<--- SIP read from 10.50.10.2:5060 ---> INVITE sip:8055480617@10.50.10.171:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.10.2:5060;rport;branch=z9hG4bK-dabb4498cc02b83fc0386cacbba8e2dd-metaswitch-1 Allow-Events: message-summary Allow-Events: refer Allow-Events: dialog Allow-Events: line-seize Max-Forwards: 70 Call-ID: C371B383@metaswitch From: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 To: CSeq: 869046971 INVITE Expires: 180 Organization: Supported: 100rel Content-Length: 122 Content-Type: application/sdp Contact: WIRELESS CALLER ;isup-oli=62 v=0 o=- 2600422519 2600422519 IN IP4 10.50.10.13 s=- c=IN IP4 10.50.10.13 t=0 0 m=audio 33758 RTP/AVP 0 a=ptime:20 <-------------> [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: INVITE sip:8055480617@10.50.10.171:5060;transport=udp SIP/2.0 (61) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.10.2:5060;rport;branch=z9hG4bK-dabb4498cc02b83fc0386cacbba8e2dd-metaswitch-1 (99) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: Allow-Events: message-summary (29) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: Allow-Events: refer (19) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: Allow-Events: dialog (20) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Allow-Events: line-seize (24) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Max-Forwards: 70 (16) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Call-ID: C371B383@metaswitch (28) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: From: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 (99) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: To: (33) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: CSeq: 869046971 INVITE (22) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: Expires: 180 (12) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 12: Organization: (14) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 13: Supported: 100rel (17) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 14: Content-Length: 122 (19) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 15: Content-Type: application/sdp (29) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 16: Contact: WIRELESS CALLER ;isup-oli=62 (69) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 17: (0) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: v=0 (3) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: o=- 2600422519 2600422519 IN IP4 10.50.10.13 (44) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: s=- (3) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: c=IN IP4 10.50.10.13 (20) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: t=0 0 (5) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: m=audio 33758 RTP/AVP 0 (23) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=ptime:20 (10) --- (17 headers 7 lines) --- [May 17 10:02:39] DEBUG[7809]: chan_sip.c:2576 do_setnat: Setting NAT on RTP to Off [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4312 sip_alloc: Allocating new SIP dialog for C371B383@metaswitch - INVITE (With RTP) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:14709 handle_request: **** Received INVITE (5) - Command in SIP INVITE [May 17 10:02:39] DEBUG[7809]: chan_sip.c:1681 parse_sip_options: Begin: parsing SIP "Supported: 100rel" [May 17 10:02:39] DEBUG[7809]: chan_sip.c:1689 parse_sip_options: Found SIP option: -100rel- [May 17 10:02:39] DEBUG[7809]: chan_sip.c:1695 parse_sip_options: Matched SIP option: 100rel Sending to 10.50.10.2 : 5060 (NAT) Using INVITE request as basis request - C371B383@metaswitch Found peer '8055488091' [May 17 10:02:39] DEBUG[7809]: chan_sip.c:2576 do_setnat: Setting NAT on RTP to Off Found RTP audio format 0 Peer audio RTP is at port 10.50.10.13:33758 [May 17 10:02:39] DEBUG[7809]: chan_sip.c:5136 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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 10.50.10.13:33758 [May 17 10:02:39] DEBUG[7809]: chan_sip.c:5216 process_sdp: We're settling with these formats: 0x4 (ulaw) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:13418 handle_request_invite: Checking SIP call limits for device 8055488091 [May 17 10:02:39] DEBUG[7809]: chan_sip.c:3004 update_call_counter: Updating call counter for incoming call Looking for 8055480617 in sip (domain 10.50.10.171) [May 17 10:02:39] DEBUG[7809]: frame.c:1267 ast_codec_choose: Could not find preferred codec - Going for the best codec [May 17 10:02:39] DEBUG[7809]: chan_sip.c:3806 sip_new: *** Our native formats are 0x4 (ulaw) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:3807 sip_new: *** Joint capabilities are 0x4 (ulaw) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:3808 sip_new: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263) [May 17 10:02:39] DEBUG[7809]: frame.c:1267 ast_codec_choose: Could not find preferred codec - Going for the best codec [May 17 10:02:39] DEBUG[7809]: chan_sip.c:3809 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:3832 sip_new: This channel will not be able to handle video. [May 17 10:02:39] DEBUG[7809]: chan_sip.c:7994 build_route: build_route: Contact hop: WIRELESS CALLER ;isup-oli=62 list_route: hop: [May 17 10:02:39] DEBUG[7809]: chan_sip.c:13492 handle_request_invite: SIP/8055488091-08d62b90: New call is still down.... Trying... <--- Transmitting (no NAT) to 10.50.10.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.50.10.2:5060;rport;branch=z9hG4bK-dabb4498cc02b83fc0386cacbba8e2dd-metaswitch-1;received=10.50.10.2 From: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 To: Call-ID: C371B383@metaswitch CSeq: 869046971 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [May 17 10:02:39] DEBUG[7809]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/8055488091-08d62b90 [May 17 10:02:39] DEBUG[7787]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 8055488091 [May 17 10:02:39] DEBUG[7787]: chan_sip.c:15321 sip_devicestate: Checking device state for peer 8055488091 [May 17 10:02:39] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SIP/8055488091 - state 1 (Not in use) [May 17 10:02:39] DEBUG[7875]: pbx.c:1809 pbx_extension_helper: Launching 'Goto' -- Executing [8055480617@sip:1] Goto("SIP/8055488091-08d62b90", "line2|8055488092|1") in new stack -- Goto (line2,8055488092,1) [May 17 10:02:39] DEBUG[7875]: pbx.c:1809 pbx_extension_helper: Launching 'Answer' -- Executing [8055488092@line2:1] Answer("SIP/8055488091-08d62b90", "") in new stack [May 17 10:02:39] DEBUG[7875]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/8055488091-08d62b90 [May 17 10:02:39] DEBUG[7875]: chan_sip.c:3464 sip_answer: SIP answering channel: SIP/8055488091-08d62b90 [May 17 10:02:39] DEBUG[7875]: chan_sip.c:6432 transmit_response_with_sdp: Setting framing from config on incoming call [May 17 10:02:39] DEBUG[7875]: chan_sip.c:6198 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [May 17 10:02:39] DEBUG[7875]: chan_sip.c:6199 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.50.10.171 port 13754 Adding codec 0x4 (ulaw) to SDP [May 17 10:02:39] DEBUG[7875]: chan_sip.c:6330 add_sdp: -- Done with adding codecs to SDP [May 17 10:02:39] DEBUG[7875]: chan_sip.c:6375 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) <--- Reliably Transmitting (no NAT) to 10.50.10.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.10.2:5060;rport;branch=z9hG4bK-dabb4498cc02b83fc0386cacbba8e2dd-metaswitch-1;received=10.50.10.2 From: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 To: ;tag=as48f37b45 Call-ID: C371B383@metaswitch CSeq: 869046971 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 182 v=0 o=root 7783 7783 IN IP4 10.50.10.171 s=session c=IN IP4 10.50.10.171 t=0 0 m=audio 13754 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [May 17 10:02:39] DEBUG[7875]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #108 [May 17 10:02:39] DEBUG[7875]: pbx.c:1809 pbx_extension_helper: Launching 'SLATrunk' -- Executing [8055488092@line2:2] SLATrunk("SIP/8055488091-08d62b90", "line2") in new stack [May 17 10:02:39] DEBUG[7875]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SLA:station1_line2 [May 17 10:02:39] DEBUG[7875]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SLA:station2_line2 [May 17 10:02:39] DEBUG[7875]: chan_zap.c:7817 zt_request: Using channel -2 [May 17 10:02:39] DEBUG[7875]: channel.c:2911 set_format: Set channel Zap/pseudo-2087517722 to read format slin [May 17 10:02:39] DEBUG[7875]: channel.c:2911 set_format: Set channel Zap/pseudo-2087517722 to write format slin -- Created MeetMe conference 1023 for conference 'SLA_line2' [May 17 10:02:39] DEBUG[7875]: channel.c:2501 ast_indicate_data: Driver for channel 'SIP/8055488091-08d62b90' does not support indication 3, emulating it [May 17 10:02:39] DEBUG[7875]: channel.c:2911 set_format: Set channel SIP/8055488091-08d62b90 to write format slin [May 17 10:02:39] DEBUG[7875]: channel.c:2030 ast_settimeout: Scheduling timer at 160 sample intervals [May 17 10:02:39] DEBUG[7875]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel meetme:SLA_line2 [May 17 10:02:39] DEBUG[7875]: channel.c:2911 set_format: Set channel SIP/8055488091-08d62b90 to write format ulaw [May 17 10:02:39] DEBUG[7875]: channel.c:2030 ast_settimeout: Scheduling timer at 0 sample intervals [May 17 10:02:39] DEBUG[7875]: channel.c:2911 set_format: Set channel SIP/8055488091-08d62b90 to write format slin [May 17 10:02:39] DEBUG[7875]: channel.c:2911 set_format: Set channel SIP/8055488091-08d62b90 to read format slin [May 17 10:02:39] DEBUG[7875]: app_meetme.c:1641 conf_run: <--- SIP read from 10.50.10.2:5060 ---> ACK sip:8055480617@10.50.10.171 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.2:5060;branch=z9hG4bK-a8c6544eaacec82b2084b8fcf483d78b-metaswitch-1 Allow-Events: message-summary Allow-Events: refer Allow-Events: dialog Allow-Events: line-seize Max-Forwards: 70 Call-ID: C371B383@metaswitch From: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 To: ;tag=as48f37b45 CSeq: 869046971 ACK Contact: WIRELESS CALLER ;isup-oli=62 Organization: Content-Length: 0 <-------------> Placed channel SIP/8055488091-08d62b90 in ZAP conf 1023 [May 17 10:02:39] DEBUG[7787]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 8055488091 [May 17 10:02:39] DEBUG[7787]: chan_sip.c:15321 sip_devicestate: Checking device state for peer 8055488091 [May 17 10:02:39] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SIP/8055488091 - state 1 (Not in use) [May 17 10:02:39] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station1_line2 [May 17 10:02:39] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SLA:station1_line2 - state 6 (Ringing) [May 17 10:02:39] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station1_line2 [May 17 10:02:39] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station1_line2 Reliably Transmitting (no NAT) to 10.50.10.177:5060: NOTIFY sip:station1@10.50.10.177 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK4a812ab3;rport From: Line 2 ;tag=as364fefba To: Trunk02 ;tag=676fadec36d2598 Contact: Call-ID: 8656389e4098c6e1ef9676ad4129fce6@10.50.10.177 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 247

early

--- [May 17 10:02:39] DEBUG[7787]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #110 Extension Changed station1_line2 new state Ringing for Notify User station1 [May 17 10:02:39] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station2_line2 [May 17 10:02:39] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SLA:station2_line2 - state 6 (Ringing) [May 17 10:02:39] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station2_line2 [May 17 10:02:39] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station2_line2 Reliably Transmitting (no NAT) to 10.50.10.183:5060: NOTIFY sip:station2@10.50.10.183 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK16e422b1;rport From: sip:station2_line2@10.50.10.171:5060;tag=as73da2e01 To: Trunk02 ;tag=3e52f55d7f04b26 Contact: Call-ID: 0fbce7d09c3594aa4ae1eb6724f58925@10.50.10.183 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 247

early

--- [May 17 10:02:39] DEBUG[7787]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #111 Extension Changed station2_line2 new state Ringing for Notify User station2 [May 17 10:02:39] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "meetme" - number: SLA_line2 [May 17 10:02:39] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for meetme:SLA_line2 - state 2 (In use) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:15387 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4312 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:2576 do_setnat: Setting NAT on RTP to Off [May 17 10:02:39] DEBUG[7803]: chan_sip.c:3806 sip_new: *** Our native formats are 0x4 (ulaw) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: ACK sip:8055480617@10.50.10.171 SIP/2.0 (39) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.10.2:5060;branch=z9hG4bK-a8c6544eaacec82b2084b8fcf483d78b-metaswitch-1 (93) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: Allow-Events: message-summary (29) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: Allow-Events: refer (19) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: Allow-Events: dialog (20) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Allow-Events: line-seize (24) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Max-Forwards: 70 (16) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Call-ID: C371B383@metaswitch (28) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: From: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 (99) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: To: ;tag=as48f37b45 (48) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: CSeq: 869046971 ACK (19) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: Contact: WIRELESS CALLER ;isup-oli=62 (69) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 12: Organization: (14) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 13: Content-Length: 0 (17) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 14: (0) --- (14 headers 0 lines) --- [May 17 10:02:39] DEBUG[7809]: chan_sip.c:14709 handle_request: **** Received ACK (6) - Command in SIP ACK [May 17 10:02:39] DEBUG[7809]: chan_sip.c:2080 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #108 [May 17 10:02:39] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on 'C371B383@metaswitch' of Response 869046971: Match Not Found [May 17 10:02:39] DEBUG[7803]: chan_sip.c:3807 sip_new: *** Joint capabilities are 0x0 (nothing) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:3808 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:3809 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:3811 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:3832 sip_new: This channel will not be able to handle video. [May 17 10:02:39] DEBUG[7803]: channel.c:3376 ast_channel_inherit_variables: Not copying variable STACK-line2-8055488092-2. [May 17 10:02:39] DEBUG[7803]: channel.c:3376 ast_channel_inherit_variables: Not copying variable STACK-line2-8055488092-1. [May 17 10:02:39] DEBUG[7803]: channel.c:3376 ast_channel_inherit_variables: Not copying variable STACK-sip-8055480617-1. [May 17 10:02:39] DEBUG[7803]: channel.c:3376 ast_channel_inherit_variables: Not copying variable SIPCALLID. [May 17 10:02:39] DEBUG[7803]: channel.c:3376 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [May 17 10:02:39] DEBUG[7803]: channel.c:3376 ast_channel_inherit_variables: Not copying variable SIPURI. [May 17 10:02:39] DEBUG[7803]: chan_sip.c:2831 sip_call: Outgoing Call for station1 [May 17 10:02:39] DEBUG[7803]: chan_sip.c:3004 update_call_counter: Updating call counter for outgoing call [May 17 10:02:39] DEBUG[7803]: chan_sip.c:2846 sip_call: Our T38 capability (0), joint T38 capability (0) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:6198 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: False [May 17 10:02:39] DEBUG[7876]: app_queue.c:546 changethread: Device 'SIP/8055488091' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 17 10:02:39] DEBUG[7877]: app_queue.c:546 changethread: Device 'SIP/8055488091' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 17 10:02:39] DEBUG[7803]: chan_sip.c:6199 add_sdp: ** Our prefcodec: 0x4 (ulaw) Audio is at 10.50.10.171 port 16270 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP [May 17 10:02:39] DEBUG[7803]: chan_sip.c:6330 add_sdp: -- Done with adding codecs to SDP [May 17 10:02:39] DEBUG[7803]: chan_sip.c:6375 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 0: INVITE sip:station1@10.50.10.177 SIP/2.0 (40) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK4860bebb;rport (63) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 2: From: "WIRELESS CALLER" ;tag=as3f73485e (68) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 3: To: (31) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 4: Contact: (38) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 5: Call-ID: 28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171 (54) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 6: CSeq: 102 INVITE (16) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 7: User-Agent: Asterisk PBX (24) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 8: Max-Forwards: 70 (16) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 9: Date: Thu, 17 May 2007 17:02:39 GMT (35) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 11: Supported: replaces (19) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 12: Content-Type: application/sdp (29) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 13: Content-Length: 285 (19) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 14: (0) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: v=0 (3) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: o=root 7783 7783 IN IP4 10.50.10.171 (36) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: s=session (9) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: c=IN IP4 10.50.10.171 (21) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: t=0 0 (5) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: m=audio 16270 RTP/AVP 0 8 3 101 (31) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: a=fmtp:101 0-16 (15) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: a=silenceSupp:off - - - - (25) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: a=ptime:20 (10) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 10.50.10.177:5060: INVITE sip:station1@10.50.10.177 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK4860bebb;rport From: "WIRELESS CALLER" ;tag=as3f73485e To: Contact: Call-ID: 28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 17 May 2007 17:02:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 7783 7783 IN IP4 10.50.10.171 s=session c=IN IP4 10.50.10.171 t=0 0 m=audio 16270 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [May 17 10:02:39] DEBUG[7875]: rtp.c:2701 ast_rtp_write: Ooh, format changed from unknown to ulaw [May 17 10:02:39] DEBUG[7875]: rtp.c:2718 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [May 17 10:02:39] DEBUG[7803]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #112 -- Called station1 [May 17 10:02:39] DEBUG[7803]: chan_sip.c:15387 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4312 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:2576 do_setnat: Setting NAT on RTP to Off [May 17 10:02:39] DEBUG[7803]: chan_sip.c:3806 sip_new: *** Our native formats are 0x4 (ulaw) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:3807 sip_new: *** Joint capabilities are 0x0 (nothing) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:3808 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:3809 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:3811 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:3832 sip_new: This channel will not be able to handle video. [May 17 10:02:39] DEBUG[7803]: channel.c:3376 ast_channel_inherit_variables: Not copying variable STACK-line2-8055488092-2. [May 17 10:02:39] DEBUG[7803]: channel.c:3376 ast_channel_inherit_variables: Not copying variable STACK-line2-8055488092-1. [May 17 10:02:39] DEBUG[7803]: channel.c:3376 ast_channel_inherit_variables: Not copying variable STACK-sip-8055480617-1. [May 17 10:02:39] DEBUG[7803]: channel.c:3376 ast_channel_inherit_variables: Not copying variable SIPCALLID. [May 17 10:02:39] DEBUG[7803]: channel.c:3376 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [May 17 10:02:39] DEBUG[7803]: channel.c:3376 ast_channel_inherit_variables: Not copying variable SIPURI. [May 17 10:02:39] DEBUG[7803]: chan_sip.c:2831 sip_call: Outgoing Call for station2 [May 17 10:02:39] DEBUG[7803]: chan_sip.c:3004 update_call_counter: Updating call counter for outgoing call [May 17 10:02:39] DEBUG[7803]: chan_sip.c:2846 sip_call: Our T38 capability (0), joint T38 capability (0) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:6198 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: False [May 17 10:02:39] DEBUG[7803]: chan_sip.c:6199 add_sdp: ** Our prefcodec: 0x4 (ulaw) Audio is at 10.50.10.171 port 19730 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP [May 17 10:02:39] DEBUG[7803]: chan_sip.c:6330 add_sdp: -- Done with adding codecs to SDP [May 17 10:02:39] DEBUG[7803]: chan_sip.c:6375 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 0: INVITE sip:station2@10.50.10.183 SIP/2.0 (40) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK581f07ea;rport (63) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 2: From: "WIRELESS CALLER" ;tag=as31d30179 (68) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 3: To: (31) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 4: Contact: (38) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 5: Call-ID: 6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171 (54) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 6: CSeq: 102 INVITE (16) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 7: User-Agent: Asterisk PBX (24) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 8: Max-Forwards: 70 (16) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 9: Date: Thu, 17 May 2007 17:02:39 GMT (35) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 11: Supported: replaces (19) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 12: Content-Type: application/sdp (29) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 13: Content-Length: 285 (19) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4577 parse_request: Header 14: (0) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: v=0 (3) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: o=root 7783 7783 IN IP4 10.50.10.171 (36) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: s=session (9) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: c=IN IP4 10.50.10.171 (21) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: t=0 0 (5) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: m=audio 19730 RTP/AVP 0 8 3 101 (31) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: a=fmtp:101 0-16 (15) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: a=silenceSupp:off - - - - (25) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: a=ptime:20 (10) [May 17 10:02:39] DEBUG[7803]: chan_sip.c:4609 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 10.50.10.183:5060: INVITE sip:station2@10.50.10.183 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK581f07ea;rport From: "WIRELESS CALLER" ;tag=as31d30179 To: Contact: Call-ID: 6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 17 May 2007 17:02:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 7783 7783 IN IP4 10.50.10.171 s=session c=IN IP4 10.50.10.171 t=0 0 m=audio 19730 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [May 17 10:02:39] DEBUG[7803]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #114 -- Called station2 <--- SIP read from 10.50.10.177:5060 ---> SIP/2.0 200 OK Call-ID: 8656389e4098c6e1ef9676ad4129fce6@10.50.10.177 CSeq: 103 NOTIFY From: Line 2 ;tag=as364fefba To: Trunk02 ;tag=676fadec36d2598 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK4a812ab3;rport Content-Length: 0 Contact: Trunk02 Supported: replaces User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 200 OK (14) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 8656389e4098c6e1ef9676ad4129fce6@10.50.10.177 (54) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 103 NOTIFY (16) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: Line 2 ;tag=as364fefba (66) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: Trunk02 ;tag=676fadec36d2598 (59) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK4a812ab3;rport (63) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Contact: Trunk02 (44) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Supported: replaces (19) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [May 17 10:02:39] DEBUG[7809]: chan_sip.c:2080 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #110 [May 17 10:02:39] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '8656389e4098c6e1ef9676ad4129fce6@10.50.10.177' of Request 103: Match Not Found SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.50.10.177:5060 ---> SIP/2.0 100 Trying Call-ID: 28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171 CSeq: 102 INVITE From: "WIRELESS CALLER" ;tag=as3f73485e To: ;tag=eebbd6a54ff83f3 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK4860bebb;rport Content-Length: 0 User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 100 Trying (18) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171 (54) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 102 INVITE (16) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: "WIRELESS CALLER" ;tag=as3f73485e (68) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: ;tag=eebbd6a54ff83f3 (51) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK4860bebb;rport (63) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [May 17 10:02:39] DEBUG[7809]: chan_sip.c:2123 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #112 - INVITE (got response) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:2132 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171' Request 102: Found [May 17 10:02:39] DEBUG[7809]: chan_sip.c:11658 handle_response_invite: SIP response 100 to standard invite <--- SIP read from 10.50.10.183:5060 ---> SIP/2.0 200 OK Call-ID: 0fbce7d09c3594aa4ae1eb6724f58925@10.50.10.183 CSeq: 103 NOTIFY From: sip:station2_line2@10.50.10.171:5060;tag=as73da2e01 To: Trunk02 ;tag=3e52f55d7f04b26 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK16e422b1;rport Content-Length: 0 Contact: Trunk02 Supported: replaces User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 <-------------> [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 200 OK (14) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 0fbce7d09c3594aa4ae1eb6724f58925@10.50.10.183 (54) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 103 NOTIFY (16) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: sip:station2_line2@10.50.10.171:5060;tag=as73da2e01 (57) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: Trunk02 ;tag=3e52f55d7f04b26 (59) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK16e422b1;rport (63) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Contact: Trunk02 (44) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Supported: replaces (19) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 (70) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [May 17 10:02:39] DEBUG[7809]: chan_sip.c:2080 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #111 [May 17 10:02:39] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '0fbce7d09c3594aa4ae1eb6724f58925@10.50.10.183' of Request 103: Match Not Found SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.50.10.183:5060 ---> SIP/2.0 180 Ringing Call-ID: 6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171 CSeq: 102 INVITE From: "WIRELESS CALLER" ;tag=as31d30179 To: ;tag=239603364e93a0b Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK581f07ea;rport Content-Length: 0 Call-Info: ;appearance-index=1 Allow-Events: talk, hold, conference Contact: User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 <-------------> [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 180 Ringing (19) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171 (54) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 102 INVITE (16) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: "WIRELESS CALLER" ;tag=as31d30179 (68) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: ;tag=239603364e93a0b (51) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK581f07ea;rport (63) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Call-Info: ;appearance-index=1 (48) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Allow-Events: talk, hold, conference (36) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: Contact: (36) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 (70) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: (0) --- (11 headers 0 lines) --- [May 17 10:02:39] DEBUG[7809]: chan_sip.c:2123 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #114 - INVITE (got response) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:2132 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171' Request 102: Found [May 17 10:02:39] DEBUG[7809]: chan_sip.c:11658 handle_response_invite: SIP response 180 to standard invite [May 17 10:02:39] DEBUG[7809]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/station2-08d7fe60 <--- SIP read from 10.50.10.177:5060 ---> SIP/2.0 180 Ringing Call-ID: 28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171 CSeq: 102 INVITE From: "WIRELESS CALLER" ;tag=as3f73485e To: ;tag=eebbd6a54ff83f3 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK4860bebb;rport Content-Length: 0 Allow-Events: talk, hold, conference Contact: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 180 Ringing (19) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171 (54) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 102 INVITE (16) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: "WIRELESS CALLER" ;tag=as3f73485e (68) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: ;tag=eebbd6a54ff83f3 (51) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK4860bebb;rport (63) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Allow-Events: talk, hold, conference (36) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: -- SIP/station2-08d7fe60 is ringing Header 8: Contact: (36) [May 17 10:02:39] DEBUG[7787]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - station2 [May 17 10:02:39] DEBUG[7787]: chan_sip.c:15321 sip_devicestate: Checking device state for peer station2 [May 17 10:02:39] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SIP/station2 - state 1 (Not in use) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [May 17 10:02:39] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [May 17 10:02:39] DEBUG[7883]: app_queue.c:546 changethread: Device 'SIP/station2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 17 10:02:39] DEBUG[7809]: chan_sip.c:2132 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171' Request 102: Found [May 17 10:02:39] DEBUG[7809]: chan_sip.c:11658 handle_response_invite: SIP response 180 to standard invite [May 17 10:02:39] DEBUG[7809]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/station1-08d77868 -- SIP/station1-08d77868 is ringing [May 17 10:02:39] DEBUG[7787]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - station1 [May 17 10:02:39] DEBUG[7787]: chan_sip.c:15321 sip_devicestate: Checking device state for peer station1 [May 17 10:02:39] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SIP/station1 - state 1 (Not in use) [May 17 10:02:39] DEBUG[7884]: app_queue.c:546 changethread: Device 'SIP/station1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 10.50.10.2:5060 ---> OPTIONS sip:metaswitch@10.50.10.171:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.10.2:5060;rport;branch=z9hG4bK-5b728ad8c950342bf724f2c570b0dff3-metaswitch-1 Allow-Events: message-summary Allow-Events: refer Allow-Events: dialog Allow-Events: line-seize Max-Forwards: 70 Call-ID: A59906C1@metaswitch From: ;tag=metaswitch+1+0+1ba8a8c6 CSeq: 1070899235 OPTIONS Organization: Supported: 100rel Content-Length: 0 Contact: To: <-------------> [May 17 10:02:42] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: OPTIONS sip:metaswitch@10.50.10.171:5060;transport=udp SIP/2.0 (62) [May 17 10:02:42] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.10.2:5060;rport;branch=z9hG4bK-5b728ad8c950342bf724f2c570b0dff3-metaswitch-1 (99) [May 17 10:02:42] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: Allow-Events: message-summary (29) [May 17 10:02:42] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: Allow-Events: refer (19) [May 17 10:02:42] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: Allow-Events: dialog (20) [May 17 10:02:42] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Allow-Events: line-seize (24) [May 17 10:02:42] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Max-Forwards: 70 (16) [May 17 10:02:42] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Call-ID: A59906C1@metaswitch (28) [May 17 10:02:42] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: From: ;tag=metaswitch+1+0+1ba8a8c6 (66) [May 17 10:02:42] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: CSeq: 1070899235 OPTIONS (24) [May 17 10:02:42] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: Organization: (14) [May 17 10:02:42] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: Supported: 100rel (17) [May 17 10:02:42] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 12: Content-Length: 0 (17) [May 17 10:02:42] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 13: Contact: (41) [May 17 10:02:42] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 14: To: (33) [May 17 10:02:42] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 15: (0) --- (15 headers 0 lines) --- [May 17 10:02:42] DEBUG[7809]: chan_sip.c:4312 sip_alloc: Allocating new SIP dialog for A59906C1@metaswitch - OPTIONS (No RTP) [May 17 10:02:42] DEBUG[7809]: chan_sip.c:14709 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for metaswitch in default (domain 10.50.10.171) <--- Transmitting (no NAT) to 10.50.10.2:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.50.10.2:5060;rport;branch=z9hG4bK-5b728ad8c950342bf724f2c570b0dff3-metaswitch-1;received=10.50.10.2 From: ;tag=metaswitch+1+0+1ba8a8c6 To: ;tag=as6b52dd80 Call-ID: A59906C1@metaswitch CSeq: 1070899235 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'A59906C1@metaswitch' in 32000 ms (Method: OPTIONS) [May 17 10:02:42] DEBUG[7809]: chan_sip.c:14927 sipsock_read: SIP message could not be handled, bad request: A59906C1@metaswitch [May 17 10:02:44] DEBUG[7809]: chan_sip.c:2011 __sip_autodestruct: Auto destroying SIP dialog '476D22A4@metaswitch' [May 17 10:02:44] DEBUG[7809]: chan_sip.c:3110 sip_destroy: Destroying SIP dialog 476D22A4@metaswitch Really destroying SIP dialog '476D22A4@metaswitch' Method: OPTIONS [May 17 10:02:47] DEBUG[7809]: chan_sip.c:2011 __sip_autodestruct: Auto destroying SIP dialog '0E002550@metaswitch' [May 17 10:02:47] DEBUG[7809]: chan_sip.c:3110 sip_destroy: Destroying SIP dialog 0E002550@metaswitch Really destroying SIP dialog '0E002550@metaswitch' Method: OPTIONS [May 17 10:02:49] DEBUG[7803]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SLA:station1_line2 [May 17 10:02:49] DEBUG[7803]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SLA:station2_line2 [May 17 10:02:49] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station1_line2 [May 17 10:02:49] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SLA:station1_line2 - state 1 (Not in use) [May 17 10:02:49] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station1_line2 [May 17 10:02:49] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station1_line2 Reliably Transmitting (no NAT) to 10.50.10.177:5060: NOTIFY sip:station1@10.50.10.177 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK183ffc77;rport From: Line 2 ;tag=as364fefba To: Trunk02 ;tag=676fadec36d2598 Contact: Call-ID: 8656389e4098c6e1ef9676ad4129fce6@10.50.10.177 CSeq: 104 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 230

terminated

--- [May 17 10:02:49] DEBUG[7787]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #117 Extension Changed station1_line2 new state Idle for Notify User station1 [May 17 10:02:49] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station2_line2 [May 17 10:02:49] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SLA:station2_line2 - state 1 (Not in use) [May 17 10:02:49] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station2_line2 [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station2_line2 Reliably Transmitting (no NAT) to 10.50.10.183:5060: NOTIFY sip:station2@10.50.10.183 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK5b21c13c;rport From: sip:station2_line2@10.50.10.171:5060;tag=as73da2e01 To: Trunk02 ;tag=3e52f55d7f04b26 Contact: Call-ID: 0fbce7d09c3594aa4ae1eb6724f58925@10.50.10.183 CSeq: 104 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 230

terminated

--- [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7787]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #118 Extension Changed station2_line2 new state Idle for Notify User station2 [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 's' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'sla_stations' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'SIP/station2-08d7fe60' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:02:39' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:02:49' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '10' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '0' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'NO ANSWER' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '1179421359.12' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7803]: channel.c:1726 ast_hangup: Hanging up channel 'SIP/station2-08d7fe60' [May 17 10:02:49] DEBUG[7803]: chan_sip.c:3313 sip_hangup: Hangup call SIP/station2-08d7fe60, SIP callid 6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171) [May 17 10:02:49] DEBUG[7803]: chan_sip.c:3336 sip_hangup: Hanging up channel in state Ringing (not UP) Scheduling destruction of SIP dialog '6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171' in 32000 ms (Method: INVITE) [May 17 10:02:49] DEBUG[7803]: chan_sip.c:2072 __sip_ack: Acked pending invite 102 [May 17 10:02:49] DEBUG[7803]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171' of Request 102: Match Not Found Reliably Transmitting (no NAT) to 10.50.10.183:5060: CANCEL sip:station2@10.50.10.183 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK581f07ea;rport From: "WIRELESS CALLER" ;tag=as31d30179 To: Call-ID: 6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [May 17 10:02:49] DEBUG[7803]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #120 Scheduling destruction of SIP dialog '6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171' in 32000 ms (Method: INVITE) [May 17 10:02:49] DEBUG[7803]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/station2-08d7fe60 [May 17 10:02:49] DEBUG[7787]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - station2 [May 17 10:02:49] DEBUG[7787]: chan_sip.c:15321 sip_devicestate: Checking device state for peer station2 [May 17 10:02:49] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SIP/station2 - state 1 (Not in use) [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 's' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'sla_stations' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'SIP/station1-08d77868' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:02:39' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:02:49' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '10' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '0' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'NO ANSWER' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '1179421359.11' [May 17 10:02:49] DEBUG[7803]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7887]: app_queue.c:546 changethread: Device 'SIP/station2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 17 10:02:49] DEBUG[7875]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel meetme:SLA_line2 [May 17 10:02:49] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "meetme" - number: SLA_line2 [May 17 10:02:49] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 's' [May 17 10:02:49] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'default' [May 17 10:02:49] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'Zap/pseudo-2087517722' [May 17 10:02:49] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:02:39' [May 17 10:02:49] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:02:49' [May 17 10:02:49] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '10' [May 17 10:02:49] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '0' [May 17 10:02:49] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'NO ANSWER' [May 17 10:02:49] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [May 17 10:02:49] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '1179421359.10' [May 17 10:02:49] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:02:49] DEBUG[7875]: channel.c:1726 ast_hangup: Hanging up channel 'Zap/pseudo-2087517722' [May 17 10:02:49] DEBUG[7875]: chan_zap.c:2412 zt_hangup: zt_hangup(Zap/pseudo-2087517722) [May 17 10:02:49] DEBUG[7875]: chan_zap.c:2446 zt_hangup: Hangup: channel: -2 index = 0, normal = 24, callwait = -1, thirdcall = -1 [May 17 10:02:49] DEBUG[7875]: chan_zap.c:2874 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/pseudo-2087517722 [May 17 10:02:49] DEBUG[7875]: chan_zap.c:1403 update_conf: Updated conferencing on -2, with 0 conference users -- Hungup 'Zap/pseudo-2087517722' [May 17 10:02:49] DEBUG[7875]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel Zap/pseudo-2087517722 [May 17 10:02:49] DEBUG[7875]: pbx.c:1809 pbx_extension_helper: Launching 'Dial' -- Executing [8055488092@line2:3] Dial("SIP/8055488091-08d62b90", "SIP/5480002|10") in new stack [May 17 10:02:49] DEBUG[7875]: chan_sip.c:15387 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4312 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:2576 do_setnat: Setting NAT on RTP to Off [May 17 10:02:49] DEBUG[7875]: frame.c:1267 ast_codec_choose: Could not find preferred codec - Going for the best codec [May 17 10:02:49] DEBUG[7875]: chan_sip.c:3806 sip_new: *** Our native formats are 0x4 (ulaw) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:3807 sip_new: *** Joint capabilities are 0x0 (nothing) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:3808 sip_new: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263) [May 17 10:02:49] DEBUG[7875]: frame.c:1267 ast_codec_choose: Could not find preferred codec - Going for the best codec [May 17 10:02:49] DEBUG[7875]: chan_sip.c:3809 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:3811 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:3832 sip_new: This channel will not be able to handle video. [May 17 10:02:49] DEBUG[7875]: rtp.c:1598 ast_rtp_make_compatible: Seeded SDP of 'SIP/5480002-08d5e090' with that of 'SIP/8055488091-08d62b90' [May 17 10:02:49] DEBUG[7875]: channel.c:3376 ast_channel_inherit_variables: Not copying variable STACK-line2-8055488092-3. [May 17 10:02:49] DEBUG[7875]: channel.c:3376 ast_channel_inherit_variables: Not copying variable SLATRUNK_STATUS. [May 17 10:02:49] DEBUG[7875]: channel.c:3376 ast_channel_inherit_variables: Not copying variable MEETMESECS. [May 17 10:02:49] DEBUG[7875]: channel.c:3376 ast_channel_inherit_variables: Not copying variable STACK-line2-8055488092-2. [May 17 10:02:49] DEBUG[7875]: channel.c:3376 ast_channel_inherit_variables: Not copying variable STACK-line2-8055488092-1. [May 17 10:02:49] DEBUG[7875]: channel.c:3376 ast_channel_inherit_variables: Not copying variable STACK-sip-8055480617-1. [May 17 10:02:49] DEBUG[7875]: channel.c:3376 ast_channel_inherit_variables: Not copying variable SIPCALLID. [May 17 10:02:49] DEBUG[7875]: channel.c:3376 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [May 17 10:02:49] DEBUG[7875]: channel.c:3376 ast_channel_inherit_variables: Not copying variable SIPURI. [May 17 10:02:49] DEBUG[7875]: chan_sip.c:2831 sip_call: Outgoing Call for 5480002 [May 17 10:02:49] DEBUG[7875]: chan_sip.c:3004 update_call_counter: Updating call counter for outgoing call [May 17 10:02:49] DEBUG[7875]: chan_sip.c:2846 sip_call: Our T38 capability (0), joint T38 capability (0) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:6198 add_sdp: ** Our capability: 0x8000e (gsm|ulaw|alaw|h263) Video flag: False [May 17 10:02:49] DEBUG[7875]: chan_sip.c:6199 add_sdp: ** Our prefcodec: 0x4 (ulaw) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:6216 add_sdp: This call needs video offers, but there's no video support enabled! Audio is at 10.50.10.171 port 13994 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [May 17 10:02:49] DEBUG[7875]: chan_sip.c:6330 add_sdp: -- Done with adding codecs to SDP [May 17 10:02:49] DEBUG[7875]: chan_sip.c:6375 add_sdp: Done building SDP. Settling with this capability: 0x8000e (gsm|ulaw|alaw|h263) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 0: INVITE sip:5480002@10.50.10.177 SIP/2.0 (39) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK057f1f59;rport (63) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 2: From: "WIRELESS CALLER" ;tag=as294ff023 (68) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 3: To: (30) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 4: Contact: (38) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 5: Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 (54) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 6: CSeq: 102 INVITE (16) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 7: User-Agent: Asterisk PBX (24) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 8: Max-Forwards: 70 (16) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 9: Date: Thu, 17 May 2007 17:02:49 GMT (35) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 11: Supported: replaces (19) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 12: Content-Type: application/sdp (29) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 13: Content-Length: 285 (19) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 14: (0) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: v=0 (3) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: o=root 7783 7783 IN IP4 10.50.10.171 (36) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: s=session (9) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: c=IN IP4 10.50.10.171 (21) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: t=0 0 (5) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: m=audio 13994 RTP/AVP 0 3 8 101 (31) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=fmtp:101 0-16 (15) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=silenceSupp:off - - - - (25) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=ptime:20 (10) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 10.50.10.177:5060: INVITE sip:5480002@10.50.10.177 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK057f1f59;rport From: "WIRELESS CALLER" ;tag=as294ff023 To: Contact: Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 17 May 2007 17:02:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 7783 7783 IN IP4 10.50.10.171 s=session c=IN IP4 10.50.10.171 t=0 0 m=audio 13994 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [May 17 10:02:49] DEBUG[7875]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #122 -- Called 5480002 [May 17 10:02:49] DEBUG[7875]: channel.c:2911 set_format: Set channel SIP/8055488091-08d62b90 to write format ulaw [May 17 10:02:49] DEBUG[7875]: channel.c:2911 set_format: Set channel SIP/8055488091-08d62b90 to read format ulaw [May 17 10:02:49] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for meetme:SLA_line2 - state 4 (Invalid) [May 17 10:02:49] DEBUG[7787]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for Zap - pseudo [May 17 10:02:49] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for Zap/pseudo - state 0 (Unknown) [May 17 10:02:49] DEBUG[7803]: channel.c:1726 ast_hangup: Hanging up channel 'SIP/station1-08d77868' [May 17 10:02:49] DEBUG[7803]: chan_sip.c:3313 sip_hangup: Hangup call SIP/station1-08d77868, SIP callid 28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171) [May 17 10:02:49] DEBUG[7803]: chan_sip.c:3336 sip_hangup: Hanging up channel in state Ringing (not UP) Scheduling destruction of SIP dialog '28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171' in 32000 ms (Method: INVITE) [May 17 10:02:49] DEBUG[7803]: chan_sip.c:2072 __sip_ack: Acked pending invite 102 [May 17 10:02:49] DEBUG[7803]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171' of Request 102: Match Not Found Reliably Transmitting (no NAT) to 10.50.10.177:5060: CANCEL sip:station1@10.50.10.177 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK4860bebb;rport From: "WIRELESS CALLER" ;tag=as3f73485e To: Call-ID: 28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [May 17 10:02:49] DEBUG[7803]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #125 Scheduling destruction of SIP dialog '28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171' in 32000 ms (Method: INVITE) [May 17 10:02:49] DEBUG[7803]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/station1-08d77868 [May 17 10:02:49] DEBUG[7787]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - station1 [May 17 10:02:49] DEBUG[7787]: chan_sip.c:15321 sip_devicestate: Checking device state for peer station1 [May 17 10:02:49] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SIP/station1 - state 1 (Not in use) [May 17 10:02:49] DEBUG[7889]: app_queue.c:546 changethread: Device 'Zap/pseudo' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. [May 17 10:02:49] DEBUG[7890]: app_queue.c:546 changethread: Device 'SIP/station1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 10.50.10.177:5060 ---> SIP/2.0 200 OK Call-ID: 8656389e4098c6e1ef9676ad4129fce6@10.50.10.177 CSeq: 104 NOTIFY From: Line 2 ;tag=as364fefba To: Trunk02 ;tag=676fadec36d2598 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK183ffc77;rport Content-Length: 0 Contact: Trunk02 Supported: replaces User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 200 OK (14) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 8656389e4098c6e1ef9676ad4129fce6@10.50.10.177 (54) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 104 NOTIFY (16) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: Line 2 ;tag=as364fefba (66) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: Trunk02 ;tag=676fadec36d2598 (59) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK183ffc77;rport (63) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Contact: Trunk02 (44) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Supported: replaces (19) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [May 17 10:02:49] DEBUG[7809]: chan_sip.c:2080 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #117 [May 17 10:02:49] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '8656389e4098c6e1ef9676ad4129fce6@10.50.10.177' of Request 104: Match Not Found SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.50.10.177:5060 ---> SIP/2.0 200 OK Call-ID: 28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171 CSeq: 102 CANCEL From: "WIRELESS CALLER" ;tag=as3f73485e To: ;tag=eebbd6a54ff83f3 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK4860bebb;rport Content-Length: 0 Contact: Supported: replaces User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 200 OK (14) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171 (54) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 102 CANCEL (16) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: "WIRELESS CALLER" ;tag=as3f73485e (68) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: ;tag=eebbd6a54ff83f3 (51) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK4860bebb;rport (63) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Contact: (36) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Supported: replaces (19) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [May 17 10:02:49] DEBUG[7809]: chan_sip.c:2080 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #125 [May 17 10:02:49] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171' of Request 102: Match Not Found <--- SIP read from 10.50.10.177:5060 ---> SIP/2.0 487 Request Terminated Call-ID: 28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171 CSeq: 102 INVITE From: "WIRELESS CALLER" ;tag=as3f73485e To: ;tag=eebbd6a54ff83f3 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK4860bebb;rport Content-Length: 0 User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 487 Request Terminated (30) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171 (54) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 102 INVITE (16) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: "WIRELESS CALLER" ;tag=as3f73485e (68) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: ;tag=eebbd6a54ff83f3 (51) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK4860bebb;rport (63) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [May 17 10:02:49] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171' of Request 102: Match Found [May 17 10:02:49] DEBUG[7809]: chan_sip.c:11658 handle_response_invite: SIP response 487 to standard invite Transmitting (no NAT) to 10.50.10.177:5060: ACK sip:station1@10.50.10.177 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK4860bebb;rport From: "WIRELESS CALLER" ;tag=as3f73485e To: ;tag=eebbd6a54ff83f3 Contact: Call-ID: 28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [May 17 10:02:49] DEBUG[7809]: chan_sip.c:3004 update_call_counter: Updating call counter for outgoing call <--- SIP read from 10.50.10.177:5060 ---> SIP/2.0 100 Trying Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 CSeq: 102 INVITE From: "WIRELESS CALLER" ;tag=as294ff023 To: ;tag=3acb42d788a1ba4 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK057f1f59;rport Content-Length: 0 User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 100 Trying (18) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 (54) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 102 INVITE (16) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: "WIRELESS CALLER" ;tag=as294ff023 (68) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: ;tag=3acb42d788a1ba4 (50) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK057f1f59;rport (63) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [May 17 10:02:49] DEBUG[7809]: chan_sip.c:2123 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #122 - INVITE (got response) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:2132 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '368c400d429a7fb14282ed567649675d@10.50.10.171' Request 102: Found [May 17 10:02:49] DEBUG[7809]: chan_sip.c:11658 handle_response_invite: SIP response 100 to standard invite <--- SIP read from 10.50.10.183:5060 ---> SIP/2.0 487 Request Terminated Call-ID: 6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171 CSeq: 102 INVITE From: "WIRELESS CALLER" ;tag=as31d30179 To: ;tag=239603364e93a0b Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK581f07ea;rport Content-Length: 0 User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 <-------------> [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 487 Request Terminated (30) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171 (54) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 102 INVITE (16) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: "WIRELESS CALLER" ;tag=as31d30179 (68) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: ;tag=239603364e93a0b (51) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK581f07ea;rport (63) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 (70) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [May 17 10:02:49] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171' of Request 102: Match Found [May 17 10:02:49] DEBUG[7809]: chan_sip.c:11658 handle_response_invite: SIP response 487 to standard invite Transmitting (no NAT) to 10.50.10.183:5060: ACK sip:station2@10.50.10.183 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK581f07ea;rport From: "WIRELESS CALLER" ;tag=as31d30179 To: ;tag=239603364e93a0b Contact: Call-ID: 6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [May 17 10:02:49] DEBUG[7809]: chan_sip.c:3004 update_call_counter: Updating call counter for outgoing call <--- SIP read from 10.50.10.183:5060 ---> SIP/2.0 200 OK Call-ID: 0fbce7d09c3594aa4ae1eb6724f58925@10.50.10.183 CSeq: 104 NOTIFY From: sip:station2_line2@10.50.10.171:5060;tag=as73da2e01 To: Trunk02 ;tag=3e52f55d7f04b26 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK5b21c13c;rport Content-Length: 0 Contact: Trunk02 Supported: replaces User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 <-------------> [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 200 OK (14) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 0fbce7d09c3594aa4ae1eb6724f58925@10.50.10.183 (54) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 104 NOTIFY (16) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: sip:station2_line2@10.50.10.171:5060;tag=as73da2e01 (57) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: Trunk02 ;tag=3e52f55d7f04b26 (59) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK5b21c13c;rport (63) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Contact: Trunk02 (44) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Supported: replaces (19) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 (70) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [May 17 10:02:49] DEBUG[7809]: chan_sip.c:2080 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #118 [May 17 10:02:49] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '0fbce7d09c3594aa4ae1eb6724f58925@10.50.10.183' of Request 104: Match Not Found SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.50.10.183:5060 ---> SIP/2.0 200 OK Call-ID: 6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171 CSeq: 102 CANCEL From: "WIRELESS CALLER" ;tag=as31d30179 To: ;tag=239603364e93a0b Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK581f07ea;rport Content-Length: 0 Supported: replaces User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 <-------------> [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 200 OK (14) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171 (54) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 102 CANCEL (16) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: "WIRELESS CALLER" ;tag=as31d30179 (68) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: ;tag=239603364e93a0b (51) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK581f07ea;rport (63) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Supported: replaces (19) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 (70) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [May 17 10:02:49] DEBUG[7809]: chan_sip.c:2080 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #120 [May 17 10:02:49] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171' of Request 102: Match Not Found <--- SIP read from 10.50.10.177:5060 ---> SIP/2.0 180 Ringing Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 CSeq: 102 INVITE From: "WIRELESS CALLER" ;tag=as294ff023 To: ;tag=3acb42d788a1ba4 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK057f1f59;rport Content-Length: 0 Allow-Events: talk, hold, conference Contact: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 180 Ringing (19) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 (54) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 102 INVITE (16) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: "WIRELESS CALLER" ;tag=as294ff023 (68) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: ;tag=3acb42d788a1ba4 (50) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK057f1f59;rport (63) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Allow-Events: talk, hold, conference (36) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Contact: (35) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [May 17 10:02:49] DEBUG[7809]: chan_sip.c:2132 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '368c400d429a7fb14282ed567649675d@10.50.10.171' Request 102: Found [May 17 10:02:49] DEBUG[7809]: chan_sip.c:11658 handle_response_invite: SIP response 180 to standard invite [May 17 10:02:49] DEBUG[7809]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/5480002-08d5e090 [May 17 10:02:49] DEBUG[7787]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5480002 [May 17 10:02:49] DEBUG[7787]: chan_sip.c:15321 sip_devicestate: Checking device state for peer 5480002 -- SIP/5480002-08d5e090 is ringing [May 17 10:02:49] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SIP/5480002 - state 1 (Not in use) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:16970 sip_set_rtp_peer: Sending reinvite on SIP 'C371B383@metaswitch' - It's audio soon redirected to IP 10.50.10.171 [May 17 10:02:49] DEBUG[7875]: chan_sip.c:5651 reqprep: Strict routing enforced for session C371B383@metaswitch set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.10.2, port 5060 [May 17 10:02:49] DEBUG[7875]: chan_sip.c:6198 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [May 17 10:02:49] DEBUG[7891]: app_queue.c:546 changethread: Device 'SIP/5480002' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 17 10:02:49] DEBUG[7875]: chan_sip.c:6199 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.50.10.171 port 13754 Adding codec 0x4 (ulaw) to SDP [May 17 10:02:49] DEBUG[7875]: chan_sip.c:6330 add_sdp: -- Done with adding codecs to SDP [May 17 10:02:49] DEBUG[7875]: chan_sip.c:6375 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:1622 initialize_initreq: Initializing already initialized SIP dialog C371B383@metaswitch (presumably reinvite) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 0: INVITE sip:8054584444@10.50.10.2:5060 SIP/2.0 (45) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK49d43634 (57) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 2: From: ;tag=as48f37b45 (50) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 3: To: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 (97) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 4: Contact: (38) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 5: Call-ID: C371B383@metaswitch (28) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 6: CSeq: 102 INVITE (16) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 7: User-Agent: Asterisk PBX (24) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 8: Max-Forwards: 70 (16) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 10: Supported: replaces (19) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 12: Content-Type: application/sdp (29) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 13: Content-Length: 182 (19) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 14: (0) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: v=0 (3) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: o=root 7783 7784 IN IP4 10.50.10.171 (36) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: s=session (9) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: c=IN IP4 10.50.10.171 (21) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: t=0 0 (5) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: m=audio 13754 RTP/AVP 0 (23) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=silenceSupp:off - - - - (25) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=ptime:20 (10) [May 17 10:02:49] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 10.50.10.2:5060: INVITE sip:8054584444@10.50.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK49d43634 From: ;tag=as48f37b45 To: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 Contact: Call-ID: C371B383@metaswitch CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 182 v=0 o=root 7783 7784 IN IP4 10.50.10.171 s=session c=IN IP4 10.50.10.171 t=0 0 m=audio 13754 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [May 17 10:02:49] DEBUG[7875]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #127 [May 17 10:02:49] DEBUG[7875]: rtp.c:1527 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/8055488091-08d62b90' with that of 'SIP/5480002-08d5e090' [May 17 10:02:49] DEBUG[7875]: channel.c:2501 ast_indicate_data: Driver for channel 'SIP/8055488091-08d62b90' does not support indication 3, emulating it [May 17 10:02:49] DEBUG[7875]: channel.c:2911 set_format: Set channel SIP/8055488091-08d62b90 to write format slin [May 17 10:02:49] DEBUG[7875]: channel.c:2030 ast_settimeout: Scheduling timer at 160 sample intervals [May 17 10:02:49] DEBUG[7875]: channel.c:2398 __ast_read: Generator got voice, switching to phase locked mode [May 17 10:02:49] DEBUG[7875]: channel.c:2030 ast_settimeout: Scheduling timer at 0 sample intervals [May 17 10:02:49] DEBUG[7875]: rtp.c:2571 ast_rtp_raw_write: Difference is 2704, ms is 358 <--- SIP read from 10.50.10.2:5060 ---> SIP/2.0 100 Trying Call-ID: C371B383@metaswitch CSeq: 102 INVITE From: ;tag=as48f37b45 To: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK49d43634 Server: DC-SIP/2.0 Organization: Supported: 100rel Contact: WIRELESS CALLER ;isup-oli=62 Content-Length: 0 <-------------> [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 100 Trying (18) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: C371B383@metaswitch (28) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 102 INVITE (16) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: ;tag=as48f37b45 (50) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 (97) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK49d43634 (57) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Server: DC-SIP/2.0 (18) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Organization: (14) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Supported: 100rel (17) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: Contact: WIRELESS CALLER ;isup-oli=62 (69) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: Content-Length: 0 (17) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: (0) --- (11 headers 0 lines) --- [May 17 10:02:49] DEBUG[7809]: chan_sip.c:2123 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #127 - INVITE (got response) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:2132 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on 'C371B383@metaswitch' Request 102: Found [May 17 10:02:49] DEBUG[7809]: chan_sip.c:11656 handle_response_invite: SIP response 100 to RE-invite on outgoing call C371B383@metaswitch <--- SIP read from 10.50.10.2:5060 ---> SIP/2.0 200 OK Call-ID: C371B383@metaswitch CSeq: 102 INVITE From: ;tag=as48f37b45 To: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK49d43634 Server: DC-SIP/2.0 Organization: Allow-Events: message-summary Allow-Events: refer Allow-Events: dialog Allow-Events: line-seize Supported: 100rel Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: REGISTER Allow: OPTIONS Allow: PRACK Allow: UPDATE Allow: SUBSCRIBE Allow: NOTIFY Allow: REFER Accept-Encoding: identity Accept: application/sdp Accept: application/simple-message-summary Accept: message/sipfrag Accept: application/isup Accept: application/x-simple-call-service-info Accept: multipart/mixed Contact: WIRELESS CALLER ;isup-oli=62 Content-Length: 122 Content-Type: application/sdp v=0 o=- 2600422519 2600422519 IN IP4 10.50.10.13 s=- c=IN IP4 10.50.10.13 t=0 0 m=audio 33758 RTP/AVP 0 a=ptime:20 <-------------> [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 200 OK (14) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: C371B383@metaswitch (28) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 102 INVITE (16) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: ;tag=as48f37b45 (50) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 (97) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK49d43634 (57) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Server: DC-SIP/2.0 (18) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Organization: (14) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Allow-Events: message-summary (29) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: Allow-Events: refer (19) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: Allow-Events: dialog (20) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: Allow-Events: line-seize (24) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 12: Supported: 100rel (17) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 13: Allow: INVITE (13) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 14: Allow: ACK (10) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 15: Allow: CANCEL (13) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 16: Allow: BYE (10) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 17: Allow: REGISTER (15) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 18: Allow: OPTIONS (14) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 19: Allow: PRACK (12) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 20: Allow: UPDATE (13) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 21: Allow: SUBSCRIBE (16) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 22: Allow: NOTIFY (13) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 23: Allow: REFER (12) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 24: Accept-Encoding: identity (25) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 25: Accept: application/sdp (23) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 26: Accept: application/simple-message-summary (42) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 27: Accept: message/sipfrag (23) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 28: Accept: application/isup (24) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 29: Accept: application/x-simple-call-service-info (46) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 30: Accept: multipart/mixed (23) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 31: Contact: WIRELESS CALLER ;isup-oli=62 (69) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 32: Content-Length: 122 (19) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 33: Content-Type: application/sdp (29) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 34: (0) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: v=0 (3) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: o=- 2600422519 2600422519 IN IP4 10.50.10.13 (44) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: s=- (3) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: c=IN IP4 10.50.10.13 (20) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: t=0 0 (5) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: m=audio 33758 RTP/AVP 0 (23) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=ptime:20 (10) --- (34 headers 7 lines) --- [May 17 10:02:49] DEBUG[7809]: chan_sip.c:2072 __sip_ack: Acked pending invite 102 [May 17 10:02:49] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on 'C371B383@metaswitch' of Request 102: Match Not Found [May 17 10:02:49] DEBUG[7809]: chan_sip.c:11656 handle_response_invite: SIP response 200 to RE-invite on outgoing call C371B383@metaswitch Found RTP audio format 0 Peer audio RTP is at port 10.50.10.13:33758 [May 17 10:02:49] DEBUG[7809]: chan_sip.c:5136 process_sdp: T38 state changed to 0 on channel SIP/8055488091-08d62b90 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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 10.50.10.13:33758 [May 17 10:02:49] DEBUG[7809]: chan_sip.c:5216 process_sdp: We're settling with these formats: 0x4 (ulaw) [May 17 10:02:49] DEBUG[7809]: chan_sip.c:5223 process_sdp: We have an owner, now see if we need to change this call [May 17 10:02:49] DEBUG[7809]: chan_sip.c:3004 update_call_counter: Updating call counter for incoming call [May 17 10:02:49] DEBUG[7809]: chan_sip.c:7994 build_route: build_route: Contact hop: WIRELESS CALLER ;isup-oli=62 list_route: hop: [May 17 10:02:49] DEBUG[7809]: chan_sip.c:5651 reqprep: Strict routing enforced for session C371B383@metaswitch set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.10.2, port 5060 Transmitting (no NAT) to 10.50.10.2:5060: ACK sip:8054584444@10.50.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK09a28352 From: ;tag=as48f37b45 To: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 Contact: Call-ID: C371B383@metaswitch CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/8055488091-08d62b90 requested special control 17, passing it to SIP/5480002-08d5e090 <--- SIP read from 10.50.10.177:5060 ---> SIP/2.0 200 OK Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 CSeq: 102 INVITE From: "WIRELESS CALLER" ;tag=as294ff023 To: ;tag=3acb42d788a1ba4 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK057f1f59;rport Content-Length: 253 Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Supported: replaces Contact: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 725269576 IN IP4 10.50.10.177 s=SIP Call c=IN IP4 10.50.10.177 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=silenceSupp:off - - - - <-------------> [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 200 OK (14) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 (54) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 102 INVITE (16) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: "WIRELESS CALLER" ;tag=as294ff023 (68) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: ;tag=3acb42d788a1ba4 (50) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK057f1f59;rport (63) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 253 (19) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Allow-Events: talk,hold,conference (34) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO (53) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: Content-Type: application/sdp (29) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: Supported: replaces (19) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: Contact: (35) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 12: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 13: (0) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: v=0 (3) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: o=MxSIP 0 725269576 IN IP4 10.50.10.177 (39) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: s=SIP Call (10) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: c=IN IP4 10.50.10.177 (21) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: t=0 0 (5) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: m=audio 3000 RTP/AVP 0 8 101 (28) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=fmtp:101 0-15 (15) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=ptime:20 (10) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=silenceSupp:off - - - - (25) --- (13 headers 12 lines) --- [May 17 10:02:58] DEBUG[7809]: chan_sip.c:2072 __sip_ack: Acked pending invite 102 [May 17 10:02:58] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '368c400d429a7fb14282ed567649675d@10.50.10.171' of Request 102: Match Not Found [May 17 10:02:58] DEBUG[7809]: chan_sip.c:11658 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.10.177:3000 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [May 17 10:02:58] DEBUG[7809]: chan_sip.c:5136 process_sdp: T38 state changed to 0 on channel SIP/5480002-08d5e090 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.50.10.177:3000 [May 17 10:02:58] DEBUG[7809]: chan_sip.c:5216 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:5223 process_sdp: We have an owner, now see if we need to change this call [May 17 10:02:58] DEBUG[7809]: chan_sip.c:3004 update_call_counter: Updating call counter for outgoing call [May 17 10:02:58] DEBUG[7809]: chan_sip.c:7994 build_route: build_route: Contact hop: list_route: hop: [May 17 10:02:58] DEBUG[7809]: chan_sip.c:5651 reqprep: Strict routing enforced for session 368c400d429a7fb14282ed567649675d@10.50.10.171 set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.10.177, port 5060 Transmitting (no NAT) to 10.50.10.177:5060: ACK sip:5480002@10.50.10.177 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK7b05739e;rport From: "WIRELESS CALLER" ;tag=as294ff023 To: ;tag=3acb42d788a1ba4 Contact: Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- Call on SIP/5480002-08d5e090 left from hold [May 17 10:02:58] DEBUG[7875]: channel.c:2911 set_format: Set channel SIP/8055488091-08d62b90 to write format ulaw [May 17 10:02:58] DEBUG[7875]: channel.c:2030 ast_settimeout: Scheduling timer at 0 sample intervals [May 17 10:02:58] DEBUG[7875]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/5480002-08d5e090 -- SIP/5480002-08d5e090 answered SIP/8055488091-08d62b90 [May 17 10:02:58] DEBUG[7875]: chan_sip.c:16970 sip_set_rtp_peer: Sending reinvite on SIP 'C371B383@metaswitch' - It's audio soon redirected to IP 10.50.10.177 [May 17 10:02:58] DEBUG[7875]: chan_sip.c:5651 reqprep: Strict routing enforced for session C371B383@metaswitch set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.10.2, port 5060 [May 17 10:02:58] DEBUG[7875]: chan_sip.c:6198 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [May 17 10:02:58] DEBUG[7875]: chan_sip.c:6199 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.50.10.171 port 13754 Adding codec 0x4 (ulaw) to SDP [May 17 10:02:58] DEBUG[7875]: chan_sip.c:6330 add_sdp: -- Done with adding codecs to SDP [May 17 10:02:58] DEBUG[7875]: chan_sip.c:6375 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:1622 initialize_initreq: Initializing already initialized SIP dialog C371B383@metaswitch (presumably reinvite) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 0: INVITE sip:8054584444@10.50.10.2:5060 SIP/2.0 (45) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK27d4fe78 (57) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 2: From: ;tag=as48f37b45 (50) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 3: To: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 (97) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 4: Contact: (38) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 5: Call-ID: C371B383@metaswitch (28) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 6: CSeq: 103 INVITE (16) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 7: User-Agent: Asterisk PBX (24) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 8: Max-Forwards: 70 (16) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 10: Supported: replaces (19) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 12: Content-Type: application/sdp (29) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 13: Content-Length: 181 (19) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 14: (0) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: v=0 (3) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: o=root 7783 7785 IN IP4 10.50.10.177 (36) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: s=session (9) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: c=IN IP4 10.50.10.177 (21) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: t=0 0 (5) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: m=audio 3000 RTP/AVP 0 (22) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=silenceSupp:off - - - - (25) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=ptime:20 (10) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 10.50.10.2:5060: INVITE sip:8054584444@10.50.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK27d4fe78 From: ;tag=as48f37b45 To: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 Contact: Call-ID: C371B383@metaswitch CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 181 v=0 o=root 7783 7785 IN IP4 10.50.10.177 s=session c=IN IP4 10.50.10.177 t=0 0 m=audio 3000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [May 17 10:02:58] DEBUG[7875]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #128 [May 17 10:02:58] DEBUG[7875]: rtp.c:1527 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/8055488091-08d62b90' with that of 'SIP/5480002-08d5e090' -- Native bridging SIP/8055488091-08d62b90 and SIP/5480002-08d5e090 [May 17 10:02:58] DEBUG[7875]: chan_sip.c:16970 sip_set_rtp_peer: Sending reinvite on SIP '368c400d429a7fb14282ed567649675d@10.50.10.171' - It's audio soon redirected to IP 10.50.10.13 [May 17 10:02:58] DEBUG[7875]: chan_sip.c:5651 reqprep: Strict routing enforced for session 368c400d429a7fb14282ed567649675d@10.50.10.171 set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.10.177, port 5060 [May 17 10:02:58] DEBUG[7875]: chan_sip.c:6198 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [May 17 10:02:58] DEBUG[7875]: chan_sip.c:6199 add_sdp: ** Our prefcodec: 0x4 (ulaw) Audio is at 10.50.10.171 port 13994 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [May 17 10:02:58] DEBUG[7875]: chan_sip.c:6330 add_sdp: -- Done with adding codecs to SDP [May 17 10:02:58] DEBUG[7875]: chan_sip.c:6375 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:1622 initialize_initreq: Initializing already initialized SIP dialog 368c400d429a7fb14282ed567649675d@10.50.10.171 (presumably reinvite) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 0: INVITE sip:5480002@10.50.10.177 SIP/2.0 (39) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK0e3b708a;rport (63) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 2: From: "WIRELESS CALLER" ;tag=as294ff023 (68) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 3: To: ;tag=3acb42d788a1ba4 (50) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 4: Contact: (38) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 5: Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 (54) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 6: CSeq: 103 INVITE (16) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 7: User-Agent: Asterisk PBX (24) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 8: Max-Forwards: 70 (16) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 10: Supported: replaces (19) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 12: Content-Type: application/sdp (29) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 13: Content-Length: 260 (19) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 14: (0) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: v=0 (3) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: o=root 7783 7784 IN IP4 10.50.10.13 (35) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: s=session (9) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: c=IN IP4 10.50.10.13 (20) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: t=0 0 (5) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: m=audio 33758 RTP/AVP 0 8 101 (29) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=fmtp:101 0-16 (15) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=silenceSupp:off - - - - (25) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=ptime:20 (10) [May 17 10:02:58] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 10.50.10.177:5060: INVITE sip:5480002@10.50.10.177 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK0e3b708a;rport From: "WIRELESS CALLER" ;tag=as294ff023 To: ;tag=3acb42d788a1ba4 Contact: Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 260 v=0 o=root 7783 7784 IN IP4 10.50.10.13 s=session c=IN IP4 10.50.10.13 t=0 0 m=audio 33758 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [May 17 10:02:58] DEBUG[7875]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #129 [May 17 10:02:58] DEBUG[7787]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5480002 [May 17 10:02:58] DEBUG[7787]: chan_sip.c:15321 sip_devicestate: <--- SIP read from 10.50.10.2:5060 ---> SIP/2.0 100 Trying Call-ID: C371B383@metaswitch CSeq: 103 INVITE From: ;tag=as48f37b45 To: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK27d4fe78 Server: DC-SIP/2.0 Organization: Supported: 100rel Contact: WIRELESS CALLER ;isup-oli=62 Content-Length: 0 <-------------> Checking device state for peer 5480002 [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 100 Trying (18) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: C371B383@metaswitch (28) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 103 INVITE (16) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: ;tag=as48f37b45 (50) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 (97) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK27d4fe78 (57) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Server: DC-SIP/2.0 (18) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Organization: (14) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Supported: 100rel (17) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: Contact: WIRELESS CALLER ;isup-oli=62 (69) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: Content-Length: 0 (17) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: (0) --- (11 headers 0 lines) --- [May 17 10:02:58] DEBUG[7809]: chan_sip.c:2123 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #128 - INVITE (got response) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:2132 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on 'C371B383@metaswitch' Request 103: Found [May 17 10:02:58] DEBUG[7809]: chan_sip.c:11656 handle_response_invite: SIP response 100 to RE-invite on outgoing call C371B383@metaswitch <--- SIP read from 10.50.10.2:5060 ---> SIP/2.0 200 OK Call-ID: C371B383@metaswitch CSeq: 103 INVITE From: ;tag=as48f37b45 To: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK27d4fe78 Server: DC-SIP/2.0 Organization: Allow-Events: message-summary Allow-Events: refer Allow-Events: dialog Allow-Events: line-seize Supported: 100rel Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: REGISTER Allow: OPTIONS Allow: PRACK Allow: UPDATE Allow: SUBSCRIBE Allow: NOTIFY Allow: REFER Accept-Encoding: identity Accept: application/sdp Accept: application/simple-message-summary Accept: message/sipfrag Accept: application/isup Accept: application/x-simple-call-service-info Accept: multipart/mixed Contact: WIRELESS CALLER ;isup-oli=62 Content-Length: 122 Content-Type: application/sdp v=0 o=- 2600422519 2600422519 IN IP4 10.50.10.13 s=- c=IN IP4 10.50.10.13 t=0 0 m=audio 33758 RTP/AVP 0 a=ptime:20 <-------------> [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 200 OK (14) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: C371B383@metaswitch (28) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 103 INVITE (16) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: ;tag=as48f37b45 (50) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 (97) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK27d4fe78 (57) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Server: DC-SIP/2.0 (18) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Organization: (14) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Allow-Events: message-summary (29) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: Allow-Events: refer (19) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: Allow-Events: dialog (20) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: Allow-Events: line-seize (24) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 12: Supported: 100rel (17) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 13: Allow: INVITE (13) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 14: Allow: ACK (10) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 15: Allow: CANCEL (13) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 16: Allow: BYE (10) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 17: Allow: REGISTER (15) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 18: Allow: OPTIONS (14) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 19: Allow: PRACK (12) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 20: Allow: UPDATE (13) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 21: Allow: SUBSCRIBE (16) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 22: Allow: NOTIFY (13) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 23: Allow: REFER (12) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 24: Accept-Encoding: identity (25) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 25: Accept: application/sdp (23) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 26: Accept: application/simple-message-summary (42) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 27: Accept: message/sipfrag (23) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 28: Accept: application/isup (24) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 29: Accept: application/x-simple-call-service-info (46) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 30: Accept: multipart/mixed (23) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 31: Contact: WIRELESS CALLER ;isup-oli=62 (69) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 32: Content-Length: 122 (19) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 33: Content-Type: application/sdp (29) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 34: (0) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: v=0 (3) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: o=- 2600422519 2600422519 IN IP4 10.50.10.13 (44) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: s=- (3) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: c=IN IP4 10.50.10.13 (20) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: t=0 0 (5) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: m=audio 33758 RTP/AVP 0 (23) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=ptime:20 (10) --- (34 headers 7 lines) --- [May 17 10:02:58] DEBUG[7809]: chan_sip.c:2072 __sip_ack: Acked pending invite 103 [May 17 10:02:58] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on 'C371B383@metaswitch' of Request 103: Match Not Found [May 17 10:02:58] DEBUG[7809]: chan_sip.c:11656 handle_response_invite: SIP response 200 to RE-invite on outgoing call C371B383@metaswitch Found RTP audio format 0 Peer audio RTP is at port 10.50.10.13:33758 [May 17 10:02:58] DEBUG[7809]: chan_sip.c:5136 process_sdp: T38 state changed to 0 on channel SIP/8055488091-08d62b90 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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 10.50.10.13:33758 [May 17 10:02:58] DEBUG[7809]: chan_sip.c:5216 process_sdp: We're settling with these formats: 0x4 (ulaw) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:5223 process_sdp: We have an owner, now see if we need to change this call [May 17 10:02:58] DEBUG[7809]: chan_sip.c:3004 update_call_counter: Updating call counter for incoming call [May 17 10:02:58] DEBUG[7809]: chan_sip.c:7933 build_route: build_route: Retaining previous route: [May 17 10:02:58] DEBUG[7809]: chan_sip.c:11789 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [May 17 10:02:58] DEBUG[7809]: chan_sip.c:11794 handle_response_invite: T38 state changed to 0 on channel SIP [May 17 10:02:58] DEBUG[7809]: chan_sip.c:11797 handle_response_invite: T38 state changed to 0 on channel SIP/8055488091-08d62b90 [May 17 10:02:58] DEBUG[7809]: chan_sip.c:5651 reqprep: Strict routing enforced for session C371B383@metaswitch set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.10.2, port 5060 Transmitting (no NAT) to 10.50.10.2:5060: ACK sip:8054584444@10.50.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK43269f4f From: ;tag=as48f37b45 To: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 Contact: Call-ID: C371B383@metaswitch CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [May 17 10:02:58] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SIP/5480002 - state 1 (Not in use) [May 17 10:02:58] DEBUG[7892]: app_queue.c:546 changethread: Device 'SIP/5480002' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 10.50.10.177:5060 ---> SIP/2.0 100 Trying Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 CSeq: 103 INVITE From: "WIRELESS CALLER" ;tag=as294ff023 To: ;tag=3acb42d788a1ba4 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK0e3b708a;rport Content-Length: 0 User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 100 Trying (18) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 (54) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 103 INVITE (16) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: "WIRELESS CALLER" ;tag=as294ff023 (68) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: ;tag=3acb42d788a1ba4 (50) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK0e3b708a;rport (63) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [May 17 10:02:58] DEBUG[7809]: chan_sip.c:2123 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #129 - INVITE (got response) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:2132 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '368c400d429a7fb14282ed567649675d@10.50.10.171' Request 103: Found [May 17 10:02:58] DEBUG[7809]: chan_sip.c:11656 handle_response_invite: SIP response 100 to RE-invite on outgoing call 368c400d429a7fb14282ed567649675d@10.50.10.171 <--- SIP read from 10.50.10.177:5060 ---> SIP/2.0 200 OK Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 CSeq: 103 INVITE From: "WIRELESS CALLER" ;tag=as294ff023 To: ;tag=3acb42d788a1ba4 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK0e3b708a;rport Content-Length: 253 Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Supported: replaces Contact: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 725269576 IN IP4 10.50.10.177 s=SIP Call c=IN IP4 10.50.10.177 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=silenceSupp:off - - - - <-------------> [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 200 OK (14) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 (54) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 103 INVITE (16) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: "WIRELESS CALLER" ;tag=as294ff023 (68) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: ;tag=3acb42d788a1ba4 (50) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK0e3b708a;rport (63) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 253 (19) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Allow-Events: talk,hold,conference (34) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO (53) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: Content-Type: application/sdp (29) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: Supported: replaces (19) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: Contact: (35) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 12: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 13: (0) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: v=0 (3) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: o=MxSIP 0 725269576 IN IP4 10.50.10.177 (39) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: s=SIP Call (10) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: c=IN IP4 10.50.10.177 (21) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: t=0 0 (5) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: m=audio 3000 RTP/AVP 0 8 101 (28) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=fmtp:101 0-15 (15) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=ptime:20 (10) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=silenceSupp:off - - - - (25) --- (13 headers 12 lines) --- [May 17 10:02:58] DEBUG[7809]: chan_sip.c:2072 __sip_ack: Acked pending invite 103 [May 17 10:02:58] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '368c400d429a7fb14282ed567649675d@10.50.10.171' of Request 103: Match Not Found [May 17 10:02:58] DEBUG[7809]: chan_sip.c:11656 handle_response_invite: SIP response 200 to RE-invite on outgoing call 368c400d429a7fb14282ed567649675d@10.50.10.171 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.50.10.177:3000 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [May 17 10:02:58] DEBUG[7809]: chan_sip.c:5136 process_sdp: T38 state changed to 0 on channel SIP/5480002-08d5e090 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.50.10.177:3000 [May 17 10:02:58] DEBUG[7809]: chan_sip.c:5216 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [May 17 10:02:58] DEBUG[7809]: chan_sip.c:5223 process_sdp: We have an owner, now see if we need to change this call [May 17 10:02:58] DEBUG[7809]: chan_sip.c:3004 update_call_counter: Updating call counter for outgoing call [May 17 10:02:58] DEBUG[7809]: chan_sip.c:7933 build_route: build_route: Retaining previous route: [May 17 10:02:58] DEBUG[7809]: chan_sip.c:11789 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [May 17 10:02:58] DEBUG[7809]: chan_sip.c:11794 handle_response_invite: T38 state changed to 0 on channel SIP [May 17 10:02:58] DEBUG[7809]: chan_sip.c:11797 handle_response_invite: T38 state changed to 0 on channel SIP/5480002-08d5e090 [May 17 10:02:58] DEBUG[7809]: chan_sip.c:5651 reqprep: Strict routing enforced for session 368c400d429a7fb14282ed567649675d@10.50.10.171 set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.10.177, port 5060 Transmitting (no NAT) to 10.50.10.177:5060: ACK sip:5480002@10.50.10.177 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK46809366;rport From: "WIRELESS CALLER" ;tag=as294ff023 To: ;tag=3acb42d788a1ba4 Contact: Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- <--- SIP read from 10.50.10.2:5060 ---> BYE sip:8055480617@10.50.10.171 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.2:5060;rport;branch=z9hG4bK-59e262de6f05ecab1a0e3e04328cd2ca-metaswitch-1 Allow-Events: message-summary Allow-Events: refer Allow-Events: dialog Allow-Events: line-seize Max-Forwards: 70 Call-ID: C371B383@metaswitch From: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 To: ;tag=as48f37b45 CSeq: 869046972 BYE Organization: Supported: 100rel Content-Length: 0 <-------------> [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: BYE sip:8055480617@10.50.10.171 SIP/2.0 (39) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.10.2:5060;rport;branch=z9hG4bK-59e262de6f05ecab1a0e3e04328cd2ca-metaswitch-1 (99) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: Allow-Events: message-summary (29) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: Allow-Events: refer (19) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: Allow-Events: dialog (20) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Allow-Events: line-seize (24) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Max-Forwards: 70 (16) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Call-ID: C371B383@metaswitch (28) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: From: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 (99) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: To: ;tag=as48f37b45 (48) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: CSeq: 869046972 BYE (19) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: Organization: (14) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 12: Supported: 100rel (17) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 13: Content-Length: 0 (17) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 14: (0) --- (14 headers 0 lines) --- [May 17 10:03:05] DEBUG[7809]: chan_sip.c:14709 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 10.50.10.2 : 5060 (NAT) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:1634 sip_alreadygone: Setting SIP_ALREADYGONE on dialog C371B383@metaswitch [May 17 10:03:05] DEBUG[7809]: chan_sip.c:14263 handle_request_bye: Received bye, issuing owner hangup <--- Transmitting (NAT) to 10.50.10.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.10.2:5060;rport;branch=z9hG4bK-59e262de6f05ecab1a0e3e04328cd2ca-metaswitch-1;received=10.50.10.2 From: WIRELESS CALLER ;tag=metaswitch+1+25591f+1b539c6e;isup-oli=62 To: ;tag=as48f37b45 Call-ID: C371B383@metaswitch CSeq: 869046972 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [May 17 10:03:05] DEBUG[7875]: rtp.c:2886 bridge_native_loop: Oooh, got a hangup [May 17 10:03:05] DEBUG[7875]: chan_sip.c:16970 sip_set_rtp_peer: Sending reinvite on SIP '368c400d429a7fb14282ed567649675d@10.50.10.171' - It's audio soon redirected to IP 10.50.10.171 [May 17 10:03:05] DEBUG[7875]: chan_sip.c:5651 reqprep: Strict routing enforced for session 368c400d429a7fb14282ed567649675d@10.50.10.171 set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.10.177, port 5060 [May 17 10:03:05] DEBUG[7875]: chan_sip.c:6198 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [May 17 10:03:05] DEBUG[7875]: chan_sip.c:6199 add_sdp: ** Our prefcodec: 0x4 (ulaw) Audio is at 10.50.10.171 port 13994 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [May 17 10:03:05] DEBUG[7875]: chan_sip.c:6330 add_sdp: -- Done with adding codecs to SDP [May 17 10:03:05] DEBUG[7875]: chan_sip.c:6375 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:1622 initialize_initreq: Initializing already initialized SIP dialog 368c400d429a7fb14282ed567649675d@10.50.10.171 (presumably reinvite) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 0: INVITE sip:5480002@10.50.10.177 SIP/2.0 (39) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK79489821;rport (63) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 2: From: "WIRELESS CALLER" ;tag=as294ff023 (68) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 3: To: ;tag=3acb42d788a1ba4 (50) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 4: Contact: (38) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 5: Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 (54) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 6: CSeq: 104 INVITE (16) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 7: User-Agent: Asterisk PBX (24) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 8: Max-Forwards: 70 (16) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 10: Supported: replaces (19) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 12: Content-Type: application/sdp (29) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 13: Content-Length: 262 (19) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4577 parse_request: Header 14: (0) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: v=0 (3) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: o=root 7783 7785 IN IP4 10.50.10.171 (36) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: s=session (9) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: c=IN IP4 10.50.10.171 (21) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: t=0 0 (5) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: m=audio 13994 RTP/AVP 0 8 101 (29) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=fmtp:101 0-16 (15) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=silenceSupp:off - - - - (25) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=ptime:20 (10) [May 17 10:03:05] DEBUG[7875]: chan_sip.c:4609 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 10.50.10.177:5060: INVITE sip:5480002@10.50.10.177 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK79489821;rport From: "WIRELESS CALLER" ;tag=as294ff023 To: ;tag=3acb42d788a1ba4 Contact: Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 CSeq: 104 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 262 v=0 o=root 7783 7785 IN IP4 10.50.10.171 s=session c=IN IP4 10.50.10.171 t=0 0 m=audio 13994 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [May 17 10:03:05] DEBUG[7875]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #130 [May 17 10:03:05] DEBUG[7875]: channel.c:4115 ast_channel_bridge: Returning from native bridge, channels: SIP/8055488091-08d62b90, SIP/5480002-08d5e090 [May 17 10:03:05] DEBUG[7875]: channel.c:1726 ast_hangup: Hanging up channel 'SIP/5480002-08d5e090' [May 17 10:03:05] DEBUG[7875]: chan_sip.c:3313 sip_hangup: Hangup call SIP/5480002-08d5e090, SIP callid 368c400d429a7fb14282ed567649675d@10.50.10.171) Scheduling destruction of SIP dialog '368c400d429a7fb14282ed567649675d@10.50.10.171' in 32000 ms (Method: INVITE) [May 17 10:03:05] DEBUG[7875]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/5480002-08d5e090 [May 17 10:03:05] DEBUG[7787]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5480002 [May 17 10:03:05] DEBUG[7787]: chan_sip.c:15321 sip_devicestate: Checking device state for peer 5480002 [May 17 10:03:05] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SIP/5480002 - state 1 (Not in use) [May 17 10:03:05] DEBUG[7875]: rtp.c:1476 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [May 17 10:03:05] DEBUG[7875]: app_dial.c:1679 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [May 17 10:03:05] DEBUG[7875]: pbx.c:2407 __ast_pbx_run: Spawn extension (line2,8055488092,3) exited non-zero on 'SIP/8055488091-08d62b90' == Spawn extension (line2, 8055488092, 3) exited non-zero on 'SIP/8055488091-08d62b90' [May 17 10:03:05] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '"WIRELESS CALLER" <8054584444>' [May 17 10:03:05] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '8054584444' [May 17 10:03:05] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '8055488092' [May 17 10:03:05] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'line2' [May 17 10:03:05] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'SIP/8055488091-08d62b90' [May 17 10:03:05] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'SIP/5480002-08d5e090' [May 17 10:03:05] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'Dial' [May 17 10:03:05] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'SIP/5480002|10' [May 17 10:03:05] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:02:39' [May 17 10:03:05] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:02:39' [May 17 10:03:05] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:03:05' [May 17 10:03:05] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '26' [May 17 10:03:05] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '26' [May 17 10:03:05] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [May 17 10:03:05] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [May 17 10:03:05] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:05] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '1179421359.9' [May 17 10:03:05] DEBUG[7875]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:05] DEBUG[7875]: channel.c:1726 ast_hangup: Hanging up channel 'SIP/8055488091-08d62b90' [May 17 10:03:05] DEBUG[7875]: chan_sip.c:3313 sip_hangup: Hangup call SIP/8055488091-08d62b90, SIP callid C371B383@metaswitch) [May 17 10:03:05] DEBUG[7875]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/8055488091-08d62b90 [May 17 10:03:05] DEBUG[7787]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 8055488091 [May 17 10:03:05] DEBUG[7787]: chan_sip.c:15321 sip_devicestate: Checking device state for peer 8055488091 [May 17 10:03:05] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SIP/8055488091 - state 1 (Not in use) [May 17 10:03:05] DEBUG[7894]: app_queue.c:546 changethread: Device 'SIP/8055488091' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 17 10:03:05] DEBUG[7893]: app_queue.c:546 changethread: Device 'SIP/5480002' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 10.50.10.177:5060 ---> SIP/2.0 100 Trying Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 CSeq: 104 INVITE From: "WIRELESS CALLER" ;tag=as294ff023 To: ;tag=3acb42d788a1ba4 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK79489821;rport Content-Length: 0 User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 100 Trying (18) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 (54) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 104 INVITE (16) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: "WIRELESS CALLER" ;tag=as294ff023 (68) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: ;tag=3acb42d788a1ba4 (50) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK79489821;rport (63) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [May 17 10:03:05] DEBUG[7809]: chan_sip.c:2123 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #130 - INVITE (got response) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:2132 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '368c400d429a7fb14282ed567649675d@10.50.10.171' Request 104: Found [May 17 10:03:05] DEBUG[7809]: chan_sip.c:11658 handle_response_invite: SIP response 100 to standard invite [May 17 10:03:05] DEBUG[7809]: chan_sip.c:5651 reqprep: Strict routing enforced for session 368c400d429a7fb14282ed567649675d@10.50.10.171 set_destination: Parsing for address/port to send to set_destination: set destination to 10.50.10.177, port 5060 Reliably Transmitting (no NAT) to 10.50.10.177:5060: BYE sip:5480002@10.50.10.177 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK08257f28;rport From: "WIRELESS CALLER" ;tag=as294ff023 To: ;tag=3acb42d788a1ba4 Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 CSeq: 105 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [May 17 10:03:05] DEBUG[7809]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #132 Scheduling destruction of SIP dialog '368c400d429a7fb14282ed567649675d@10.50.10.171' in 32000 ms (Method: INVITE) Really destroying SIP dialog 'C371B383@metaswitch' Method: BYE <--- SIP read from 10.50.10.177:5060 ---> SIP/2.0 200 OK Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 CSeq: 104 INVITE From: "WIRELESS CALLER" ;tag=as294ff023 To: ;tag=3acb42d788a1ba4 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK79489821;rport Content-Length: 253 Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Supported: replaces Contact: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 725269576 IN IP4 10.50.10.177 s=SIP Call c=IN IP4 10.50.10.177 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=silenceSupp:off - - - - <-------------> [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 200 OK (14) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 (54) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 104 INVITE (16) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: "WIRELESS CALLER" ;tag=as294ff023 (68) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: ;tag=3acb42d788a1ba4 (50) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK79489821;rport (63) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 253 (19) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Allow-Events: talk,hold,conference (34) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO (53) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: Content-Type: application/sdp (29) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: Supported: replaces (19) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: Contact: (35) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 12: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 13: (0) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: v=0 (3) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: o=MxSIP 0 725269576 IN IP4 10.50.10.177 (39) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: s=SIP Call (10) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: c=IN IP4 10.50.10.177 (21) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: t=0 0 (5) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: m=audio 3000 RTP/AVP 0 8 101 (28) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=fmtp:101 0-15 (15) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=ptime:20 (10) [May 17 10:03:05] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=silenceSupp:off - - - - (25) --- (13 headers 12 lines) --- <--- SIP read from 10.50.10.177:5060 ---> SIP/2.0 200 OK Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 CSeq: 105 BYE From: "WIRELESS CALLER" ;tag=as294ff023 To: ;tag=3acb42d788a1ba4 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK08257f28;rport Content-Length: 0 Supported: replaces User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 200 OK (14) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 (54) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 105 BYE (13) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: "WIRELESS CALLER" ;tag=as294ff023 (68) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: ;tag=3acb42d788a1ba4 (50) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK08257f28;rport (63) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Supported: replaces (19) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [May 17 10:03:06] DEBUG[7809]: chan_sip.c:2080 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #132 [May 17 10:03:06] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '368c400d429a7fb14282ed567649675d@10.50.10.171' of Request 105: Match Not Found <--- SIP read from 10.50.10.177:5060 ---> SIP/2.0 200 OK Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 CSeq: 104 INVITE From: "WIRELESS CALLER" ;tag=as294ff023 To: ;tag=3acb42d788a1ba4 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK79489821;rport Content-Length: 253 Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Supported: replaces Contact: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 725269576 IN IP4 10.50.10.177 s=SIP Call c=IN IP4 10.50.10.177 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=silenceSupp:off - - - - <-------------> [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 200 OK (14) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 (54) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 104 INVITE (16) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: "WIRELESS CALLER" ;tag=as294ff023 (68) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: ;tag=3acb42d788a1ba4 (50) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK79489821;rport (63) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 253 (19) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Allow-Events: talk,hold,conference (34) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO (53) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: Content-Type: application/sdp (29) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: Supported: replaces (19) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: Contact: (35) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 12: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 13: (0) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: v=0 (3) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: o=MxSIP 0 725269576 IN IP4 10.50.10.177 (39) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: s=SIP Call (10) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: c=IN IP4 10.50.10.177 (21) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: t=0 0 (5) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: m=audio 3000 RTP/AVP 0 8 101 (28) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=fmtp:101 0-15 (15) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=ptime:20 (10) [May 17 10:03:06] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=silenceSupp:off - - - - (25) --- (13 headers 12 lines) --- <--- SIP read from 10.50.10.177:5060 ---> SIP/2.0 200 OK Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 CSeq: 104 INVITE From: "WIRELESS CALLER" ;tag=as294ff023 To: ;tag=3acb42d788a1ba4 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK79489821;rport Content-Length: 253 Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Supported: replaces Contact: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 725269576 IN IP4 10.50.10.177 s=SIP Call c=IN IP4 10.50.10.177 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=silenceSupp:off - - - - <-------------> [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 200 OK (14) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 (54) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 104 INVITE (16) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: "WIRELESS CALLER" ;tag=as294ff023 (68) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: ;tag=3acb42d788a1ba4 (50) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK79489821;rport (63) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 253 (19) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Allow-Events: talk,hold,conference (34) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO (53) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: Content-Type: application/sdp (29) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: Supported: replaces (19) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: Contact: (35) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 12: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 13: (0) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: v=0 (3) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: o=MxSIP 0 725269576 IN IP4 10.50.10.177 (39) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: s=SIP Call (10) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: c=IN IP4 10.50.10.177 (21) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: t=0 0 (5) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: m=audio 3000 RTP/AVP 0 8 101 (28) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=fmtp:101 0-15 (15) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=ptime:20 (10) [May 17 10:03:07] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=silenceSupp:off - - - - (25) --- (13 headers 12 lines) --- <--- SIP read from 10.50.10.177:5060 ---> SIP/2.0 200 OK Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 CSeq: 104 INVITE From: "WIRELESS CALLER" ;tag=as294ff023 To: ;tag=3acb42d788a1ba4 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK79489821;rport Content-Length: 253 Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Supported: replaces Contact: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 725269576 IN IP4 10.50.10.177 s=SIP Call c=IN IP4 10.50.10.177 t=0 0 m=audio 3000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=silenceSupp:off - - - - <-------------> [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 200 OK (14) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 368c400d429a7fb14282ed567649675d@10.50.10.171 (54) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 104 INVITE (16) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: "WIRELESS CALLER" ;tag=as294ff023 (68) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: ;tag=3acb42d788a1ba4 (50) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK79489821;rport (63) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 253 (19) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Allow-Events: talk,hold,conference (34) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO (53) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: Content-Type: application/sdp (29) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: Supported: replaces (19) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: Contact: (35) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 12: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 13: (0) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: v=0 (3) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: o=MxSIP 0 725269576 IN IP4 10.50.10.177 (39) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: s=SIP Call (10) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: c=IN IP4 10.50.10.177 (21) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: t=0 0 (5) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: m=audio 3000 RTP/AVP 0 8 101 (28) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=fmtp:101 0-15 (15) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=ptime:20 (10) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=silenceSupp:off - - - - (25) --- (13 headers 12 lines) --- <--- SIP read from 10.50.10.2:5060 ---> OPTIONS sip:metaswitch@10.50.10.171:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.10.2:5060;rport;branch=z9hG4bK-2056aa0cdeed658fd709aa3ed3dde856-metaswitch-1 Allow-Events: message-summary Allow-Events: refer Allow-Events: dialog Allow-Events: line-seize Max-Forwards: 70 Call-ID: 4BADF9E0@metaswitch From: ;tag=metaswitch+1+0+47dfb834 CSeq: 7714547 OPTIONS Organization: Supported: 100rel Content-Length: 0 Contact: To: <-------------> [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: OPTIONS sip:metaswitch@10.50.10.171:5060;transport=udp SIP/2.0 (62) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.10.2:5060;rport;branch=z9hG4bK-2056aa0cdeed658fd709aa3ed3dde856-metaswitch-1 (99) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: Allow-Events: message-summary (29) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: Allow-Events: refer (19) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: Allow-Events: dialog (20) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Allow-Events: line-seize (24) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Max-Forwards: 70 (16) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Call-ID: 4BADF9E0@metaswitch (28) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: From: ;tag=metaswitch+1+0+47dfb834 (66) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: CSeq: 7714547 OPTIONS (21) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: Organization: (14) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: Supported: 100rel (17) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 12: Content-Length: 0 (17) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 13: Contact: (41) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 14: To: (33) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 15: (0) --- (15 headers 0 lines) --- [May 17 10:03:09] DEBUG[7809]: chan_sip.c:4312 sip_alloc: Allocating new SIP dialog for 4BADF9E0@metaswitch - OPTIONS (No RTP) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:14709 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for metaswitch in default (domain 10.50.10.171) <--- Transmitting (no NAT) to 10.50.10.2:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.50.10.2:5060;rport;branch=z9hG4bK-2056aa0cdeed658fd709aa3ed3dde856-metaswitch-1;received=10.50.10.2 From: ;tag=metaswitch+1+0+47dfb834 To: ;tag=as06513b3f Call-ID: 4BADF9E0@metaswitch CSeq: 7714547 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '4BADF9E0@metaswitch' in 32000 ms (Method: OPTIONS) [May 17 10:03:09] DEBUG[7809]: chan_sip.c:14927 sipsock_read: SIP message could not be handled, bad request: 4BADF9E0@metaswitch <--- SIP read from 10.50.10.183:5060 ---> INVITE sip:station2_line1@10.50.10.171:5060 SIP/2.0 Max-Forwards: 70 Content-Length: 290 Via: SIP/2.0/UDP 10.50.10.183;branch=z9hG4bKe23397219 Call-ID: 3428005a72b7241c9648d73e6c486076@10.50.10.183 From: Trunk01 ;tag=355161e0eb78137 To: station2_line1 CSeq: 1682996280 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Contact: Trunk01 Supported: replaces User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 v=0 o=MxSIP 0 796738348 IN IP4 10.50.10.183 s=SIP Call c=IN IP4 10.50.10.183 t=0 0 m=audio 3000 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=silenceSupp:on - - - - <-------------> [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: INVITE sip:station2_line1@10.50.10.171:5060 SIP/2.0 (51) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Max-Forwards: 70 (16) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: Content-Length: 290 (19) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: Via: SIP/2.0/UDP 10.50.10.183;branch=z9hG4bKe23397219 (53) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: Call-ID: 3428005a72b7241c9648d73e6c486076@10.50.10.183 (54) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: From: Trunk01 ;tag=355161e0eb78137 (66) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: To: station2_line1 (57) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: CSeq: 1682996280 INVITE (23) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Supported: timer (16) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: Allow-Events: talk,hold,conference (34) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO (53) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: Content-Type: application/sdp (29) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 12: Contact: Trunk01 (44) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 13: Supported: replaces (19) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 14: User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 (70) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 15: (0) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: v=0 (3) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: o=MxSIP 0 796738348 IN IP4 10.50.10.183 (39) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: s=SIP Call (10) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: c=IN IP4 10.50.10.183 (21) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: t=0 0 (5) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: m=audio 3000 RTP/AVP 0 8 18 101 (31) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:18 G729/8000 (21) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=fmtp:101 0-15 (15) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=ptime:30 (10) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=sendrecv (10) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=silenceSupp:on - - - - (24) --- (15 headers 14 lines) --- [May 17 10:03:11] DEBUG[7809]: chan_sip.c:2576 do_setnat: Setting NAT on RTP to Off [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4312 sip_alloc: Allocating new SIP dialog for 3428005a72b7241c9648d73e6c486076@10.50.10.183 - INVITE (With RTP) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:14709 handle_request: **** Received INVITE (5) - Command in SIP INVITE [May 17 10:03:11] DEBUG[7809]: chan_sip.c:1681 parse_sip_options: Begin: parsing SIP "Supported: timer" [May 17 10:03:11] DEBUG[7809]: chan_sip.c:1689 parse_sip_options: Found SIP option: -timer- [May 17 10:03:11] DEBUG[7809]: chan_sip.c:1695 parse_sip_options: Matched SIP option: timer Sending to 10.50.10.183 : 5060 (no NAT) Using INVITE request as basis request - 3428005a72b7241c9648d73e6c486076@10.50.10.183 [May 17 10:03:11] DEBUG[7809]: chan_sip.c:2576 do_setnat: Setting NAT on RTP to Off <--- Reliably Transmitting (no NAT) to 10.50.10.183:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.50.10.183;branch=z9hG4bKe23397219;received=10.50.10.183 From: Trunk01 ;tag=355161e0eb78137 To: station2_line1 ;tag=as0b648b37 Call-ID: 3428005a72b7241c9648d73e6c486076@10.50.10.183 CSeq: 1682996280 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="41c7ea9f" Content-Length: 0 <------------> [May 17 10:03:11] DEBUG[7809]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #135 Scheduling destruction of SIP dialog '3428005a72b7241c9648d73e6c486076@10.50.10.183' in 32000 ms (Method: INVITE) Found user 'station2' <--- SIP read from 10.50.10.183:5060 ---> ACK sip:station2_line1@10.50.10.171:5060 SIP/2.0 Max-Forwards: 70 Content-Length: 0 Via: SIP/2.0/UDP 10.50.10.183;branch=z9hG4bKe23397219 Call-ID: 3428005a72b7241c9648d73e6c486076@10.50.10.183 From: Trunk01 ;tag=355161e0eb78137 To: station2_line1 ;tag=as0b648b37 CSeq: 1682996280 ACK User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 <-------------> [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: ACK sip:station2_line1@10.50.10.171:5060 SIP/2.0 (48) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Max-Forwards: 70 (16) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: Content-Length: 0 (17) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: Via: SIP/2.0/UDP 10.50.10.183;branch=z9hG4bKe23397219 (53) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: Call-ID: 3428005a72b7241c9648d73e6c486076@10.50.10.183 (54) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: From: Trunk01 ;tag=355161e0eb78137 (66) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: To: station2_line1 ;tag=as0b648b37 (72) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: CSeq: 1682996280 ACK (20) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 (70) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [May 17 10:03:11] DEBUG[7809]: chan_sip.c:14709 handle_request: **** Received ACK (6) - Command in SIP ACK [May 17 10:03:11] DEBUG[7809]: chan_sip.c:2080 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #135 [May 17 10:03:11] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '3428005a72b7241c9648d73e6c486076@10.50.10.183' of Response 1682996280: Match Not Found <--- SIP read from 10.50.10.183:5060 ---> INVITE sip:station2_line1@10.50.10.171:5060 SIP/2.0 Max-Forwards: 70 Content-Length: 290 Via: SIP/2.0/UDP 10.50.10.183;branch=z9hG4bKba80ab6ea Call-ID: 3428005a72b7241c9648d73e6c486076@10.50.10.183 From: Trunk01 ;tag=355161e0eb78137 To: station2_line1 CSeq: 1682996281 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Supported: replaces Proxy-Authorization:Digest response="0f91a4ecc23e5061c32aa10c6781e736",username="station2",realm="asterisk",nonce="41c7ea9f",algorithm=MD5,uri="sip:station2_line1@10.50.10.171:5060" Contact: Trunk01 User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 v=0 o=MxSIP 0 796738348 IN IP4 10.50.10.183 s=SIP Call c=IN IP4 10.50.10.183 t=0 0 m=audio 3000 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=silenceSupp:on - - - - <-------------> [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: INVITE sip:station2_line1@10.50.10.171:5060 SIP/2.0 (51) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Max-Forwards: 70 (16) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: Content-Length: 290 (19) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: Via: SIP/2.0/UDP 10.50.10.183;branch=z9hG4bKba80ab6ea (53) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: Call-ID: 3428005a72b7241c9648d73e6c486076@10.50.10.183 (54) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: From: Trunk01 ;tag=355161e0eb78137 (66) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: To: station2_line1 (57) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: CSeq: 1682996281 INVITE (23) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Supported: timer (16) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: Allow-Events: talk,hold,conference (34) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO (53) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: Content-Type: application/sdp (29) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 12: Supported: replaces (19) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 13: Proxy-Authorization:Digest response="0f91a4ecc23e5061c32aa10c6781e736",username="station2",realm="asterisk",nonce="41c7ea9f",algorithm=MD5,uri="sip:station2_line1@10.50.10.171:5060" (181) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 14: Contact: Trunk01 (44) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 15: User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 (70) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 16: (0) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: v=0 (3) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: o=MxSIP 0 796738348 IN IP4 10.50.10.183 (39) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: s=SIP Call (10) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: c=IN IP4 10.50.10.183 (21) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: t=0 0 (5) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: m=audio 3000 RTP/AVP 0 8 18 101 (31) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:18 G729/8000 (21) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=fmtp:101 0-15 (15) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=ptime:30 (10) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=sendrecv (10) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:4609 parse_request: Line: a=silenceSupp:on - - - - (24) --- (16 headers 14 lines) --- [May 17 10:03:11] DEBUG[7809]: chan_sip.c:14709 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 10.50.10.183 : 5060 (no NAT) Using INVITE request as basis request - 3428005a72b7241c9648d73e6c486076@10.50.10.183 [May 17 10:03:11] DEBUG[7809]: chan_sip.c:2576 do_setnat: Setting NAT on RTP to Off Found user 'station2' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.50.10.183:3000 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [May 17 10:03:11] DEBUG[7809]: chan_sip.c:5136 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.50.10.183:3000 [May 17 10:03:11] DEBUG[7809]: chan_sip.c:5216 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:13418 handle_request_invite: Checking SIP call limits for device station2 [May 17 10:03:11] DEBUG[7809]: chan_sip.c:3004 update_call_counter: Updating call counter for incoming call Looking for station2_line1 in sla_stations (domain 10.50.10.171) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:3806 sip_new: *** Our native formats are 0x4 (ulaw) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:3807 sip_new: *** Joint capabilities are 0xc (ulaw|alaw) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:3808 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:3809 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [May 17 10:03:11] DEBUG[7809]: chan_sip.c:3832 sip_new: This channel will not be able to handle video. [May 17 10:03:11] DEBUG[7809]: chan_sip.c:7994 build_route: build_route: Contact hop: Trunk01 list_route: hop: [May 17 10:03:11] DEBUG[7809]: chan_sip.c:13492 handle_request_invite: SIP/station2-08d66e30: New call is still down.... Trying... <--- Transmitting (no NAT) to 10.50.10.183:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.50.10.183;branch=z9hG4bKba80ab6ea;received=10.50.10.183 From: Trunk01 ;tag=355161e0eb78137 To: station2_line1 Call-ID: 3428005a72b7241c9648d73e6c486076@10.50.10.183 CSeq: 1682996281 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [May 17 10:03:11] DEBUG[7809]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/station2-08d66e30 [May 17 10:03:11] DEBUG[7787]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - station2 [May 17 10:03:11] DEBUG[7787]: chan_sip.c:15321 sip_devicestate: Checking device state for peer station2 [May 17 10:03:11] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SIP/station2 - state 1 (Not in use) [May 17 10:03:11] DEBUG[7895]: pbx.c:1809 pbx_extension_helper: Launching 'SLAStation' -- Executing [station2_line1@sla_stations:1] SLAStation("SIP/station2-08d66e30", "station2_line1") in new stack [May 17 10:03:11] DEBUG[7895]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SLA:station1_line1 [May 17 10:03:11] DEBUG[7895]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SLA:station2_line1 [May 17 10:03:11] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station1_line1 [May 17 10:03:11] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SLA:station1_line1 - state 2 (In use) [May 17 10:03:11] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station1_line1 [May 17 10:03:11] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station1_line1 Reliably Transmitting (no NAT) to 10.50.10.177:5060: NOTIFY sip:station1@10.50.10.177 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK1a5432d6;rport From: Line 1 ;tag=as4404e61b To: Trunk01 ;tag=a3a3bd7906dfaf5 Contact: Call-ID: 0871b9d69a4a057cf9138a1a7ebb36ad@10.50.10.177 CSeq: 105 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 229

confirmed

--- [May 17 10:03:11] DEBUG[7787]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #137 Extension Changed station1_line1 new state InUse for Notify User station1 [May 17 10:03:11] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station2_line1 [May 17 10:03:11] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SLA:station2_line1 - state 2 (In use) [May 17 10:03:11] DEBUG[7896]: app_queue.c:546 changethread: Device 'SIP/station2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 17 10:03:11] DEBUG[7897]: channel.c:3376 ast_channel_inherit_variables: Not copying variable STACK-sla_stations-station2_line1-1. [May 17 10:03:11] DEBUG[7897]: channel.c:3376 ast_channel_inherit_variables: Not copying variable SIPCALLID. [May 17 10:03:11] DEBUG[7897]: channel.c:3376 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [May 17 10:03:11] DEBUG[7897]: channel.c:3376 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [May 17 10:03:11] DEBUG[7897]: channel.c:3376 ast_channel_inherit_variables: Not copying variable SIPURI. -- Called disa@line1_outbound [May 17 10:03:11] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station2_line1 [May 17 10:03:11] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station2_line1 Reliably Transmitting (no NAT) to 10.50.10.183:5060: NOTIFY sip:station2@10.50.10.183 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK5b868075;rport From: sip:station2_line1@10.50.10.171:5060;tag=as0a72300c To: Trunk01 ;tag=db0222544e54736 Contact: Call-ID: 0135d7249cd406b94e41bc097f840b5f@10.50.10.183 CSeq: 105 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 229

confirmed

--- [May 17 10:03:11] DEBUG[7787]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #138 Extension Changed station2_line1 new state InUse for Notify User station2 [May 17 10:03:11] DEBUG[7900]: pbx.c:1809 pbx_extension_helper: Launching 'DISA' -- Executing [disa@line1_outbound:1] DISA("Local/disa@line1_outbound-933b,2", "no-password|line1_outbound") in new stack [May 17 10:03:11] DEBUG[7900]: app_disa.c:157 disa_exec: Digittimeout: 5000 <--- SIP read from 10.50.10.177:5060 ---> SIP/2.0 200 OK Call-ID: 0871b9d69a4a057cf9138a1a7ebb36ad@10.50.10.177 CSeq: 105 NOTIFY From: Line 1 ;tag=as4404e61b To: Trunk01 ;tag=a3a3bd7906dfaf5 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK1a5432d6;rport Content-Length: 0 Contact: Trunk01 Supported: replaces User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 200 OK (14) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 0871b9d69a4a057cf9138a1a7ebb36ad@10.50.10.177 (54) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 105 NOTIFY (16) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: Line 1 ;tag=as4404e61b (66) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: Trunk01 ;tag=a3a3bd7906dfaf5 (64) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK1a5432d6;rport (63) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Contact: Trunk01 (44) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Supported: replaces (19) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [May 17 10:03:12] DEBUG[7809]: chan_sip.c:2080 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #137 [May 17 10:03:12] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '0871b9d69a4a057cf9138a1a7ebb36ad@10.50.10.177' of Request 105: Match Not Found SIP Response message for INCOMING dialog NOTIFY arrived [May 17 10:03:12] DEBUG[7900]: app_disa.c:158 disa_exec: Responsetimeout: 10000 [May 17 10:03:12] DEBUG[7900]: app_disa.c:169 disa_exec: Mailbox: [May 17 10:03:12] DEBUG[7900]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel Local/disa@line1_outbound-933b,2 [May 17 10:03:12] DEBUG[7787]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for Local - disa@line1_outbound [May 17 10:03:12] DEBUG[7787]: chan_local.c:145 local_devicestate: Checking if extension disa@line1_outbound exists (devicestate) [May 17 10:03:12] DEBUG[7787]: channel.c:1057 channel_find_locked: Avoiding initial deadlock for channel '0x8d687a0' [May 17 10:03:12] DEBUG[7900]: app_disa.c:186 disa_exec: Context: line1_outbound [May 17 10:03:12] DEBUG[7900]: app_disa.c:190 disa_exec: DISA no-password login success [May 17 10:03:12] DEBUG[7900]: channel.c:2911 set_format: Set channel Local/disa@line1_outbound-933b,2 to write format slin [May 17 10:03:12] DEBUG[7900]: channel.c:2030 ast_settimeout: Scheduling timer at 160 sample intervals [May 17 10:03:12] DEBUG[7901]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel Local/disa@line1_outbound-933b,1 <--- SIP read from 10.50.10.183:5060 ---> SIP/2.0 200 OK Call-ID: 0135d7249cd406b94e41bc097f840b5f@10.50.10.183 CSeq: 105 NOTIFY From: sip:station2_line1@10.50.10.171:5060;tag=as0a72300c To: Trunk01 ;tag=db0222544e54736 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK5b868075;rport Content-Length: 0 Contact: Trunk01 Supported: replaces User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 <-------------> [May 17 10:03:12] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for Local/disa@line1_outbound - state 2 (In use) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 200 OK (14) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 0135d7249cd406b94e41bc097f840b5f@10.50.10.183 (54) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 105 NOTIFY (16) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: sip:station2_line1@10.50.10.171:5060;tag=as0a72300c (57) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: Trunk01 ;tag=db0222544e54736 (59) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK5b868075;rport (63) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Contact: Trunk01 (44) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Supported: replaces (19) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 (70) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [May 17 10:03:12] DEBUG[7809]: chan_sip.c:2080 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #138 [May 17 10:03:12] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '0135d7249cd406b94e41bc097f840b5f@10.50.10.183' of Request 105: Match Not Found SIP Response message for INCOMING dialog NOTIFY arrived [May 17 10:03:12] DEBUG[7902]: app_queue.c:546 changethread: Device 'Local/disa@line1_outbound' changed to state '2' (In use) but we don't care because they're not a member of any queue. [May 17 10:03:12] DEBUG[7787]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for Local - disa@line1_outbound [May 17 10:03:12] DEBUG[7787]: chan_local.c:145 local_devicestate: Checking if extension disa@line1_outbound exists (devicestate) [May 17 10:03:12] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for Local/disa@line1_outbound - state 2 (In use) [May 17 10:03:12] DEBUG[7903]: app_queue.c:546 changethread: Device 'Local/disa@line1_outbound' changed to state '2' (In use) but we don't care because they're not a member of any queue. -- Local/disa@line1_outbound-933b,1 answered [May 17 10:03:12] DEBUG[7897]: chan_zap.c:7817 zt_request: Using channel -2 [May 17 10:03:12] DEBUG[7897]: channel.c:2911 set_format: Set channel Zap/pseudo-528115053 to read format slin [May 17 10:03:12] DEBUG[7897]: channel.c:2911 set_format: Set channel Zap/pseudo-528115053 to write format slin -- Created MeetMe conference 1023 for conference 'SLA_line1' [May 17 10:03:12] DEBUG[7895]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/station2-08d66e30 [May 17 10:03:12] DEBUG[7895]: chan_sip.c:3464 sip_answer: SIP answering channel: SIP/station2-08d66e30 [May 17 10:03:12] DEBUG[7787]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - station2 [May 17 10:03:12] DEBUG[7895]: chan_sip.c:6432 transmit_response_with_sdp: Setting framing from config on incoming call [May 17 10:03:12] DEBUG[7787]: chan_sip.c:15321 sip_devicestate: Checking device state for peer station2 [May 17 10:03:12] DEBUG[7895]: chan_sip.c:6198 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True [May 17 10:03:12] DEBUG[7895]: chan_sip.c:6199 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 10.50.10.171 port 18710 [May 17 10:03:12] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SIP/station2 - state 1 (Not in use) Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [May 17 10:03:12] DEBUG[7895]: chan_sip.c:6330 add_sdp: -- Done with adding codecs to SDP [May 17 10:03:12] DEBUG[7895]: chan_sip.c:6375 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) <--- Reliably Transmitting (no NAT) to 10.50.10.183:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.10.183;branch=z9hG4bKba80ab6ea;received=10.50.10.183 From: Trunk01 ;tag=355161e0eb78137 To: station2_line1 ;tag=as7e66826b Call-ID: 3428005a72b7241c9648d73e6c486076@10.50.10.183 CSeq: 1682996281 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 262 v=0 o=root 7783 7783 IN IP4 10.50.10.171 s=session c=IN IP4 10.50.10.171 t=0 0 m=audio 18710 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [May 17 10:03:12] DEBUG[7895]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #139 [May 17 10:03:12] DEBUG[7895]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel meetme:SLA_line1 [May 17 10:03:12] DEBUG[7895]: channel.c:2911 set_format: Set channel SIP/station2-08d66e30 to write format slin [May 17 10:03:12] DEBUG[7895]: channel.c:2911 set_format: Set channel SIP/station2-08d66e30 to read format slin [May 17 10:03:12] DEBUG[7895]: app_meetme.c:1641 conf_run: Placed channel SIP/station2-08d66e30 in ZAP conf 1023 [May 17 10:03:12] DEBUG[7897]: channel.c:2911 set_format: Set channel Local/disa@line1_outbound-933b,1 to write format slin [May 17 10:03:12] DEBUG[7897]: channel.c:2911 set_format: Set channel Local/disa@line1_outbound-933b,1 to read format slin [May 17 10:03:12] DEBUG[7897]: app_meetme.c:1641 conf_run: Placed channel Local/disa@line1_outbound-933b,1 in ZAP conf 1023 [May 17 10:03:12] DEBUG[7904]: app_queue.c:546 changethread: Device 'SIP/station2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 17 10:03:12] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "meetme" - number: SLA_line1 [May 17 10:03:12] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for meetme:SLA_line1 - state 2 (In use) [May 17 10:03:12] DEBUG[7895]: rtp.c:2701 ast_rtp_write: Ooh, format changed from unknown to ulaw [May 17 10:03:12] DEBUG[7895]: rtp.c:2718 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [May 17 10:03:12] DEBUG[7900]: channel.c:2398 __ast_read: Generator got voice, switching to phase locked mode [May 17 10:03:12] DEBUG[7900]: channel.c:2030 ast_settimeout: Scheduling timer at 0 sample intervals <--- SIP read from 10.50.10.183:5060 ---> ACK sip:station2_line1@10.50.10.171 SIP/2.0 Max-Forwards: 70 Content-Length: 0 Via: SIP/2.0/UDP 10.50.10.183;branch=z9hG4bK9cba8cc08 Call-ID: 3428005a72b7241c9648d73e6c486076@10.50.10.183 From: Trunk01 ;tag=355161e0eb78137 To: station2_line1 ;tag=as7e66826b CSeq: 1682996281 ACK Proxy-Authorization:Digest response="0f91a4ecc23e5061c32aa10c6781e736",username="station2",realm="asterisk",nonce="41c7ea9f",algorithm=MD5,uri="sip:station2_line1@10.50.10.171:5060" User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 <-------------> [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: ACK sip:station2_line1@10.50.10.171 SIP/2.0 (43) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Max-Forwards: 70 (16) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: Content-Length: 0 (17) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: Via: SIP/2.0/UDP 10.50.10.183;branch=z9hG4bK9cba8cc08 (53) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: Call-ID: 3428005a72b7241c9648d73e6c486076@10.50.10.183 (54) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: From: Trunk01 ;tag=355161e0eb78137 (66) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: To: station2_line1 ;tag=as7e66826b (72) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: CSeq: 1682996281 ACK (20) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Proxy-Authorization:Digest response="0f91a4ecc23e5061c32aa10c6781e736",username="station2",realm="asterisk",nonce="41c7ea9f",algorithm=MD5,uri="sip:station2_line1@10.50.10.171:5060" (181) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 (70) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [May 17 10:03:12] DEBUG[7809]: chan_sip.c:14709 handle_request: **** Received ACK (6) - Command in SIP ACK [May 17 10:03:12] DEBUG[7809]: chan_sip.c:2080 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #139 [May 17 10:03:12] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '3428005a72b7241c9648d73e6c486076@10.50.10.183' of Response 1682996281: Match Not Found <--- SIP read from 10.50.10.2:5060 ---> OPTIONS sip:metaswitch@10.50.10.171:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.10.2:5060;rport;branch=z9hG4bK-3abfe997eed869c7a05e505453a2b564-metaswitch-1 Allow-Events: message-summary Allow-Events: refer Allow-Events: dialog Allow-Events: line-seize Max-Forwards: 70 Call-ID: 7DA569F1@metaswitch From: ;tag=metaswitch+1+0+da4d346f CSeq: 83600129 OPTIONS Organization: Supported: 100rel Content-Length: 0 Contact: To: <-------------> [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: OPTIONS sip:metaswitch@10.50.10.171:5060;transport=udp SIP/2.0 (62) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.10.2:5060;rport;branch=z9hG4bK-3abfe997eed869c7a05e505453a2b564-metaswitch-1 (99) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: Allow-Events: message-summary (29) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: Allow-Events: refer (19) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: Allow-Events: dialog (20) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Allow-Events: line-seize (24) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Max-Forwards: 70 (16) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Call-ID: 7DA569F1@metaswitch (28) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: From: ;tag=metaswitch+1+0+da4d346f (66) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: CSeq: 83600129 OPTIONS (22) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: Organization: (14) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: Supported: 100rel (17) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 12: Content-Length: 0 (17) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 13: Contact: (41) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 14: To: (33) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 15: (0) --- (15 headers 0 lines) --- [May 17 10:03:12] DEBUG[7809]: chan_sip.c:4312 sip_alloc: Allocating new SIP dialog for 7DA569F1@metaswitch - OPTIONS (No RTP) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:14709 handle_request: **** Received OPTIONS (3) - Command in SIP OPTIONS Looking for metaswitch in default (domain 10.50.10.171) <--- Transmitting (no NAT) to 10.50.10.2:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.50.10.2:5060;rport;branch=z9hG4bK-3abfe997eed869c7a05e505453a2b564-metaswitch-1;received=10.50.10.2 From: ;tag=metaswitch+1+0+da4d346f To: ;tag=as7fd62623 Call-ID: 7DA569F1@metaswitch CSeq: 83600129 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '7DA569F1@metaswitch' in 32000 ms (Method: OPTIONS) [May 17 10:03:12] DEBUG[7809]: chan_sip.c:14927 sipsock_read: SIP message could not be handled, bad request: 7DA569F1@metaswitch [May 17 10:03:14] DEBUG[7809]: chan_sip.c:2011 __sip_autodestruct: Auto destroying SIP dialog 'A59906C1@metaswitch' [May 17 10:03:14] DEBUG[7809]: chan_sip.c:3110 sip_destroy: Destroying SIP dialog A59906C1@metaswitch Really destroying SIP dialog 'A59906C1@metaswitch' Method: OPTIONS <--- SIP read from 10.50.10.183:5060 ---> BYE sip:station2_line1@10.50.10.171 SIP/2.0 Max-Forwards: 70 Content-Length: 0 Via: SIP/2.0/UDP 10.50.10.183;branch=z9hG4bK2b0e016a9 Call-ID: 3428005a72b7241c9648d73e6c486076@10.50.10.183 From: Trunk01 ;tag=355161e0eb78137 To: station2_line1 ;tag=as7e66826b CSeq: 1682996282 BYE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Supported: replaces Proxy-Authorization:Digest response="009c6f76faa4cc90f06c043f5c2eb60e",username="station2",realm="asterisk",nonce="41c7ea9f",algorithm=MD5,uri="sip:station2_line1@10.50.10.171" User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 <-------------> [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: BYE sip:station2_line1@10.50.10.171 SIP/2.0 (43) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Max-Forwards: 70 (16) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: Content-Length: 0 (17) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: Via: SIP/2.0/UDP 10.50.10.183;branch=z9hG4bK2b0e016a9 (53) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: Call-ID: 3428005a72b7241c9648d73e6c486076@10.50.10.183 (54) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: From: Trunk01 ;tag=355161e0eb78137 (66) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: To: station2_line1 ;tag=as7e66826b (72) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: CSeq: 1682996282 BYE (20) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Supported: timer (16) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: Allow-Events: talk,hold,conference (34) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO (53) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: Supported: replaces (19) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 12: Proxy-Authorization:Digest response="009c6f76faa4cc90f06c043f5c2eb60e",username="station2",realm="asterisk",nonce="41c7ea9f",algorithm=MD5,uri="sip:station2_line1@10.50.10.171" (176) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 13: User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 (70) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 14: (0) --- (14 headers 0 lines) --- [May 17 10:03:14] DEBUG[7809]: chan_sip.c:14709 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 10.50.10.183 : 5060 (no NAT) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:1634 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 3428005a72b7241c9648d73e6c486076@10.50.10.183 [May 17 10:03:14] DEBUG[7809]: chan_sip.c:14263 handle_request_bye: Received bye, issuing owner hangup <--- Transmitting (no NAT) to 10.50.10.183:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.50.10.183;branch=z9hG4bK2b0e016a9;received=10.50.10.183 From: Trunk01 ;tag=355161e0eb78137 To: station2_line1 ;tag=as7e66826b Call-ID: 3428005a72b7241c9648d73e6c486076@10.50.10.183 CSeq: 1682996282 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [May 17 10:03:14] WARNING[7895]: app_meetme.c:2172 conf_run: Unable to write frame to channel SIP/station2-08d66e30 [May 17 10:03:14] DEBUG[7895]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SLA:station1_line1 [May 17 10:03:14] DEBUG[7895]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SLA:station2_line1 [May 17 10:03:14] DEBUG[7895]: pbx.c:2428 __ast_pbx_run: Extension station2_line1, priority 1 returned normally even though call was hung up [May 17 10:03:14] DEBUG[7895]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7895]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7895]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'station2_line1' [May 17 10:03:14] DEBUG[7895]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'sla_stations' [May 17 10:03:14] DEBUG[7895]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'SIP/station2-08d66e30' [May 17 10:03:14] DEBUG[7895]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7895]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'SLAStation' [May 17 10:03:14] DEBUG[7895]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'station2_line1' [May 17 10:03:14] DEBUG[7895]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:03:11' [May 17 10:03:14] DEBUG[7895]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:03:12' [May 17 10:03:14] DEBUG[7895]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:03:14' [May 17 10:03:14] DEBUG[7895]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '3' [May 17 10:03:14] DEBUG[7895]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2' [May 17 10:03:14] DEBUG[7895]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [May 17 10:03:14] DEBUG[7895]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [May 17 10:03:14] DEBUG[7895]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7895]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '1179421391.14' [May 17 10:03:14] DEBUG[7895]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7895]: channel.c:1726 ast_hangup: Hanging up channel 'SIP/station2-08d66e30' [May 17 10:03:14] DEBUG[7895]: chan_sip.c:3313 sip_hangup: Hangup call SIP/station2-08d66e30, SIP callid 3428005a72b7241c9648d73e6c486076@10.50.10.183) [May 17 10:03:14] DEBUG[7895]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/station2-08d66e30 [May 17 10:03:14] DEBUG[7897]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel meetme:SLA_line1 [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 's' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'default' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'Zap/pseudo-528115053' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:03:12' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:03:14' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '0' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'NO ANSWER' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '1179421392.17' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7897]: channel.c:1726 ast_hangup: Hanging up channel 'Zap/pseudo-528115053' [May 17 10:03:14] DEBUG[7897]: chan_zap.c:2412 zt_hangup: zt_hangup(Zap/pseudo-528115053) [May 17 10:03:14] DEBUG[7897]: chan_zap.c:2446 zt_hangup: Hangup: channel: -2 index = 0, normal = 32, callwait = -1, thirdcall = -1 [May 17 10:03:14] DEBUG[7897]: chan_zap.c:2874 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/pseudo-528115053 [May 17 10:03:14] DEBUG[7897]: chan_zap.c:1403 update_conf: Updated conferencing on -2, with 0 conference users -- Hungup 'Zap/pseudo-528115053' [May 17 10:03:14] DEBUG[7897]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel Zap/pseudo-528115053 [May 17 10:03:14] DEBUG[7897]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SLA:station1_line1 [May 17 10:03:14] DEBUG[7897]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SLA:station2_line1 [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'disa' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'line1_outbound' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'Local/disa@line1_outbound-933b,1' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:03:11' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:03:12' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:03:14' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '3' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '1179421391.15' [May 17 10:03:14] DEBUG[7897]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7897]: channel.c:1726 ast_hangup: Hanging up channel 'Local/disa@line1_outbound-933b,1' [May 17 10:03:14] DEBUG[7900]: channel.c:2911 set_format: Set channel Local/disa@line1_outbound-933b,2 to write format ulaw [May 17 10:03:14] DEBUG[7900]: channel.c:2030 ast_settimeout: Scheduling timer at 0 sample intervals [May 17 10:03:14] DEBUG[7900]: pbx.c:2407 __ast_pbx_run: Spawn extension (line1_outbound,disa,1) exited non-zero on 'Local/disa@line1_outbound-933b,2' == Spawn extension (line1_outbound, disa, 1) exited non-zero on 'Local/disa@line1_outbound-933b,2' [May 17 10:03:14] DEBUG[7900]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '"IC" <8055480003>' [May 17 10:03:14] DEBUG[7900]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '8055480003' [May 17 10:03:14] DEBUG[7900]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'disa' [May 17 10:03:14] DEBUG[7900]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'line1_outbound' [May 17 10:03:14] DEBUG[7900]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'Local/disa@line1_outbound-933b,2' [May 17 10:03:14] DEBUG[7900]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7900]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'DISA' [May 17 10:03:14] DEBUG[7900]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'no-password|line1_outbound' [May 17 10:03:14] DEBUG[7900]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:03:11' [May 17 10:03:14] DEBUG[7900]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:03:12' [May 17 10:03:14] DEBUG[7900]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2007-05-17 10:03:14' [May 17 10:03:14] DEBUG[7900]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '3' [May 17 10:03:14] DEBUG[7900]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '2' [May 17 10:03:14] DEBUG[7900]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [May 17 10:03:14] DEBUG[7900]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [May 17 10:03:14] DEBUG[7900]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7900]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '1179421391.16' [May 17 10:03:14] DEBUG[7900]: pbx.c:1662 pbx_substitute_variables_helper_full: Function result is '' [May 17 10:03:14] DEBUG[7900]: channel.c:1726 ast_hangup: Hanging up channel 'Local/disa@line1_outbound-933b,2' [May 17 10:03:14] DEBUG[7900]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel Local/disa@line1_outbound-933b,2 [May 17 10:03:14] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station1_line1 [May 17 10:03:14] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SLA:station1_line1 - state 1 (Not in use) [May 17 10:03:14] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station1_line1 [May 17 10:03:14] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station1_line1 Reliably Transmitting (no NAT) to 10.50.10.177:5060: NOTIFY sip:station1@10.50.10.177 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK5323c257;rport From: Line 1 ;tag=as4404e61b To: Trunk01 ;tag=a3a3bd7906dfaf5 Contact: Call-ID: 0871b9d69a4a057cf9138a1a7ebb36ad@10.50.10.177 CSeq: 106 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 230

terminated

--- [May 17 10:03:14] DEBUG[7787]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #142 [May 17 10:03:14] DEBUG[7897]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel Local/disa@line1_outbound-933b,1 Extension Changed station1_line1 new state Idle for Notify User station1 [May 17 10:03:14] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station2_line1 [May 17 10:03:14] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SLA:station2_line1 - state 1 (Not in use) [May 17 10:03:14] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station2_line1 [May 17 10:03:14] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station2_line1 Reliably Transmitting (no NAT) to 10.50.10.183:5060: NOTIFY sip:station2@10.50.10.183 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK596716c8;rport From: sip:station2_line1@10.50.10.171:5060;tag=as0a72300c To: Trunk01 ;tag=db0222544e54736 Contact: Call-ID: 0135d7249cd406b94e41bc097f840b5f@10.50.10.183 CSeq: 106 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 230

terminated

--- [May 17 10:03:14] DEBUG[7787]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #143 Extension Changed station2_line1 new state Idle for Notify User station2 [May 17 10:03:14] DEBUG[7787]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - station2 [May 17 10:03:14] DEBUG[7787]: chan_sip.c:15321 sip_devicestate: Checking device state for peer station2 [May 17 10:03:14] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SIP/station2 - state 1 (Not in use) [May 17 10:03:14] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "meetme" - number: SLA_line1 [May 17 10:03:14] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for meetme:SLA_line1 - state 4 (Invalid) [May 17 10:03:14] DEBUG[7787]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for Zap - pseudo [May 17 10:03:14] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for Zap/pseudo - state 0 (Unknown) [May 17 10:03:14] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station1_line1 [May 17 10:03:14] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SLA:station1_line1 - state 1 (Not in use) [May 17 10:03:14] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station1_line1 [May 17 10:03:14] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station2_line1 [May 17 10:03:14] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for SLA:station2_line1 - state 1 (Not in use) [May 17 10:03:14] DEBUG[7787]: devicestate.c:157 ast_device_state: Checking if I can find provider for "SLA" - number: station2_line1 [May 17 10:03:14] DEBUG[7787]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for Local - disa@line1_outbound [May 17 10:03:14] DEBUG[7787]: chan_local.c:145 local_devicestate: Checking if extension disa@line1_outbound exists (devicestate) [May 17 10:03:14] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for Local/disa@line1_outbound - state 1 (Not in use) [May 17 10:03:14] DEBUG[7787]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for Local - disa@line1_outbound [May 17 10:03:14] DEBUG[7787]: chan_local.c:145 local_devicestate: Checking if extension disa@line1_outbound exists (devicestate) [May 17 10:03:14] DEBUG[7787]: devicestate.c:287 do_state_change: Changing state for Local/disa@line1_outbound - state 1 (Not in use) [May 17 10:03:14] DEBUG[7908]: app_queue.c:546 changethread: Device 'SIP/station2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 17 10:03:14] DEBUG[7910]: app_queue.c:546 changethread: Device 'Zap/pseudo' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. [May 17 10:03:14] DEBUG[7913]: app_queue.c:546 changethread: Device 'Local/disa@line1_outbound' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [May 17 10:03:14] DEBUG[7914]: app_queue.c:546 changethread: Device 'Local/disa@line1_outbound' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 10.50.10.177:5060 ---> SIP/2.0 200 OK Call-ID: 0871b9d69a4a057cf9138a1a7ebb36ad@10.50.10.177 CSeq: 106 NOTIFY From: Line 1 ;tag=as4404e61b To: Trunk01 ;tag=a3a3bd7906dfaf5 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK5323c257;rport Content-Length: 0 Contact: Trunk01 Supported: replaces User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 <-------------> [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 200 OK (14) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 0871b9d69a4a057cf9138a1a7ebb36ad@10.50.10.177 (54) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 106 NOTIFY (16) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: Line 1 ;tag=as4404e61b (66) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: Trunk01 ;tag=a3a3bd7906dfaf5 (64) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK5323c257;rport (63) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Contact: Trunk01 (44) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Supported: replaces (19) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (71) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [May 17 10:03:14] DEBUG[7809]: chan_sip.c:2080 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #142 [May 17 10:03:14] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '0871b9d69a4a057cf9138a1a7ebb36ad@10.50.10.177' of Request 106: Match Not Found SIP Response message for INCOMING dialog NOTIFY arrived Really destroying SIP dialog '3428005a72b7241c9648d73e6c486076@10.50.10.183' Method: BYE <--- SIP read from 10.50.10.183:5060 ---> SIP/2.0 200 OK Call-ID: 0135d7249cd406b94e41bc097f840b5f@10.50.10.183 CSeq: 106 NOTIFY From: sip:station2_line1@10.50.10.171:5060;tag=as0a72300c To: Trunk01 ;tag=db0222544e54736 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK596716c8;rport Content-Length: 0 Contact: Trunk01 Supported: replaces User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 <-------------> [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 200 OK (14) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Call-ID: 0135d7249cd406b94e41bc097f840b5f@10.50.10.183 (54) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: CSeq: 106 NOTIFY (16) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: From: sip:station2_line1@10.50.10.171:5060;tag=as0a72300c (57) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: To: Trunk01 ;tag=db0222544e54736 (59) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK596716c8;rport (63) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Contact: Trunk01 (44) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Supported: replaces (19) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: User-Agent: Aastra 55i/2.0.1.2000 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5 (70) [May 17 10:03:14] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [May 17 10:03:14] DEBUG[7809]: chan_sip.c:2080 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #143 [May 17 10:03:14] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '0135d7249cd406b94e41bc097f840b5f@10.50.10.183' of Request 106: Match Not Found SIP Response message for INCOMING dialog NOTIFY arrived [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4312 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: OPTIONS sip:10.50.10.2 SIP/2.0 (30) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK6381ad86;rport (63) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: From: "asterisk" ;tag=as40f34b01 (59) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: To: (20) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: Contact: (36) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Call-ID: 5c16e80c147868964c31dd4f415f243d@10.50.10.171 (54) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: CSeq: 102 OPTIONS (17) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: User-Agent: Asterisk PBX (24) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: Max-Forwards: 70 (16) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: Date: Thu, 17 May 2007 17:03:19 GMT (35) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 11: Supported: replaces (19) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 12: Content-Length: 0 (17) Reliably Transmitting (no NAT) to 10.50.10.2:5060: OPTIONS sip:10.50.10.2 SIP/2.0 Via: SIP/2.0/UDP 10.50.10.171:5060;branch=z9hG4bK6381ad86;rport From: "asterisk" ;tag=as40f34b01 To: Contact: Call-ID: 5c16e80c147868964c31dd4f415f243d@10.50.10.171 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 17 May 2007 17:03:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [May 17 10:03:19] DEBUG[7809]: chan_sip.c:1976 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #144 <--- SIP read from 10.50.10.2:5060 ---> SIP/2.0 403 From: URI not recognized From: "asterisk" ;tag=as40f34b01 To: ;tag=metaswitch+1+0+e3b7d43b Via: SIP/2.0/UDP 10.50.10.171:5060;rport=5060;branch=z9hG4bK6381ad86 Server: DC-SIP/2.0 Organization: Content-Length: 0 Call-ID: 5c16e80c147868964c31dd4f415f243d@10.50.10.171 CSeq: 102 OPTIONS <-------------> [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 0: SIP/2.0 403 From: URI not recognized (36) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 1: From: "asterisk" ;tag=as40f34b01 (59) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 2: To: ;tag=metaswitch+1+0+e3b7d43b (48) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 3: Via: SIP/2.0/UDP 10.50.10.171:5060;rport=5060;branch=z9hG4bK6381ad86 (68) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 4: Server: DC-SIP/2.0 (18) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 5: Organization: (14) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 6: Content-Length: 0 (17) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 7: Call-ID: 5c16e80c147868964c31dd4f415f243d@10.50.10.171 (54) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 8: CSeq: 102 OPTIONS (17) [May 17 10:03:19] DEBUG[7809]: chan_sip.c:4577 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [May 17 10:03:19] DEBUG[7809]: chan_sip.c:2080 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #144 [May 17 10:03:19] DEBUG[7809]: chan_sip.c:2090 __sip_ack: Stopping retransmission on '5c16e80c147868964c31dd4f415f243d@10.50.10.171' of Request 102: Match Not Found Really destroying SIP dialog '5c16e80c147868964c31dd4f415f243d@10.50.10.171' Method: OPTIONS [May 17 10:03:21] DEBUG[7809]: chan_sip.c:2011 __sip_autodestruct: Auto destroying SIP dialog '6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171' [May 17 10:03:21] DEBUG[7809]: chan_sip.c:3110 sip_destroy: Destroying SIP dialog 6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171 Really destroying SIP dialog '6a62ed196a44e51952f41ed10d67a1e4@10.50.10.171' Method: INVITE [May 17 10:03:21] DEBUG[7809]: chan_sip.c:2011 __sip_autodestruct: Auto destroying SIP dialog '28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171' [May 17 10:03:21] DEBUG[7809]: chan_sip.c:3110 sip_destroy: Destroying SIP dialog 28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171 Really destroying SIP dialog '28ac46fe49d687ab3b3cc84c0ccbc81f@10.50.10.171' Method: INVITE