INVITE sip:test@2.2.2.2 SIP/2.0 Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK2f6bbaab;rport From: "me" ;tag=as085838a9 To: Contact: Call-ID: 04fb9ab41d7fb1c673422ea811439234@3.3.3.3 CSeq: 102 INVITE User-Agent: R Switch v1.1.12 Max-Forwards: 70 Date: Wed, 25 Jul 2007 18:03:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 162 v=0 o=root 6638 6638 IN IP4 3.3.3.3 s=session c=IN IP4 3.3.3.3 t=0 0 m=audio 19044 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - <-------------> [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Header 0: INVITE sip:test@2.2.2.2 SIP/2.0 (37) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Header 1: Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK2f6bbaab;rport (65) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Header 2: From: "me" ;tag=as085838a9 (50) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Header 3: To: (28) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Header 4: Contact: (33) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Header 5: Call-ID: 04fb9ab41d7fb1c673422ea811439234@3.3.3.3 (56) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Header 7: User-Agent: R Switch v1.1.12 (28) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Header 9: Date: Wed, 25 Jul 2007 18:03:00 GMT (35) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Header 11: Content-Type: application/sdp (29) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Header 12: Content-Length: 162 (19) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Header 13: (0) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Line: v=0 (3) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Line: o=root 6638 6638 IN IP4 3.3.3.3 (38) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Line: s=session (9) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Line: c=IN IP4 3.3.3.3 (23) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Line: t=0 0 (5) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Line: m=audio 19044 RTP/AVP 8 (23) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Jul 25 21:12:53] VERBOSE[26649] logger.c: --- (13 headers 8 lines) --- [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Setting NAT on RTP to On [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Allocating new SIP dialog for 04fb9ab41d7fb1c673422ea811439234@3.3.3.3 - INVITE (With RTP) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jul 25 21:12:53] VERBOSE[26649] logger.c: Sending to 3.3.3.3 : 5060 (NAT) [Jul 25 21:12:53] VERBOSE[26649] logger.c: Using INVITE request as basis request - 04fb9ab41d7fb1c673422ea811439234@3.3.3.3 [Jul 25 21:12:53] VERBOSE[26649] logger.c: Found no matching peer or user for '3.3.3.3:5060' [Jul 25 21:12:53] VERBOSE[26649] logger.c: Found RTP audio format 8 [Jul 25 21:12:53] VERBOSE[26649] logger.c: Peer audio RTP is at port 3.3.3.3:19044 [Jul 25 21:12:53] VERBOSE[26649] logger.c: Found description format PCMA for ID 8 [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: T38 state changed to 0 on channel [Jul 25 21:12:53] VERBOSE[26649] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [Jul 25 21:12:53] VERBOSE[26649] logger.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [Jul 25 21:12:53] VERBOSE[26649] logger.c: Peer audio RTP is at port 3.3.3.3:19044 [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Checking SIP call limits for device [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: Updating call counter for incoming call [Jul 25 21:12:53] VERBOSE[26649] logger.c: Looking for test in ivrmanager (domain 2.2.2.2) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: *** Our capabilities are 0x8 (alaw) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: This channel will not be able to handle video. [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: build_route: Contact hop: [Jul 25 21:12:53] VERBOSE[26649] logger.c: list_route: hop: [Jul 25 21:12:53] DEBUG[26649] chan_sip.c: SIP/3.3.3.3-006c69c0: New call is still down.... Trying... [Jul 25 21:12:53] VERBOSE[26649] logger.c: <--- Transmitting (NAT) to 3.3.3.3:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK2f6bbaab;received=3.3.3.3;rport=5060 From: "me" ;tag=as085838a9 To: Call-ID: 04fb9ab41d7fb1c673422ea811439234@3.3.3.3 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jul 25 21:12:53] DEBUG[26649] devicestate.c: Notification of state change to be queued on device/channel SIP/3.3.3.3-006c69c0 [Jul 25 21:12:53] DEBUG[26628] devicestate.c: No provider found, checking channel drivers for SIP - 3.3.3.3 [Jul 25 21:12:53] DEBUG[26628] chan_sip.c: Checking device state for peer 3.3.3.3 [Jul 25 21:12:53] DEBUG[26628] devicestate.c: Changing state for SIP/3.3.3.3 - state 2 (In use) [Jul 25 21:12:53] DEBUG[28006] pbx.c: Launching 'Answer' [Jul 25 21:12:53] VERBOSE[28006] logger.c: -- Executing [test@ivrmanager:1] Answer("SIP/3.3.3.3-006c69c0", "") in new stack [Jul 25 21:12:53] DEBUG[28006] devicestate.c: Notification of state change to be queued on device/channel SIP/3.3.3.3-006c69c0 [Jul 25 21:12:53] DEBUG[26628] devicestate.c: No provider found, checking channel drivers for SIP - 3.3.3.3 [Jul 25 21:12:53] DEBUG[26628] chan_sip.c: Checking device state for peer 3.3.3.3 [Jul 25 21:12:53] DEBUG[26628] channel.c: Avoiding initial deadlock for channel '0x6a3ef0' [Jul 25 21:12:53] DEBUG[28006] chan_sip.c: SIP answering channel: SIP/3.3.3.3-006c69c0 [Jul 25 21:12:53] DEBUG[28006] chan_sip.c: Setting framing from config on incoming call [Jul 25 21:12:53] DEBUG[28006] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True [Jul 25 21:12:53] DEBUG[28006] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jul 25 21:12:53] VERBOSE[28006] logger.c: Audio is at 2.2.2.2 port 12242 [Jul 25 21:12:53] VERBOSE[28006] logger.c: Adding codec 0x8 (alaw) to SDP [Jul 25 21:12:53] DEBUG[28006] chan_sip.c: -- Done with adding codecs to SDP [Jul 25 21:12:53] DEBUG[28006] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=24) [Jul 25 21:12:53] DEBUG[28006] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jul 25 21:12:53] VERBOSE[28006] logger.c: <--- Reliably Transmitting (NAT) to 3.3.3.3:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK2f6bbaab;received=3.3.3.3;rport=5060 From: "me" ;tag=as085838a9 To: ;tag=as4c513300 Call-ID: 04fb9ab41d7fb1c673422ea811439234@3.3.3.3 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 186 v=0 o=root 26624 26624 IN IP4 2.2.2.2 s=session c=IN IP4 2.2.2.2 t=0 0 m=audio 12242 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jul 25 21:12:53] DEBUG[28006] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #101 [Jul 25 21:12:53] DEBUG[28006] pbx.c: Launching 'Read' [Jul 25 21:12:53] VERBOSE[28006] logger.c: -- Executing [test@ivrmanager:2] Read("SIP/3.3.3.3-006c69c0", "asdf|demo-congrats|1") in new stack [Jul 25 21:12:53] VERBOSE[28006] logger.c: -- Accepting a maximum of 1 digits. [Jul 25 21:12:53] DEBUG[28006] channel.c: Set channel SIP/3.3.3.3-006c69c0 to write format gsm [Jul 25 21:12:53] DEBUG[28006] rtp.c: Ooh, format changed from unknown to alaw [Jul 25 21:12:53] DEBUG[28006] rtp.c: Created smoother: format: 8 ms: 20 len: 160 [Jul 25 21:12:53] DEBUG[28006] channel.c: Scheduling timer at 160 sample intervals [Jul 25 21:12:53] VERBOSE[28006] logger.c: -- Playing 'demo-congrats' (language 'en') [Jul 25 21:12:53] DEBUG[28007] app_queue.c: Device 'SIP/3.3.3.3' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jul 25 21:12:53] DEBUG[26628] devicestate.c: Changing state for SIP/3.3.3.3 - state 2 (In use) [Jul 25 21:12:53] DEBUG[28008] app_queue.c: Device 'SIP/3.3.3.3' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jul 25 21:12:54] VERBOSE[26649] logger.c: <--- SIP read from 3.3.3.3:5060 ---> ACK sip:test@2.2.2.2 SIP/2.0 Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK51d51de1;rport From: "me" ;tag=as085838a9 To: ;tag=as4c513300 Contact: Call-ID: 04fb9ab41d7fb1c673422ea811439234@3.3.3.3 CSeq: 102 ACK User-Agent: R Switch v1.1.12 Max-Forwards: 70 Content-Length: 0 <-------------> [Jul 25 21:12:54] DEBUG[26649] chan_sip.c: Header 0: ACK sip:test@2.2.2.2 SIP/2.0 (34) [Jul 25 21:12:54] DEBUG[26649] chan_sip.c: Header 1: Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK51d51de1;rport (65) [Jul 25 21:12:54] DEBUG[26649] chan_sip.c: Header 2: From: "me" ;tag=as085838a9 (50) [Jul 25 21:12:54] DEBUG[26649] chan_sip.c: Header 3: To: ;tag=as4c513300 (43) [Jul 25 21:12:54] DEBUG[26649] chan_sip.c: Header 4: Contact: (33) [Jul 25 21:12:54] DEBUG[26649] chan_sip.c: Header 5: Call-ID: 04fb9ab41d7fb1c673422ea811439234@3.3.3.3 (56) [Jul 25 21:12:54] DEBUG[26649] chan_sip.c: Header 6: CSeq: 102 ACK (13) [Jul 25 21:12:54] DEBUG[26649] chan_sip.c: Header 7: User-Agent: R Switch v1.1.12 (28) [Jul 25 21:12:54] DEBUG[26649] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jul 25 21:12:54] DEBUG[26649] chan_sip.c: Header 9: Content-Length: 0 (17) [Jul 25 21:12:54] DEBUG[26649] chan_sip.c: Header 10: (0) [Jul 25 21:12:54] VERBOSE[26649] logger.c: --- (10 headers 0 lines) --- [Jul 25 21:12:54] DEBUG[26649] chan_sip.c: = Found Their Call ID: 04fb9ab41d7fb1c673422ea811439234@3.3.3.3 Their Tag as085838a9 Our tag: as4c513300 [Jul 25 21:12:54] DEBUG[26649] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jul 25 21:12:54] DEBUG[26649] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #101 [Jul 25 21:12:54] DEBUG[26649] chan_sip.c: Stopping retransmission on '04fb9ab41d7fb1c673422ea811439234@3.3.3.3' of Response 102: Match Not Found [Jul 25 21:12:58] VERBOSE[26649] logger.c: <--- SIP read from 3.3.3.3:5060 ---> INFO sip:test@2.2.2.2 SIP/2.0 Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK2fceea81;rport From: "me" ;tag=as085838a9 To: ;tag=as4c513300 Contact: Call-ID: 04fb9ab41d7fb1c673422ea811439234@3.3.3.3 CSeq: 103 INFO User-Agent: R Switch v1.1.12 Max-Forwards: 70 Content-Type: application/dtmf-relay Content-Length: 24 Signal=5 Duration=250 <-------------> [Jul 25 21:12:58] DEBUG[26649] chan_sip.c: Header 0: INFO sip:test@2.2.2.2 SIP/2.0 (35) [Jul 25 21:12:58] DEBUG[26649] chan_sip.c: Header 1: Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK2fceea81;rport (65) [Jul 25 21:12:58] DEBUG[26649] chan_sip.c: Header 2: From: "me" ;tag=as085838a9 (50) [Jul 25 21:12:58] DEBUG[26649] chan_sip.c: Header 3: To: ;tag=as4c513300 (43) [Jul 25 21:12:58] DEBUG[26649] chan_sip.c: Header 4: Contact: (33) [Jul 25 21:12:58] DEBUG[26649] chan_sip.c: Header 5: Call-ID: 04fb9ab41d7fb1c673422ea811439234@3.3.3.3 (56) [Jul 25 21:12:58] DEBUG[26649] chan_sip.c: Header 6: CSeq: 103 INFO (14) [Jul 25 21:12:58] DEBUG[26649] chan_sip.c: Header 7: User-Agent: R Switch v1.1.12 (28) [Jul 25 21:12:58] DEBUG[26649] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jul 25 21:12:58] DEBUG[26649] chan_sip.c: Header 9: Content-Type: application/dtmf-relay (36) [Jul 25 21:12:58] DEBUG[26649] chan_sip.c: Header 10: Content-Length: 24 (18) [Jul 25 21:12:58] DEBUG[26649] chan_sip.c: Header 11: (0) [Jul 25 21:12:58] DEBUG[26649] chan_sip.c: Line: Signal=5 (8) [Jul 25 21:12:58] DEBUG[26649] chan_sip.c: Line: Duration=250 (12) [Jul 25 21:12:58] VERBOSE[26649] logger.c: --- (11 headers 2 lines) --- [Jul 25 21:12:58] DEBUG[26649] chan_sip.c: = Found Their Call ID: 04fb9ab41d7fb1c673422ea811439234@3.3.3.3 Their Tag as085838a9 Our tag: as4c513300 [Jul 25 21:12:58] DEBUG[26649] chan_sip.c: **** Received INFO (13) - Command in SIP INFO [Jul 25 21:12:58] VERBOSE[26649] logger.c: Receiving INFO! [Jul 25 21:12:58] VERBOSE[26649] logger.c: * DTMF-relay event received: 5 [Jul 25 21:12:58] VERBOSE[26649] logger.c: <--- Transmitting (NAT) to 3.3.3.3:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK2fceea81;received=3.3.3.3;rport=5060 From: "me" ;tag=as085838a9 To: ;tag=as4c513300 Call-ID: 04fb9ab41d7fb1c673422ea811439234@3.3.3.3 CSeq: 103 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jul 25 21:12:58] DTMF[28006] channel.c: DTMF end '5' received on SIP/3.3.3.3-006c69c0, duration 250 ms [Jul 25 21:12:58] DTMF[28006] channel.c: DTMF begin emulation of '5' with duration 250 queued on SIP/3.3.3.3-006c69c0 [Jul 25 21:12:58] DTMF[28006] channel.c: DTMF end emulation of '5' queued on SIP/3.3.3.3-006c69c0 [Jul 25 21:12:58] DEBUG[28006] channel.c: Scheduling timer at 0 sample intervals [Jul 25 21:12:58] DEBUG[28006] channel.c: Set channel SIP/3.3.3.3-006c69c0 to write format alaw [Jul 25 21:12:58] VERBOSE[28006] logger.c: -- User entered '5' [Jul 25 21:12:58] DEBUG[28006] pbx.c: Launching 'Verbose' [Jul 25 21:12:58] VERBOSE[28006] logger.c: -- Executing [test@ivrmanager:3] Verbose("SIP/3.3.3.3-006c69c0", "5") in new stack [Jul 25 21:12:58] VERBOSE[28006] logger.c: 5 [Jul 25 21:12:58] DEBUG[28006] pbx.c: Launching 'Hangup' [Jul 25 21:12:58] VERBOSE[28006] logger.c: -- Executing [test@ivrmanager:4] Hangup("SIP/3.3.3.3-006c69c0", "") in new stack [Jul 25 21:12:58] DEBUG[28006] pbx.c: Spawn extension (ivrmanager,test,4) exited non-zero on 'SIP/3.3.3.3-006c69c0' [Jul 25 21:12:58] VERBOSE[28006] logger.c: == Spawn extension (ivrmanager, test, 4) exited non-zero on 'SIP/3.3.3.3-006c69c0' [Jul 25 21:12:58] DEBUG[28006] channel.c: Soft-Hanging up channel 'SIP/3.3.3.3-006c69c0' [Jul 25 21:12:58] DEBUG[28006] channel.c: Hanging up channel 'SIP/3.3.3.3-006c69c0' [Jul 25 21:12:58] DEBUG[28006] chan_sip.c: Hangup call SIP/3.3.3.3-006c69c0, SIP callid 04fb9ab41d7fb1c673422ea811439234@3.3.3.3) [Jul 25 21:12:58] VERBOSE[28006] logger.c: Scheduling destruction of SIP dialog '04fb9ab41d7fb1c673422ea811439234@3.3.3.3' in 32000 ms (Method: INFO) [Jul 25 21:12:58] DEBUG[28006] chan_sip.c: Strict routing enforced for session 04fb9ab41d7fb1c673422ea811439234@3.3.3.3 [Jul 25 21:12:58] VERBOSE[28006] logger.c: set_destination: Parsing for address/port to send to [Jul 25 21:12:58] VERBOSE[28006] logger.c: set_destination: set destination to 3.3.3.3, port 5060 [Jul 25 21:12:58] VERBOSE[28006] logger.c: Reliably Transmitting (NAT) to 3.3.3.3:5060: BYE sip:111@3.3.3.3 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK7fb0e3a2;rport From: ;tag=as4c513300 To: "me" ;tag=as085838a9 Call-ID: 04fb9ab41d7fb1c673422ea811439234@3.3.3.3 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jul 25 21:12:58] DEBUG[28006] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #104 [Jul 25 21:12:58] DEBUG[28006] devicestate.c: Notification of state change to be queued on device/channel SIP/3.3.3.3-006c69c0 [Jul 25 21:12:58] DEBUG[28006] pbx.c: Function result is '"me" <111>' [Jul 25 21:12:58] DEBUG[26628] devicestate.c: No provider found, checking channel drivers for SIP - 3.3.3.3 [Jul 25 21:12:58] DEBUG[26628] chan_sip.c: Checking device state for peer 3.3.3.3 [Jul 25 21:12:58] DEBUG[26628] devicestate.c: Changing state for SIP/3.3.3.3 - state 1 (Not in use) [Jul 25 21:12:58] DEBUG[28006] pbx.c: Function result is '111' [Jul 25 21:12:58] DEBUG[28006] pbx.c: Function result is 'test' [Jul 25 21:12:58] DEBUG[28006] pbx.c: Function result is 'ivrmanager' [Jul 25 21:12:58] DEBUG[28006] pbx.c: Function result is 'SIP/3.3.3.3-006c69c0' [Jul 25 21:12:58] DEBUG[28006] pbx.c: Function result is '' [Jul 25 21:12:58] DEBUG[28006] pbx.c: Function result is 'Hangup' [Jul 25 21:12:58] DEBUG[28006] pbx.c: Function result is '' [Jul 25 21:12:58] DEBUG[28006] pbx.c: Function result is '2007-07-25 21:12:53' [Jul 25 21:12:58] DEBUG[28006] pbx.c: Function result is '2007-07-25 21:12:53' [Jul 25 21:12:58] DEBUG[28006] pbx.c: Function result is '2007-07-25 21:12:58' [Jul 25 21:12:58] DEBUG[28006] pbx.c: Function result is '5' [Jul 25 21:12:58] DEBUG[28006] pbx.c: Function result is '5' [Jul 25 21:12:58] DEBUG[28006] pbx.c: Function result is 'ANSWERED' [Jul 25 21:12:58] DEBUG[28006] pbx.c: Function result is 'DOCUMENTATION' [Jul 25 21:12:58] DEBUG[28006] pbx.c: Function result is '' [Jul 25 21:12:58] DEBUG[28006] pbx.c: Function result is '1185387173.35' [Jul 25 21:12:58] DEBUG[28006] pbx.c: Function result is '' [Jul 25 21:12:58] DEBUG[28009] app_queue.c: Device 'SIP/3.3.3.3' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jul 25 21:12:58] VERBOSE[26649] logger.c: <--- SIP read from 3.3.3.3:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK7fb0e3a2;received=2.2.2.2;rport=5060 From: ;tag=as4c513300 To: "me" ;tag=as085838a9 Call-ID: 04fb9ab41d7fb1c673422ea811439234@3.3.3.3 CSeq: 102 BYE User-Agent: R Switch v1.1.12 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing