linux-atlantis-be*CLI> linux-atlantis-be*CLI> linux-atlantis-be*CLI> linux-atlantis-be*CLI> linux-atlantis-be*CLI> linux-atlantis-be*CLI> linux-atlantis-be*CLI> linux-atlantis-be*CLI> linux-atlantis-be*CLI> linux-atlantis-be*CLI> <-- SIP read from 217.113.77.13:56062: INVITE sip:3271492041@217.113.77.17:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK381A6C From: ;tag=FABC1342-1424 To: Date: Fri, 28 Mar 2003 17:37:02 GMT Call-ID: B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13 Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 3047958956-1618612695-2156311984-809182324 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: 1048873022 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 218 v=0 o=CiscoSystemsSIP-GW-UserAgent 7174 8387 IN IP4 217.113.77.13 s=SIP Call c=IN IP4 217.113.77.13 t=0 0 m=audio 19004 RTP/AVP 0 19 c=IN IP4 217.113.77.13 a=rtpmap:0 PCMU/8000 a=rtpmap:19 CN/8000 a=ptime:20 [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 0: INVITE sip:3271492041@217.113.77.17:5060 SIP/2.0 (48) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK381A6C (57) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 2: From: ;tag=FABC1342-1424 (48) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 3: To: (34) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 4: Date: Fri, 28 Mar 2003 17:37:02 GMT (35) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 5: Call-ID: B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13 (58) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 6: Supported: 100rel,timer (23) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 7: Min-SE: 1800 (13) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 8: Cisco-Guid: 3047958956-1618612695-2156311984-809182324 (54) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 9: User-Agent: Cisco-SIPGateway/IOS-12.x (37) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 10: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER (104) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 11: CSeq: 101 INVITE (16) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 12: Max-Forwards: 70 (16) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 13: Remote-Party-ID: ;party=calling;screen=no;privacy=off (77) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 14: Timestamp: 1048873022 (21) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 15: Contact: (38) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 16: Expires: 180 (12) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 17: Allow-Events: telephone-event (29) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 18: Content-Type: application/sdp (29) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 19: Content-Length: 218 (19) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 20: (0) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: v=0 (3) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: o=CiscoSystemsSIP-GW-UserAgent 7174 8387 IN IP4 217.113.77.13 (61) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: s=SIP Call (10) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: c=IN IP4 217.113.77.13 (22) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: t=0 0 (5) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: m=audio 19004 RTP/AVP 0 19 (26) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: c=IN IP4 217.113.77.13 (22) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: a=rtpmap:19 CN/8000 (19) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: a=ptime:20 (10) --- (20 headers 10 lines)--- [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4143 sip_alloc: Allocating new SIP dialog for B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13 - INVITE (With RTP) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:14046 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:1618 parse_sip_options: Begin: parsing SIP "Supported: 100rel,timer" [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:1626 parse_sip_options: Found SIP option: -100rel- [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:1632 parse_sip_options: Matched SIP option: 100rel [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:1626 parse_sip_options: Found SIP option: -timer- [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:1632 parse_sip_options: Matched SIP option: timer Sending to 217.113.77.13 : 5060 (no NAT) Using INVITE request as basis request - B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13 Found peer 'cisco' [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:8796 check_user_full: Setting NAT on RTP to Off [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:8804 check_user_full: Setting NAT on UDPTL to Off Found RTP audio format 0 Found RTP audio format 19 Peer audio RTP is at port 217.113.77.13:19004 [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4682 process_sdp: Peer doesn't provide T.38 UDPTL Found description format PCMU for ID 0 Found description format CN for ID 19 Got unsupported a:ptime in SDP offer [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4868 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x2 (CN), combined - 0x0 (nothing) Peer audio RTP is at port 217.113.77.13:19004 [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4938 process_sdp: We're settling with these formats: 0x4 (ulaw) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:12858 handle_request_invite: Checking SIP call limits for device [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:2879 update_call_counter: Updating call counter for incoming call Looking for 3271492041 in internal (domain 217.113.77.17) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:3638 sip_new: *** Our native formats are 0x4 (ulaw) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:3639 sip_new: *** Joint capabilities are 0x4 (ulaw) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:3640 sip_new: *** Our capabilities are 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:3641 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:3664 sip_new: This channel will not be able to handle video. [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:7622 build_route: build_route: Contact hop: list_route: hop: [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:12929 handle_request_invite: SIP/217.113.77.13-081cbff8: 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=z9hG4bK381A6C;received=217.113.77.13 From: ;tag=FABC1342-1424 To: Call-ID: B88447EE-607A11D7-8089B5B0-303B2474@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 --- [Sep 26 12:52:31] DEBUG[9575]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/217.113.77.13-081cbff8 [Sep 26 12:52:31] DEBUG[9557]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 217.113.77.13 [Sep 26 12:52:31] DEBUG[9557]: chan_sip.c:14633 sip_devicestate: Checking device state for peer 217.113.77.13 [Sep 26 12:52:31] DEBUG[9557]: channel.c:880 channel_find_locked: Avoiding initial deadlock for channel '0x81de8b8' [Sep 26 12:52:31] DEBUG[9602]: pbx.c:1684 pbx_extension_helper: Launching 'Dial' -- Executing [3271492041@internal:1] Dial("SIP/217.113.77.13-081cbff8", "SIP/3271492041@172.16.150.100||t") in new stack [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:14691 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4143 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:3638 sip_new: *** Our native formats are 0x4 (ulaw) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:3639 sip_new: *** Joint capabilities are 0x0 (nothing) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:3640 sip_new: *** Our capabilities are 0x4 (ulaw) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:3641 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:3643 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:3664 sip_new: This channel will not be able to handle video. [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:16193 sip_set_rtp_peer: Early remote bridge setting SIP '2b49e67f04246657097e033228f3f953@10.100.20.12' - Sending media to 217.113.77.13 [Sep 26 12:52:31] DEBUG[9602]: rtp.c:1510 ast_rtp_make_compatible: Seeded SDP of 'SIP/172.16.150.100-081df038' with that of 'SIP/217.113.77.13-081cbff8' [Sep 26 12:52:31] DEBUG[9602]: channel.c:3103 ast_channel_inherit_variables: Not copying variable STACK-internal-3271492041-1. [Sep 26 12:52:31] DEBUG[9602]: channel.c:3103 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Sep 26 12:52:31] DEBUG[9602]: channel.c:3103 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Sep 26 12:52:31] DEBUG[9602]: channel.c:3103 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Sep 26 12:52:31] DEBUG[9602]: channel.c:3103 ast_channel_inherit_variables: Not copying variable SIPURI. [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:2718 sip_call: Outgoing Call for 3271492041 [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:2879 update_call_counter: Updating call counter for outgoing call [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:2726 sip_call: Our T38 capability (3856), joint T38 capability (3856) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:5838 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: False [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:5839 add_sdp: ** Our prefcodec: 0x4 (ulaw) Audio is at 10.100.20.12 port 18140 Adding codec 0x4 (ulaw) to SDP [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:5991 add_sdp: -- Done with adding codecs to SDP [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:6030 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4363 parse_request: Header 0: INVITE sip:3271492041@172.16.150.100 SIP/2.0 (44) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4363 parse_request: Header 1: Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3914097c;rport (63) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4363 parse_request: Header 2: From: "7001" ;tag=as2799fb9f (51) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4363 parse_request: Header 3: To: (35) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4363 parse_request: Header 4: Contact: (32) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4363 parse_request: Header 5: Call-ID: 2b49e67f04246657097e033228f3f953@10.100.20.12 (54) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4363 parse_request: Header 6: CSeq: 102 INVITE (16) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4363 parse_request: Header 7: User-Agent: gatewaycomms (24) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4363 parse_request: Header 8: Max-Forwards: 70 (16) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4363 parse_request: Header 9: Date: Tue, 26 Sep 2006 10:52:31 GMT (35) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4363 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4363 parse_request: Header 11: Supported: replaces (19) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4363 parse_request: Header 12: Content-Type: application/sdp (29) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4363 parse_request: Header 13: Content-Length: 172 (19) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4363 parse_request: Header 14: (0) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4395 parse_request: Line: v=0 (3) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4395 parse_request: Line: o=root 9554 9554 IN IP4 217.113.77.13 (37) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4395 parse_request: Line: s=session (9) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4395 parse_request: Line: c=IN IP4 217.113.77.13 (22) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4395 parse_request: Line: t=0 0 (5) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4395 parse_request: Line: m=audio 19004 RTP/AVP 0 (23) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4395 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4395 parse_request: Line: a=silenceSupp:off - - - - (25) [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:4395 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=z9hG4bK3914097c;rport From: "7001" ;tag=as2799fb9f To: Contact: Call-ID: 2b49e67f04246657097e033228f3f953@10.100.20.12 CSeq: 102 INVITEe*CLI> User-Agent: gatewaycomms Max-Forwards: 70 Date: Tue, 26 Sep 2006 10:52:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 172 v=0 o=root 9554 9554 IN IP4 217.113.77.13 s=session c=IN IP4 217.113.77.13 t=0 0 m=audio 19004 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=sendrecv --- [Sep 26 12:52:31] DEBUG[9602]: chan_sip.c:1909 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #30 -- Called 3271492041@172.16.150.100 [Sep 26 12:52:31] DEBUG[9557]: devicestate.c:287 do_state_change: Changing state for SIP/217.113.77.13 - state 2 (In use) [Sep 26 12:52:31] DEBUG[9603]: app_queue.c:535 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. linux-atlantis-be*CLI> <-- SIP read from 172.16.150.100:5060: SIP/2.0 100 Trying Call-ID: 2b49e67f04246657097e033228f3f953@10.100.20.12 From: "7001" ;tag=as2799fb9f To: ;tag=24848 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3914097c;rport Supported: timer,100rel Content-Length: 0 [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 0: SIP/2.0 100 Trying (18) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 1: Call-ID: 2b49e67f04246657097e033228f3f953@10.100.20.12 (54) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 2: From: "7001" ;tag=as2799fb9f (51) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 3: To: ;tag=24848 (45) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 4: CSeq: 102 INVITE (16) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 5: Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3914097c;rport (63) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 6: Supported: timer,100rel (23) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 7: Content-Length: 0 (17) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 8: (0) --- (8 headers 0 lines)--- [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:2045 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #30 - INVITE (got response) [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:2054 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2b49e67f04246657097e033228f3f953@10.100.20.12' Request 102: Found [Sep 26 12:52:31] DEBUG[9575]: chan_sip.c:11214 handle_response_invite: SIP response 100 to standard invite linux-atlantis-be*CLI> <-- SIP read from 172.16.150.100:5060: SIP/2.0 183 Session Progress Call-ID: 2b49e67f04246657097e033228f3f953@10.100.20.12 From: "7001" ;tag=as2799fb9f To: ;tag=24848 Content-Type: application/sdp CSeq: 102 INVITE Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3914097c;rport Supported: timer,100rel Content-Length: 117 v=0 o=MG4000|2.0 56 56 IN IP4 10.100.1.240 s=- c=IN IP4 10.100.1.240 t=0 0 m=audio 20220 RTP/AVP 0 a=ptime:10 [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 1: Call-ID: 2b49e67f04246657097e033228f3f953@10.100.20.12 (54) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 2: From: "7001" ;tag=as2799fb9f (51) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 3: To: ;tag=24848 (45) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 4: Content-Type: application/sdp (29) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 5: CSeq: 102 INVITE (16) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 6: Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3914097c;rport (63) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 7: Supported: timer,100rel (23) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 8: Content-Length: 117 (19) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 9: (0) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: v=0 (3) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: o=MG4000|2.0 56 56 IN IP4 10.100.1.240 (38) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: s=- (3) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: c=IN IP4 10.100.1.240 (21) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: t=0 0 (5) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: m=audio 20220 RTP/AVP 0 (23) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: a=ptime:10 (10) --- (9 headers 7 lines)--- [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:2054 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2b49e67f04246657097e033228f3f953@10.100.20.12' Request 102: Found [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:11214 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 0 Peer audio RTP is at port 10.100.1.240:20220 [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4682 process_sdp: Peer doesn't provide T.38 UDPTL Got unsupported a:ptime in SDP offer [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4868 process_sdp: T38 state changed to 0 on channel SIP/172.16.150.100-081df038 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.100.1.240:20220 [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4938 process_sdp: We're settling with these formats: 0x4 (ulaw) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4945 process_sdp: We have an owner, now see if we need to change this call linux-atlantis-be*CLI> <-- SIP read from 172.16.150.100:5060: SIP/2.0 180 Ringing Call-ID: 2b49e67f04246657097e033228f3f953@10.100.20.12 From: "7001" ;tag=as2799fb9f To: ;tag=24848 Content-Type: application/sdp CSeq: 102 INVITE Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3914097c;rport Supported: timer,100rel Content-Length: 117 v=0 o=MG4000|2.0 56 56 IN IP4 10.100.1.240 s=- c=IN IP4 10.100.1.240 t=0 0 m=audio 20220 RTP/AVP 0 a=ptime:10 [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 1: Call-ID: 2b49e67f04246657097e033228f3f953@10.100.20.12 (54) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 2: From: "7001" ;tag=as2799fb9f (51) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 3: To: ;tag=24848 (45) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 4: Content-Type: application/sdp (29) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 5: CSeq: 102 INVITE (16) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 6: Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3914097c;rport (63) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 7: Supported: timer,100rel (23) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 8: Content-Length: 117 (19) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 9: (0) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: v=0 (3) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: o=MG4000|2.0 56 56 IN IP4 10.100.1.240 (38) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: s=- (3) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: c=IN IP4 10.100.1.240 (21) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: t=0 0 (5) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: m=audio 20220 RTP/AVP 0 (23) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: a=ptime:10 (10) --- (9 headers 7 lines)--- [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:2054 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2b49e67f04246657097e033228f3f953@10.100.20.12' Request 102: Found [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:11214 handle_response_invite: SIP response 180 to standard invite [Sep 26 12:52:35] DEBUG[9575]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/172.16.150.100-081df038 Found RTP audio format 0 Peer audio RTP is at port 10.100.1.240:20220 [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4682 process_sdp: Peer doesn't provide T.38 UDPTL Got unsupported a:ptime in SDP offer [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4868 process_sdp: T38 state changed to 0 on channel SIP/172.16.150.100-081df038 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.100.1.240:20220 [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4938 process_sdp: We're settling with these formats: 0x4 (ulaw) [Sep 26 12:52:35] DEBUG[9575]: chan_sip.c:4945 process_sdp: We have an owner, now see if we need to change this call [Sep 26 12:52:35] DEBUG[9557]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 172.16.150.100 [Sep 26 12:52:35] DEBUG[9557]: chan_sip.c:14633 sip_devicestate: Checking device state for peer 172.16.150.100 [Sep 26 12:52:35] DEBUG[9557]: devicestate.c:287 do_state_change: Changing state for SIP/172.16.150.100 - state 6 (Ringing) -- SIP/172.16.150.100-081df038 is making progress passing it to SIP/217.113.77.13-081cbff8 [Sep 26 12:52:35] DEBUG[9604]: app_queue.c:535 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. [Sep 26 12:52:35] DEBUG[9602]: chan_sip.c:16193 sip_set_rtp_peer: Early remote bridge setting SIP 'B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13' - Sending media to 10.100.1.240 [Sep 26 12:52:35] DEBUG[9602]: rtp.c:1445 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/217.113.77.13-081cbff8' with that of 'SIP/172.16.150.100-081df038' [Sep 26 12:52:35] DEBUG[9602]: chan_sip.c:5838 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Sep 26 12:52:35] DEBUG[9602]: chan_sip.c:5839 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 217.113.77.17 port 12616 Adding codec 0x4 (ulaw) to SDP [Sep 26 12:52:35] DEBUG[9602]: chan_sip.c:5991 add_sdp: -- Done with adding codecs to SDP [Sep 26 12:52:35] DEBUG[9602]: chan_sip.c:6030 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) 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=z9hG4bK381A6C;received=217.113.77.13 From: ;tag=FABC1342-1424 To: ;tag=as3b8aa692 Call-ID: B88447EE-607A11D7-8089B5B0-303B2474@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: 170 v=0 o=root 9554 9554 IN IP4 10.100.1.240 s=session c=IN IP4 10.100.1.240 t=0 0 m=audio 20220 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=sendrecv --- -- SIP/172.16.150.100-081df038 is ringing [Sep 26 12:52:35] DEBUG[9602]: rtp.c:1445 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/217.113.77.13-081cbff8' with that of 'SIP/172.16.150.100-081df038' [Sep 26 12:52:35] DEBUG[9602]: channel.c:2264 ast_indicate_data: Driver for channel 'SIP/217.113.77.13-081cbff8' does not support indication 3, emulating it [Sep 26 12:52:35] DEBUG[9602]: channel.c:2413 ast_prod: Prodding channel 'SIP/217.113.77.13-081cbff8' [Sep 26 12:52:35] DEBUG[9602]: channel.c:2638 set_format: Set channel SIP/217.113.77.13-081cbff8 to write format slin -- SIP/172.16.150.100-081df038 is making progress passing it to SIP/217.113.77.13-081cbff8 [Sep 26 12:52:35] DEBUG[9602]: rtp.c:1445 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/217.113.77.13-081cbff8' with that of 'SIP/172.16.150.100-081df038' linux-atlantis-be*CLI> <-- SIP read from 172.16.150.100:5060: SIP/2.0 200 OK Call-ID: 2b49e67f04246657097e033228f3f953@10.100.20.12 From: "7001" ;tag=as2799fb9f To: ;tag=24848 Content-Type: application/sdp CSeq: 102 INVITE Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3914097c;rport Contact: sip:3271492041@172.16.150.100:5060;user=phone Supported: timer,100rel Content-Length: 117 v=0 o=MG4000|2.0 56 56 IN IP4 10.100.1.240 s=- c=IN IP4 10.100.1.240 t=0 0 m=audio 20220 RTP/AVP 0 a=ptime:10 [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 0: SIP/2.0 200 OK (14) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 1: Call-ID: 2b49e67f04246657097e033228f3f953@10.100.20.12 (54) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 2: From: "7001" ;tag=as2799fb9f (51) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 3: To: ;tag=24848 (45) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 4: Content-Type: application/sdp (29) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 5: CSeq: 102 INVITE (16) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 6: Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK3914097c;rport (63) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 7: Contact: sip:3271492041@172.16.150.100:5060;user=phone (54) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 8: Supported: timer,100rel (23) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 9: Content-Length: 117 (19) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 10: (0) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: v=0 (3) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: o=MG4000|2.0 56 56 IN IP4 10.100.1.240 (38) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: s=- (3) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: c=IN IP4 10.100.1.240 (21) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: t=0 0 (5) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: m=audio 20220 RTP/AVP 0 (23) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: a=ptime:10 (10) --- (10 headers 7 lines)--- [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:1994 __sip_ack: Acked pending invite 102 [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:2012 __sip_ack: Stopping retransmission on '2b49e67f04246657097e033228f3f953@10.100.20.12' of Request 102: Match Not Found [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:11214 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 0 Peer audio RTP is at port 10.100.1.240:20220 [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4682 process_sdp: Peer doesn't provide T.38 UDPTL Got unsupported a:ptime in SDP offer [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4868 process_sdp: T38 state changed to 0 on channel SIP/172.16.150.100-081df038 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.100.1.240:20220 [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4938 process_sdp: We're settling with these formats: 0x4 (ulaw) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4945 process_sdp: We have an owner, now see if we need to change this call [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:2879 update_call_counter: Updating call counter for outgoing call [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:7622 build_route: build_route: Contact hop: sip:3271492041@172.16.150.100:5060;user=phone list_route: hop: [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:5369 reqprep: Strict routing enforced for session 2b49e67f04246657097e033228f3f953@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=z9hG4bK03861910;rport From: "7001" ;tag=as2799fb9f To: ;tag=24848 Contact: Call-ID: 2b49e67f04246657097e033228f3f953@10.100.20.12 CSeq: 102 ACK User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 --- [Sep 26 12:52:36] DEBUG[9602]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/172.16.150.100-081df038 -- SIP/172.16.150.100-081df038 answered SIP/217.113.77.13-081cbff8 [Sep 26 12:52:36] DEBUG[9602]: rtp.c:1445 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/217.113.77.13-081cbff8' with that of 'SIP/172.16.150.100-081df038' [Sep 26 12:52:36] DEBUG[9602]: channel.c:2638 set_format: Set channel SIP/217.113.77.13-081cbff8 to write format ulaw [Sep 26 12:52:36] DEBUG[9602]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/217.113.77.13-081cbff8 [Sep 26 12:52:36] DEBUG[9602]: chan_sip.c:3319 sip_answer: SIP answering channel: SIP/217.113.77.13-081cbff8 [Sep 26 12:52:36] DEBUG[9557]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 172.16.150.100 [Sep 26 12:52:36] DEBUG[9557]: chan_sip.c:14633 sip_devicestate: Checking device state for peer 172.16.150.100 [Sep 26 12:52:36] DEBUG[9557]: devicestate.c:287 do_state_change: Changing state for SIP/172.16.150.100 - state 2 (In use) [Sep 26 12:52:36] DEBUG[9602]: chan_sip.c:5838 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Sep 26 12:52:36] DEBUG[9602]: chan_sip.c:5839 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 217.113.77.17 port 12616 Adding codec 0x4 (ulaw) to SDP [Sep 26 12:52:36] DEBUG[9605]: app_queue.c:535 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. [Sep 26 12:52:36] DEBUG[9557]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 217.113.77.13 [Sep 26 12:52:36] DEBUG[9557]: chan_sip.c:14633 sip_devicestate: Checking device state for peer 217.113.77.13 [Sep 26 12:52:36] DEBUG[9557]: channel.c:880 channel_find_locked: Avoiding initial deadlock for channel '0x81de8b8' [Sep 26 12:52:36] DEBUG[9602]: chan_sip.c:5991 add_sdp: -- Done with adding codecs to SDP [Sep 26 12:52:36] DEBUG[9602]: chan_sip.c:6030 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) 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=z9hG4bK381A6C;received=217.113.77.13 From: ;tag=FABC1342-1424 To: ;tag=as3b8aa692 Call-ID: B88447EE-607A11D7-8089B5B0-303B2474@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: 170 v=0 o=root 9554 9555 IN IP4 10.100.1.240 s=session c=IN IP4 10.100.1.240 t=0 0 m=audio 20220 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=sendrecv --- [Sep 26 12:52:36] DEBUG[9602]: chan_sip.c:1909 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #32 [Sep 26 12:52:36] DEBUG[9557]: devicestate.c:287 do_state_change: Changing state for SIP/217.113.77.13 - state 2 (In use) [Sep 26 12:52:36] DEBUG[9606]: app_queue.c:535 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. linux-atlantis-be*CLI> <-- SIP read from 217.113.77.13:56062: ACK sip:3271492041@217.113.77.17:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK391E81 From: ;tag=FABC1342-1424 To: ;tag=as3b8aa692 Date: Fri, 28 Mar 2003 17:37:02 GMT Call-ID: B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13 Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 0: ACK sip:3271492041@217.113.77.17:5060 SIP/2.0 (45) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK391E81 (57) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 2: From: ;tag=FABC1342-1424 (48) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 3: To: ;tag=as3b8aa692 (49) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 4: Date: Fri, 28 Mar 2003 17:37:02 GMT (35) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 5: Call-ID: B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13 (58) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 7: CSeq: 101 ACK (13) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 8: Content-Length: 0 (17) [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 9: (0) --- (9 headers 0 lines)--- [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:14046 handle_request: **** Received ACK (6) - Command in SIP ACK [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:2002 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #32 [Sep 26 12:52:36] DEBUG[9575]: chan_sip.c:2012 __sip_ack: Stopping retransmission on 'B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13' of Response 101: Match Not Found linux-atlantis-be*CLI> <-- SIP read from 172.16.150.100:5060: INVITE sip:7001@10.100.20.12:5060;user=phone SIP/2.0 Call-ID: 2b49e67f04246657097e033228f3f953@10.100.20.12 From: ;tag=24848 To: "7001" ;tag=as2799fb9f Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.150.100:5060;branch=z9hG4bK-811000000000009d-ac106464-103 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: 105 v=0 o=MG4000|2.0 57 57 IN IP4 10.100.1.240 s=- c=IN IP4 10.100.1.240 t=0 0 m=image 20220 udptl t38 [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 0: INVITE sip:7001@10.100.20.12:5060;user=phone SIP/2.0 (52) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 1: Call-ID: 2b49e67f04246657097e033228f3f953@10.100.20.12 (54) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 2: From: ;tag=24848 (47) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 3: To: "7001" ;tag=as2799fb9f (49) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 4: Content-Type: application/sdp (29) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 5: CSeq: 1 INVITE (14) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 6: Via: SIP/2.0/UDP 172.16.150.100:5060;branch=z9hG4bK-811000000000009d-ac106464-103 (81) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 7: Contact: sip:3271492041@172.16.150.100:5060;user=phone (54) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 8: Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE (41) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 9: Supported: timer,100rel (23) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 10: Max-Forwards: 70 (16) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 11: Content-Length: 105 (19) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 12: (0) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: v=0 (3) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: o=MG4000|2.0 57 57 IN IP4 10.100.1.240 (38) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: s=- (3) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: c=IN IP4 10.100.1.240 (21) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: t=0 0 (5) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: m=image 20220 udptl t38 (23) --- (12 headers 6 lines)--- [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:14046 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:1618 parse_sip_options: Begin: parsing SIP "Supported: timer,100rel" [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:1626 parse_sip_options: Found SIP option: -timer- [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:1632 parse_sip_options: Matched SIP option: timer [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:1626 parse_sip_options: Found SIP option: -100rel- [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:1632 parse_sip_options: Matched SIP option: 100rel Sending to 172.16.150.100 : 5060 (no NAT) Got T.38 offer in SDP in dialog 2b49e67f04246657097e033228f3f953@10.100.20.12 [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4612 process_sdp: T38 state changed to 4 on channel SIP/172.16.150.100-081df038 Peer doesn't provide audio [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4678 process_sdp: Peer T.38 UDPTL is at port 10.100.1.240:20220 [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4861 process_sdp: Our T38 capability = (3856), peer T38 capability (0), joint T38 capability (3856) Capabilities: us - 0x4 (ulaw), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) [Sep 26 12:52:41] NOTICE[9575]: chan_sip.c:4896 process_sdp: No compatible codecs, not accepting this offer! Transmitting (no NAT) to 172.16.150.100:5060: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 172.16.150.100:5060;branch=z9hG4bK-811000000000009d-ac106464-103;received=172.16.150.100 From: ;tag=24848 To: "7001" ;tag=as2799fb9f Call-ID: 2b49e67f04246657097e033228f3f953@10.100.20.12 CSeq: 1 INVITE User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:14257 sipsock_read: SIP message could not be handled, bad request: 2b49e67f04246657097e033228f3f953@10.100.20.12 linux-atlantis-be*CLI> <-- SIP read from 172.16.150.100:5060: ACK sip:7001@10.100.20.12:5060;user=phone SIP/2.0 Call-ID: 2b49e67f04246657097e033228f3f953@10.100.20.12 From: ;tag=24848 To: "7001" ;tag=as2799fb9f CSeq: 1 ACK Via: SIP/2.0/UDP 172.16.150.100:5060;branch=z9hG4bK-811000000000009d-ac106464-103 Max-Forwards: 70 Content-Length: 0 [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 0: ACK sip:7001@10.100.20.12:5060;user=phone SIP/2.0 (49) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 1: Call-ID: 2b49e67f04246657097e033228f3f953@10.100.20.12 (54) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 2: From: ;tag=24848 (47) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 3: To: "7001" ;tag=as2799fb9f (49) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 4: CSeq: 1 ACK (11) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.150.100:5060;branch=z9hG4bK-811000000000009d-ac106464-103 (81) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 7: Content-Length: 0 (17) [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 8: (0) --- (8 headers 0 lines)--- [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:14046 handle_request: **** Received ACK (6) - Command in SIP ACK [Sep 26 12:52:41] DEBUG[9575]: chan_sip.c:2012 __sip_ack: Stopping retransmission on '2b49e67f04246657097e033228f3f953@10.100.20.12' of Response 1: Match Found linux-atlantis-be*CLI> <-- SIP read from 217.113.77.13:56062: INVITE sip:3271492041@217.113.77.17:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK3A584 From: ;tag=FABC1342-1424 To: ;tag=as3b8aa692 Date: Fri, 28 Mar 2003 17:37:29 GMT Call-ID: B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13 Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 3047958956-1618612695-2156311984-809182324 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: 1048873049 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 399 v=0 o=CiscoSystemsSIP-GW-UserAgent 7174 8388 IN IP4 217.113.77.13 s=SIP Call c=IN IP4 217.113.77.13 t=0 0 m=image 19004 udptl t38 c=IN IP4 217.113.77.13 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 0: INVITE sip:3271492041@217.113.77.17:5060 SIP/2.0 (48) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK3A584 (56) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 2: From: ;tag=FABC1342-1424 (48) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 3: To: ;tag=as3b8aa692 (49) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 4: Date: Fri, 28 Mar 2003 17:37:29 GMT (35) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 5: Call-ID: B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13 (58) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 6: Supported: 100rel,timer (23) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 7: Min-SE: 1800 (13) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 8: Cisco-Guid: 3047958956-1618612695-2156311984-809182324 (54) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 9: User-Agent: Cisco-SIPGateway/IOS-12.x (37) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 10: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER (104) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 11: CSeq: 102 INVITE (16) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 12: Max-Forwards: 70 (16) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 13: Remote-Party-ID: ;party=calling;screen=no;privacy=off (77) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 14: Timestamp: 1048873049 (21) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 15: Contact: (38) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 16: Expires: 180 (12) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 17: Allow-Events: telephone-event (29) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 18: Content-Type: application/sdp (29) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 19: Content-Length: 399 (19) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 20: (0) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: v=0 (3) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: o=CiscoSystemsSIP-GW-UserAgent 7174 8388 IN IP4 217.113.77.13 (61) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: s=SIP Call (10) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: c=IN IP4 217.113.77.13 (22) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: t=0 0 (5) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: m=image 19004 udptl t38 (23) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: c=IN IP4 217.113.77.13 (22) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: a=T38FaxVersion:0 (17) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: a=T38MaxBitRate:14400 (21) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: a=T38FaxFillBitRemoval:0 (24) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: a=T38FaxTranscodingMMR:0 (24) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: a=T38FaxTranscodingJBIG:0 (25) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: a=T38FaxRateManagement:transferredTCF (37) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: a=T38FaxMaxBuffer:200 (21) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: a=T38FaxMaxDatagram:72 (22) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4395 parse_request: Line: a=T38FaxUdpEC:t38UDPRedundancy (30) --- (20 headers 16 lines)--- [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:14046 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 B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13 [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4612 process_sdp: T38 state changed to 4 on channel SIP/217.113.77.13-081cbff8 Peer doesn't provide audio [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4678 process_sdp: Peer T.38 UDPTL is at port 217.113.77.13:19004 [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4795 process_sdp: FaxVersion: 0 [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4770 process_sdp: T38MaxBitRate: 14400 [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4811 process_sdp: FillBitRemoval: 0 [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4818 process_sdp: Transcoding MMR: 0 [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4825 process_sdp: Transcoding JBIG: 0 [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4832 process_sdp: RateMangement: transferredTCF [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4765 process_sdp: MaxBufferSize:200 [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4804 process_sdp: FaxMaxDatagram: 72 [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4841 process_sdp: UDP EC: t38UDPRedundancy [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4861 process_sdp: Our T38 capability = (3856), peer T38 capability (16160), joint T38 capability (3872) Capabilities: us - 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) [Sep 26 12:52:59] NOTICE[9575]: chan_sip.c:4896 process_sdp: No compatible codecs, not accepting this offer! Transmitting (no NAT) to 217.113.77.13:5060: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK3A584;received=217.113.77.13 From: ;tag=FABC1342-1424 To: ;tag=as3b8aa692 Call-ID: B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13 CSeq: 102 INVITEe*CLI> User-Agent: gatewaycomms Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:14257 sipsock_read: SIP message could not be handled, bad request: B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13 linux-atlantis-be*CLI> <-- SIP read from 217.113.77.13:56062: ACK sip:3271492041@217.113.77.17:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK3A584 From: ;tag=FABC1342-1424 To: ;tag=as3b8aa692 Date: Fri, 28 Mar 2003 17:37:29 GMT Call-ID: B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13 Max-Forwards: 70 CSeq: 102 ACK Content-Length: 0 [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 0: ACK sip:3271492041@217.113.77.17:5060 SIP/2.0 (45) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK3A584 (56) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 2: From: ;tag=FABC1342-1424 (48) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 3: To: ;tag=as3b8aa692 (49) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 4: Date: Fri, 28 Mar 2003 17:37:29 GMT (35) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 5: Call-ID: B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13 (58) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 6: Max-Forwards: 70 (16) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 7: CSeq: 102 ACK (13) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 8: Content-Length: 0 (17) [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 9: (0) --- (9 headers 0 lines)--- [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:14046 handle_request: **** Received ACK (6) - Command in SIP ACK [Sep 26 12:52:59] DEBUG[9575]: chan_sip.c:2012 __sip_ack: Stopping retransmission on 'B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13' of Response 102: Match Found linux-atlantis-be*CLI> <-- SIP read from 217.113.77.13:56062: BYE sip:3271492041@217.113.77.17:5060 SIP/2.0 Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK3BAA7 From: ;tag=FABC1342-1424 To: ;tag=as3b8aa692 Date: Fri, 28 Mar 2003 17:37:29 GMT Call-ID: B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 Timestamp: 1048873051 CSeq: 103 BYE Reason: Q.850;cause=16 Content-Length: 0 [Sep 26 12:53:00] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 0: BYE sip:3271492041@217.113.77.17:5060 SIP/2.0 (45) [Sep 26 12:53:00] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 1: Via: SIP/2.0/UDP 217.113.77.13:5060;branch=z9hG4bK3BAA7 (56) [Sep 26 12:53:00] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 2: From: ;tag=FABC1342-1424 (48) [Sep 26 12:53:00] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 3: To: ;tag=as3b8aa692 (49) [Sep 26 12:53:00] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 4: Date: Fri, 28 Mar 2003 17:37:29 GMT (35) [Sep 26 12:53:00] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 5: Call-ID: B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13 (58) [Sep 26 12:53:00] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 6: User-Agent: Cisco-SIPGateway/IOS-12.x (37) [Sep 26 12:53:00] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 7: Max-Forwards: 70 (16) [Sep 26 12:53:00] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 8: Timestamp: 1048873051 (21) [Sep 26 12:53:00] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 9: CSeq: 103 BYE (13) [Sep 26 12:53:00] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 10: Reason: Q.850;cause=16 (22) [Sep 26 12:53:00] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 11: Content-Length: 0 (17) [Sep 26 12:53:00] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 12: (0) --- (12 headers 0 lines)--- [Sep 26 12:53:00] DEBUG[9575]: chan_sip.c:14046 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 217.113.77.13 : 5060 (no NAT) 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=z9hG4bK3BAA7;received=217.113.77.13 From: ;tag=FABC1342-1424 To: ;tag=as3b8aa692 Call-ID: B88447EE-607A11D7-8089B5B0-303B2474@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 --- [Sep 26 12:53:00] DEBUG[9602]: channel.c:3573 ast_generic_bridge: Didn't get a frame from channel: SIP/217.113.77.13-081cbff8 [Sep 26 12:53:00] DEBUG[9602]: channel.c:3872 ast_channel_bridge: Bridge stops bridging channels SIP/217.113.77.13-081cbff8 and SIP/172.16.150.100-081df038 [Sep 26 12:53:00] DEBUG[9602]: channel.c:1543 ast_hangup: Hanging up channel 'SIP/172.16.150.100-081df038' [Sep 26 12:53:00] DEBUG[9602]: chan_sip.c:3180 sip_hangup: Hangup call SIP/172.16.150.100-081df038, SIP callid 2b49e67f04246657097e033228f3f953@10.100.20.12) [Sep 26 12:53:00] DEBUG[9602]: chan_sip.c:3188 sip_hangup: update_call_counter(3271492041) - decrement call limit counter on hangup [Sep 26 12:53:00] DEBUG[9602]: chan_sip.c:2879 update_call_counter: Updating call counter for outgoing call [Sep 26 12:53:00] DEBUG[9602]: chan_sip.c:5369 reqprep: Strict routing enforced for session 2b49e67f04246657097e033228f3f953@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=z9hG4bK2842d262;rport From: "7001" ;tag=as2799fb9f To: ;tag=24848 Contact: Call-ID: 2b49e67f04246657097e033228f3f953@10.100.20.12 CSeq: 103 BYE User-Agent: gatewaycomms Max-Forwards: 70 Content-Length: 0 linux-atlantis-be*CLI> --- [Sep 26 12:53:00] DEBUG[9602]: chan_sip.c:1909 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #33 [Sep 26 12:53:00] DEBUG[9602]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/172.16.150.100-081df038 [Sep 26 12:53:00] DEBUG[9602]: rtp.c:1408 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Sep 26 12:53:00] DEBUG[9602]: app_dial.c:1630 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Sep 26 12:53:00] DEBUG[9602]: pbx.c:2280 __ast_pbx_run: Spawn extension (internal,3271492041,1) exited non-zero on 'SIP/217.113.77.13-081cbff8' [Sep 26 12:53:00] DEBUG[9557]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 172.16.150.100 [Sep 26 12:53:00] DEBUG[9602]: pbx.c:1538 pbx_substitute_variables_helper_full: Function result is '7001' [Sep 26 12:53:00] DEBUG[9602]: pbx.c:1538 pbx_substitute_variables_helper_full: Function result is '7001' [Sep 26 12:53:00] DEBUG[9557]: chan_sip.c:14633 sip_devicestate: Checking device state for peer 172.16.150.100 [Sep 26 12:53:00] DEBUG[9602]: pbx.c:1538 pbx_substitute_variables_helper_full: Function result is '3271492041' [Sep 26 12:53:00] DEBUG[9602]: pbx.c:1538 pbx_substitute_variables_helper_full: Function result is 'internal' [Sep 26 12:53:00] DEBUG[9602]: pbx.c:1538 pbx_substitute_variables_helper_full: Function result is 'SIP/217.113.77.13-081cbff8' [Sep 26 12:53:00] DEBUG[9602]: pbx.c:1538 pbx_substitute_variables_helper_full: Function result is 'SIP/172.16.150.100-081df038' [Sep 26 12:53:00] DEBUG[9557]: devicestate.c:287 do_state_change: Changing state for SIP/172.16.150.100 - state 1 (Not in use) [Sep 26 12:53:00] DEBUG[9602]: pbx.c:1538 pbx_substitute_variables_helper_full: Function result is 'Dial' [Sep 26 12:53:00] DEBUG[9602]: pbx.c:1538 pbx_substitute_variables_helper_full: Function result is 'SIP/3271492041@172.16.150.100||t' [Sep 26 12:53:00] DEBUG[9607]: app_queue.c:535 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. [Sep 26 12:53:00] DEBUG[9602]: pbx.c:1538 pbx_substitute_variables_helper_full: Function result is '2006-09-26 12:52:31' [Sep 26 12:53:00] DEBUG[9602]: pbx.c:1538 pbx_substitute_variables_helper_full: Function result is '2006-09-26 12:52:36' [Sep 26 12:53:00] DEBUG[9602]: pbx.c:1538 pbx_substitute_variables_helper_full: Function result is '2006-09-26 12:53:00' [Sep 26 12:53:00] DEBUG[9602]: pbx.c:1538 pbx_substitute_variables_helper_full: Function result is '29' [Sep 26 12:53:00] DEBUG[9602]: pbx.c:1538 pbx_substitute_variables_helper_full: Function result is '24' [Sep 26 12:53:00] DEBUG[9602]: pbx.c:1538 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Sep 26 12:53:00] DEBUG[9602]: pbx.c:1538 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Sep 26 12:53:00] DEBUG[9602]: pbx.c:1538 pbx_substitute_variables_helper_full: Function result is '' [Sep 26 12:53:00] DEBUG[9602]: pbx.c:1538 pbx_substitute_variables_helper_full: Function result is '1159267951.4' [Sep 26 12:53:00] DEBUG[9602]: pbx.c:1538 pbx_substitute_variables_helper_full: Function result is '' [Sep 26 12:53:00] DEBUG[9602]: channel.c:1543 ast_hangup: Hanging up channel 'SIP/217.113.77.13-081cbff8' [Sep 26 12:53:00] DEBUG[9602]: chan_sip.c:3180 sip_hangup: Hangup call SIP/217.113.77.13-081cbff8, SIP callid B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13) [Sep 26 12:53:00] DEBUG[9602]: chan_sip.c:3188 sip_hangup: update_call_counter() - decrement call limit counter on hangup [Sep 26 12:53:00] DEBUG[9602]: chan_sip.c:2879 update_call_counter: Updating call counter for incoming call [Sep 26 12:53:00] DEBUG[9602]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/217.113.77.13-081cbff8 [Sep 26 12:53:00] DEBUG[9557]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 217.113.77.13 [Sep 26 12:53:00] DEBUG[9557]: chan_sip.c:14633 sip_devicestate: Checking device state for peer 217.113.77.13 [Sep 26 12:53:00] DEBUG[9557]: devicestate.c:287 do_state_change: Changing state for SIP/217.113.77.13 - state 1 (Not in use) [Sep 26 12:53:00] DEBUG[9608]: app_queue.c:535 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. linux-atlantis-be*CLI> <-- SIP read from 172.16.150.100:5060: SIP/2.0 200 OK Call-ID: 2b49e67f04246657097e033228f3f953@10.100.20.12 From: "7001" ;tag=as2799fb9f To: ;tag=24848 CSeq: 103 BYE Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK2842d262;rport Contact: sip:3271492041@172.16.150.100:5060;user=phone Supported: timer,100rel Content-Length: 0 [Sep 26 12:53:01] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 0: SIP/2.0 200 OK (14) [Sep 26 12:53:01] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 1: Call-ID: 2b49e67f04246657097e033228f3f953@10.100.20.12 (54) [Sep 26 12:53:01] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 2: From: "7001" ;tag=as2799fb9f (51) [Sep 26 12:53:01] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 3: To: ;tag=24848 (45) [Sep 26 12:53:01] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 4: CSeq: 103 BYE (13) [Sep 26 12:53:01] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 5: Via: SIP/2.0/UDP 10.100.20.12:5060;branch=z9hG4bK2842d262;rport (63) [Sep 26 12:53:01] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 6: Contact: sip:3271492041@172.16.150.100:5060;user=phone (54) [Sep 26 12:53:01] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 7: Supported: timer,100rel (23) [Sep 26 12:53:01] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 8: Content-Length: 0 (17) [Sep 26 12:53:01] DEBUG[9575]: chan_sip.c:4363 parse_request: Header 9: (0) --- (9 headers 0 lines)--- [Sep 26 12:53:01] DEBUG[9575]: chan_sip.c:2002 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #33 [Sep 26 12:53:01] DEBUG[9575]: chan_sip.c:2012 __sip_ack: Stopping retransmission on '2b49e67f04246657097e033228f3f953@10.100.20.12' of Request 103: Match Not Found Really destroying SIP dialog '2b49e67f04246657097e033228f3f953@10.100.20.12' Method: ACK Really destroying SIP dialog 'B88447EE-607A11D7-8089B5B0-303B2474@217.113.77.13' Method: BYE linux-atlantis-be*CLI>