (37) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxMaxBuffer:200 (21) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxMaxDatagram:72 (22) [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:4457 parse_request: Line: a=T38FaxUdpEC:t38UDPRedundancy (30) --- (20 headers 16 lines) --- [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:14188 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:14207 handle_request: Ignoring SIP message because of retransmit (INVITE Seqno 102, ours 102) Ignoring this INVITE request [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:13054 handle_request_invite: Got a SIP re-transmit of INVITE for call 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 [Oct 17 15:03:42] DEBUG[14822]: chan_sip.c:13141 handle_request_invite: SIP/217.113.77.13-081de210: This call is UP.... <--- SIP read from 217.113.77.13:56909 ---> BYE sip:3271492041@217.113.77.17:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK104963 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Date: Fri, 18 Apr 2003 19:47:59 GMT Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 Timestamp: 1050695280 CSeq: 103 BYE Reason: Q.850;cause=16 Content-Length: 0 <-------------> [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 0: BYE sip:3271492041@217.113.77.17:5060 SIP/2.0 (45) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK104963 (57) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 2: From: ;tag=6758DAB2-1B1F (48) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 3: To: ;tag=as5b009254 (49) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 4: Date: Fri, 18 Apr 2003 19:47:59 GMT (35) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 5: Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 (58) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 6: User-Agent: Cisco-SIPGateway/IOS-12.x (37) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 7: Max-Forwards: 70 (16) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 8: Timestamp: 1050695280 (21) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 9: CSeq: 103 BYE (13) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 10: Reason: Q.850;cause=16 (22) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 11: Content-Length: 0 (17) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:14188 handle_request: **** Received BYE (8) - Command in SIP BYE <--- Reliably Transmitting (no NAT) to 217.113.77.13:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK10123CE;received=217.113.77.13 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 CSeq: 101 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:1924 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #27 Sending to 217.113.77.13 : 5060 (no NAT) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:13789 handle_request_bye: Received bye, issuing owner hangup . <--- Transmitting (no NAT) to 217.113.77.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK104963;received=217.113.77.13 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 CSeq: 103 BYE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Oct 17 15:03:43] DEBUG[14831]: channel.c:3638 ast_generic_bridge: Didn't get a frame from channel: SIP/217.113.77.13-081de210 [Oct 17 15:03:43] DEBUG[14831]: channel.c:3939 ast_channel_bridge: Bridge stops bridging channels SIP/217.113.77.13-081de210 and SIP/172.16.150.100-081f0a40 [Oct 17 15:03:43] DEBUG[14831]: channel.c:1558 ast_hangup: Hanging up channel 'SIP/172.16.150.100-081f0a40' [Oct 17 15:03:43] DEBUG[14831]: chan_sip.c:3230 sip_hangup: Hangup call SIP/172.16.150.100-081f0a40, SIP callid 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12) [Oct 17 15:03:43] DEBUG[14831]: chan_sip.c:3238 sip_hangup: update_call_counter(3271492041) - decrement call limit counter on hangup [Oct 17 15:03:43] DEBUG[14831]: chan_sip.c:2931 update_call_counter: Updating call counter for outgoing call Scheduling destruction of SIP dialog '5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12' in 32000 ms (Method: ACK) [Oct 17 15:03:43] DEBUG[14831]: chan_sip.c:5469 reqprep: Strict routing enforced for session 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 set_destination: Parsing for address/port to send to set_destination: set destination to 172.16.150.100, port 5060 Reliably Transmitting (no NAT) to 172.16.150.100:5060: BYE sip:3271492041@172.16.150.100:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3e481ffc;rport From: "7001" ;tag=as427cba2d To: ;tag=27202 Contact: Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 CSeq: 104 BYE User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Oct 17 15:03:43] DEBUG[14831]: chan_sip.c:1924 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #29 [Oct 17 15:03:43] DEBUG[14831]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/172.16.150.100-081f0a40 [Oct 17 15:03:43] DEBUG[14831]: rtp.c:1416 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Oct 17 15:03:43] DEBUG[14831]: app_dial.c:1639 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Oct 17 15:03:43] DEBUG[14831]: pbx.c:2363 __ast_pbx_run: Spawn extension (internal,3271492041,1) exited non-zero on 'SIP/217.113.77.13-081de210' == Spawn extension (internal, 3271492041, 1) exited non-zero on 'SIP/217.113.77.13-081de210' [Oct 17 15:03:43] DEBUG[14802]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 172.16.150.100 [Oct 17 15:03:43] DEBUG[14802]: chan_sip.c:14773 sip_devicestate: Checking device state for peer 172.16.150.100 [Oct 17 15:03:43] DEBUG[14802]: devicestate.c:287 do_state_change: Changing state for SIP/172.16.150.100 - state 1 (Not in use) [Oct 17 15:03:43] DEBUG[14837]: app_queue.c:546 changethread: Device 'SIP/172.16.150.100' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Oct 17 15:03:43] DEBUG[14831]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '7001' [Oct 17 15:03:43] DEBUG[14831]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '7001' [Oct 17 15:03:43] DEBUG[14831]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '3271492041' [Oct 17 15:03:43] DEBUG[14831]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'internal' [Oct 17 15:03:43] DEBUG[14831]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'SIP/217.113.77.13-081de210' [Oct 17 15:03:43] DEBUG[14831]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'SIP/172.16.150.100-081f0a40' [Oct 17 15:03:43] DEBUG[14831]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'Dial' [Oct 17 15:03:43] DEBUG[14831]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'SIP/3271492041@172.16.150.100||t' [Oct 17 15:03:43] DEBUG[14831]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2006-10-17 15:02:59' [Oct 17 15:03:43] DEBUG[14831]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2006-10-17 15:03:03' [Oct 17 15:03:43] DEBUG[14831]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '2006-10-17 15:03:43' [Oct 17 15:03:43] DEBUG[14831]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '44' [Oct 17 15:03:43] DEBUG[14831]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '40' [Oct 17 15:03:43] DEBUG[14831]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Oct 17 15:03:43] DEBUG[14831]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Oct 17 15:03:43] DEBUG[14831]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '' [Oct 17 15:03:43] DEBUG[14831]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '1161090179.0' [Oct 17 15:03:43] DEBUG[14831]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '' [Oct 17 15:03:43] DEBUG[14831]: channel.c:1558 ast_hangup: Hanging up channel 'SIP/217.113.77.13-081de210' [Oct 17 15:03:43] DEBUG[14831]: chan_sip.c:3230 sip_hangup: Hangup call SIP/217.113.77.13-081de210, SIP callid 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13) [Oct 17 15:03:43] DEBUG[14831]: chan_sip.c:3238 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Oct 17 15:03:43] DEBUG[14831]: chan_sip.c:2931 update_call_counter: Updating call counter for incoming call [Oct 17 15:03:43] DEBUG[14831]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/217.113.77.13-081de210 [Oct 17 15:03:43] DEBUG[14802]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 217.113.77.13 [Oct 17 15:03:43] DEBUG[14802]: chan_sip.c:14773 sip_devicestate: Checking device state for peer 217.113.77.13 [Oct 17 15:03:43] DEBUG[14802]: devicestate.c:287 do_state_change: Changing state for SIP/217.113.77.13 - state 1 (Not in use) [Oct 17 15:03:43] DEBUG[14838]: app_queue.c:546 changethread: Device 'SIP/217.113.77.13' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 172.16.150.100:5060 ---> SIP/2.0 200 OK Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 From: "7001" ;tag=as427cba2d To: ;tag=27202 CSeq: 104 BYE Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3e481ffc;rport Contact: sip:3271492041@172.16.150.100:5060;user=phone Supported: timer,100rel Content-Length: 0 <-------------> [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 0: SIP/2.0 200 OK (14) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 1: Call-ID: 5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12 (54) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 2: From: "7001" ;tag=as427cba2d (51) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 3: To: ;tag=27202 (45) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 4: CSeq: 104 BYE (13) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 5: Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3e481ffc;rport (63) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 6: Contact: sip:3271492041@172.16.150.100:5060;user=phone (54) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 7: Supported: timer,100rel (23) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 8: Content-Length: 0 (17) [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:4425 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:2023 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #29 [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:2033 __sip_ack: Stopping retransmission on '5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12' of Request 104: Match Not Found Really destroying SIP dialog '5b2dffad0ad19a251aaa113101fe7ff9@10.100.20.12' Method: ACK [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:1819 retrans_pkt: SIP TIMER: Rescheduling retransmission #26 (2) SIP/2.0 - 1 [Oct 17 15:03:43] DEBUG[14822]: chan_sip.c:1833 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #26)) Retransmitting #2 (no NAT) to 217.113.77.13:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK10123CE;received=217.113.77.13 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 CSeq: 101 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 261 v=0 o=root 14799 14801 IN IP4 217.113.77.17 s=session c=IN IP4 217.113.77.17 t=0 0 m=image 4906 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:7200 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy --- [Oct 17 15:03:44] DEBUG[14822]: chan_sip.c:1819 retrans_pkt: SIP TIMER: Rescheduling retransmission #27 (1) SIP/2.0 - 1 [Oct 17 15:03:44] DEBUG[14822]: chan_sip.c:1833 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #27)) Retransmitting #1 (no NAT) to 217.113.77.13:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK10123CE;received=217.113.77.13 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 CSeq: 101 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Oct 17 15:03:45] DEBUG[14822]: chan_sip.c:1819 retrans_pkt: SIP TIMER: Rescheduling retransmission #27 (2) SIP/2.0 - 1 [Oct 17 15:03:45] DEBUG[14822]: chan_sip.c:1833 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #27)) Retransmitting #2 (no NAT) to 217.113.77.13:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK10123CE;received=217.113.77.13 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 CSeq: 101 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Oct 17 15:03:45] DEBUG[14822]: chan_sip.c:1819 retrans_pkt: SIP TIMER: Rescheduling retransmission #26 (3) SIP/2.0 - 1 [Oct 17 15:03:45] DEBUG[14822]: chan_sip.c:1833 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #26)) Retransmitting #3 (no NAT) to 217.113.77.13:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK10123CE;received=217.113.77.13 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 CSeq: 101 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 261 v=0 o=root 14799 14801 IN IP4 217.113.77.17 s=session c=IN IP4 217.113.77.17 t=0 0 m=image 4906 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:7200 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy --- [Oct 17 15:03:47] DEBUG[14822]: chan_sip.c:1819 retrans_pkt: SIP TIMER: Rescheduling retransmission #27 (3) SIP/2.0 - 1 [Oct 17 15:03:47] DEBUG[14822]: chan_sip.c:1833 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #27)) Retransmitting #3 (no NAT) to 217.113.77.13:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK10123CE;received=217.113.77.13 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 CSeq: 101 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Oct 17 15:03:49] DEBUG[14822]: chan_sip.c:1819 retrans_pkt: SIP TIMER: Rescheduling retransmission #26 (4) SIP/2.0 - 1 [Oct 17 15:03:49] DEBUG[14822]: chan_sip.c:1833 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #26)) Retransmitting #4 (no NAT) to 217.113.77.13:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK10123CE;received=217.113.77.13 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 CSeq: 101 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 261 v=0 o=root 14799 14801 IN IP4 217.113.77.17 s=session c=IN IP4 217.113.77.17 t=0 0 m=image 4906 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:7200 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy --- [Oct 17 15:03:51] DEBUG[14822]: chan_sip.c:1819 retrans_pkt: SIP TIMER: Rescheduling retransmission #27 (4) SIP/2.0 - 1 [Oct 17 15:03:51] DEBUG[14822]: chan_sip.c:1833 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #27)) Retransmitting #4 (no NAT) to 217.113.77.13:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK10123CE;received=217.113.77.13 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 CSeq: 101 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Oct 17 15:03:53] DEBUG[14822]: chan_sip.c:1819 retrans_pkt: SIP TIMER: Rescheduling retransmission #26 (5) SIP/2.0 - 1 [Oct 17 15:03:53] DEBUG[14822]: chan_sip.c:1833 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #26)) Retransmitting #5 (no NAT) to 217.113.77.13:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK10123CE;received=217.113.77.13 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 CSeq: 101 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 261 v=0 o=root 14799 14801 IN IP4 217.113.77.17 s=session c=IN IP4 217.113.77.17 t=0 0 m=image 4906 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:7200 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy --- [Oct 17 15:03:55] DEBUG[14822]: chan_sip.c:1819 retrans_pkt: SIP TIMER: Rescheduling retransmission #27 (5) SIP/2.0 - 1 [Oct 17 15:03:55] DEBUG[14822]: chan_sip.c:1833 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #27)) Retransmitting #5 (no NAT) to 217.113.77.13:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK10123CE;received=217.113.77.13 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 CSeq: 101 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Oct 17 15:03:57] DEBUG[14822]: chan_sip.c:1819 retrans_pkt: SIP TIMER: Rescheduling retransmission #26 (6) SIP/2.0 - 1 [Oct 17 15:03:57] DEBUG[14822]: chan_sip.c:1833 retrans_pkt: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #26)) Retransmitting #6 (no NAT) to 217.113.77.13:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK10123CE;received=217.113.77.13 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 CSeq: 101 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 261 v=0 o=root 14799 14801 IN IP4 217.113.77.17 s=session c=IN IP4 217.113.77.17 t=0 0 m=image 4906 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:7200 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy --- [Oct 17 15:03:59] DEBUG[14822]: chan_sip.c:1819 retrans_pkt: SIP TIMER: Rescheduling retransmission #27 (6) SIP/2.0 - 1 [Oct 17 15:03:59] DEBUG[14822]: chan_sip.c:1833 retrans_pkt: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #27)) Retransmitting #6 (no NAT) to 217.113.77.13:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK10123CE;received=217.113.77.13 From: ;tag=6758DAB2-1B1F To: ;tag=as5b009254 Call-ID: 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 CSeq: 101 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Oct 17 15:04:01] WARNING[14822]: chan_sip.c:1852 retrans_pkt: Maximum retries exceeded on transmission 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 for seqno 101 (Critical Response) [Oct 17 15:04:03] WARNING[14822]: chan_sip.c:1852 retrans_pkt: Maximum retries exceeded on transmission 64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13 for seqno 101 (Critical Response) Really destroying SIP dialog '64A03D1C-710D11D7-82F2B5B0-303B2474@217.113.77.13' Method: BYE