Sep 6 16:39:22 VERBOSE[19156] logger.c: Asterisk Event Logger restarted Sep 6 16:39:22 VERBOSE[19156] logger.c: Asterisk Queue Logger restarted Sep 6 16:39:30 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.135:5060: Sep 6 16:39:30 DEBUG[19153] chan_sip.c: Header 0: (0) Sep 6 16:39:30 VERBOSE[19153] logger.c: --- (0 headers 0 lines)Sep 6 16:39:30 VERBOSE[19153] logger.c: --- (0 headers 0 lines) Nat keepalive Sep 6 16:39:30 VERBOSE[19153] logger.c: --- (0 headers 0 lines) Nat keepalive --- Sep 6 16:39:31 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.124:5060: INVITE sip:3097@pfdesenv.planetarium.com.br SIP/2.0 Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKXpp8DEf3UuFEPRWG Max-Forwards: 70 User-Agent: SOYO.SIP V1.49.002 CFG0 From: "3050" ;tag=OATvZpAyGXpN3cwZ To: "3097" Call-ID: wFJkP8nIZFcMprRv@172.16.10.124 Contact: CSeq: 1 INVITE Supported: replaces Content-Type: application/sdp Content-Length: 235 v=0 o=- 74791272 41272883 IN IP4 172.16.10.124 s=SIP CALL c=IN IP4 172.16.10.124 t=0 0 m=audio 1722 RTP/AVP 18 4 0 8 3 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 0: INVITE sip:3097@pfdesenv.planetarium.com.br SIP/2.0 (51) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKXpp8DEf3UuFEPRWG (66) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 2: Max-Forwards: 70 (16) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 3: User-Agent: SOYO.SIP V1.49.002 CFG0 (36) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 4: From: "3050" ;tag=OATvZpAyGXpN3cwZ (72) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 5: To: "3097" (49) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 6: Call-ID: wFJkP8nIZFcMprRv@172.16.10.124 (39) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 7: Contact: (38) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 8: CSeq: 1 INVITE (14) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 9: Supported: replaces (19) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 10: Content-Type: application/sdp (29) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 11: Content-Length: 235 (19) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 12: (0) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: v=0 (3) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: o=- 74791272 41272883 IN IP4 172.16.10.124 (42) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: s=SIP CALL (10) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: c=IN IP4 172.16.10.124 (22) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: t=0 0 (5) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: m=audio 1722 RTP/AVP 18 4 0 8 3 (31) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: a=rtpmap:4 G723/8000 (20) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) Sep 6 16:39:31 VERBOSE[19153] logger.c: --- (12 headers 11 lines)Sep 6 16:39:31 VERBOSE[19153] logger.c: --- (12 headers 11 lines)--- Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Allocating new SIP dialog for wFJkP8nIZFcMprRv@172.16.10.124 - INVITE (With RTP) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Begin: parsing SIP "Supported: replaces" Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Found SIP option: -replaces- Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Matched SIP option: replaces Sep 6 16:39:31 DEBUG[19153] chan_sip.c: * SIP extension value: 1 for call wFJkP8nIZFcMprRv@172.16.10.124 Sep 6 16:39:31 VERBOSE[19153] logger.c: Using INVITE request as basis request - wFJkP8nIZFcMprRv@172.16.10.124 Sep 6 16:39:31 VERBOSE[19153] logger.c: Sending to 172.16.10.124 : 5060 (non-NAT) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Setting NAT on RTP to 524288 Sep 6 16:39:31 VERBOSE[19153] logger.c: Reliably Transmitting (NAT) to 172.16.10.124:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKXpp8DEf3UuFEPRWG;received=172.16.10.124 From: "3050" ;tag=OATvZpAyGXpN3cwZ To: "3097" ;tag=as0b40fcc2 Call-ID: wFJkP8nIZFcMprRv@172.16.10.124 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="21fe649f" Content-Length: 0 --- Sep 6 16:39:31 DEBUG[19153] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #17 Sep 6 16:39:31 VERBOSE[19153] logger.c: Scheduling destruction of call 'wFJkP8nIZFcMprRv@172.16.10.124' in 15000 ms Sep 6 16:39:31 VERBOSE[19153] logger.c: Found user '3050' Sep 6 16:39:31 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.124:5060: ACK sip:3097@pfdesenv.planetarium.com.br SIP/2.0 Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKXpp8DEf3UuFEPRWG Max-Forwards: 70 User-Agent: SOYO.SIP V1.49.002 CFG0 From: "3050" ;tag=OATvZpAyGXpN3cwZ To: "3097" ;tag=as0b40fcc2 Call-ID: wFJkP8nIZFcMprRv@172.16.10.124 Contact: CSeq: 1 ACK Content-Length: 0 Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 0: ACK sip:3097@pfdesenv.planetarium.com.br SIP/2.0 (48) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKXpp8DEf3UuFEPRWG (66) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 2: Max-Forwards: 70 (16) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 3: User-Agent: SOYO.SIP V1.49.002 CFG0 (36) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 4: From: "3050" ;tag=OATvZpAyGXpN3cwZ (72) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 5: To: "3097" ;tag=as0b40fcc2 (64) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 6: Call-ID: wFJkP8nIZFcMprRv@172.16.10.124 (39) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 7: Contact: (38) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 8: CSeq: 1 ACK (11) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 9: Content-Length: 0 (17) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 10: (0) Sep 6 16:39:31 VERBOSE[19153] logger.c: --- (10 headers 0 lines)Sep 6 16:39:31 VERBOSE[19153] logger.c: --- (10 headers 0 lines)--- Sep 6 16:39:31 DEBUG[19153] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Sep 6 16:39:31 DEBUG[19153] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #17 Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Stopping retransmission on 'wFJkP8nIZFcMprRv@172.16.10.124' of Response 1: Match Found Sep 6 16:39:31 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.124:5060: INVITE sip:3097@pfdesenv.planetarium.com.br SIP/2.0 Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKx3AWi2sgX4esQPyi Max-Forwards: 70 User-Agent: SOYO.SIP V1.49.002 CFG0 From: "3050" ;tag=OATvZpAyGXpN3cwZ To: "3097" Call-ID: wFJkP8nIZFcMprRv@172.16.10.124 Contact: Proxy-Authorization: Digest username="3050", realm="asterisk", nonce="21fe649f", uri="sip:3097@pfdesenv.planetarium.com.br", response="6f084e60e1db0f6e8f4b71e7fd49edf9", algorithm=MD5 CSeq: 2 INVITE Supported: replaces Content-Type: application/sdp Content-Length: 235 v=0 o=- 61047389 93070911 IN IP4 172.16.10.124 s=SIP CALL c=IN IP4 172.16.10.124 t=0 0 m=audio 1722 RTP/AVP 18 4 0 8 3 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 0: INVITE sip:3097@pfdesenv.planetarium.com.br SIP/2.0 (51) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKx3AWi2sgX4esQPyi (66) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 2: Max-Forwards: 70 (16) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 3: User-Agent: SOYO.SIP V1.49.002 CFG0 (36) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 4: From: "3050" ;tag=OATvZpAyGXpN3cwZ (72) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 5: To: "3097" (49) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 6: Call-ID: wFJkP8nIZFcMprRv@172.16.10.124 (39) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 7: Contact: (38) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 8: Proxy-Authorization: Digest username="3050", realm="asterisk", nonce="21fe649f", uri="sip:3097@pfdesenv.planetarium.com.br", response="6f084e60e1db0f6e8f4b71e7fd49edf9", algorithm=MD5 (183) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 9: CSeq: 2 INVITE (14) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 10: Supported: replaces (19) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 11: Content-Type: application/sdp (29) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 12: Content-Length: 235 (19) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 13: (0) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: v=0 (3) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: o=- 61047389 93070911 IN IP4 172.16.10.124 (42) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: s=SIP CALL (10) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: c=IN IP4 172.16.10.124 (22) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: t=0 0 (5) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: m=audio 1722 RTP/AVP 18 4 0 8 3 (31) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: a=rtpmap:4 G723/8000 (20) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) Sep 6 16:39:31 VERBOSE[19153] logger.c: --- (13 headers 11 lines)Sep 6 16:39:31 VERBOSE[19153] logger.c: --- (13 headers 11 lines)--- Sep 6 16:39:31 DEBUG[19153] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Sep 6 16:39:31 VERBOSE[19153] logger.c: Using INVITE request as basis request - wFJkP8nIZFcMprRv@172.16.10.124 Sep 6 16:39:31 VERBOSE[19153] logger.c: Sending to 172.16.10.124 : 5060 (NAT) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Setting NAT on RTP to 524288 Sep 6 16:39:31 VERBOSE[19153] logger.c: Found user '3050' Sep 6 16:39:31 VERBOSE[19153] logger.c: Found RTP audio format 18 Sep 6 16:39:31 VERBOSE[19153] logger.c: Found RTP audio format 4 Sep 6 16:39:31 VERBOSE[19153] logger.c: Found RTP audio format 0 Sep 6 16:39:31 VERBOSE[19153] logger.c: Found RTP audio format 8 Sep 6 16:39:31 VERBOSE[19153] logger.c: Found RTP audio format 3 Sep 6 16:39:31 VERBOSE[19153] logger.c: Peer audio RTP is at port 172.16.10.124:1722 Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Peer audio RTP is at port 172.16.10.124:1722 Sep 6 16:39:31 VERBOSE[19153] logger.c: Found description format G729 Sep 6 16:39:31 VERBOSE[19153] logger.c: Found description format G723 Sep 6 16:39:31 VERBOSE[19153] logger.c: Found description format PCMU Sep 6 16:39:31 VERBOSE[19153] logger.c: Found description format PCMA Sep 6 16:39:31 VERBOSE[19153] logger.c: Found description format GSM Sep 6 16:39:31 VERBOSE[19153] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x8 (alaw) Sep 6 16:39:31 VERBOSE[19153] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Checking SIP call limits for device 3050 Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Updating call counter for incoming call Sep 6 16:39:31 VERBOSE[19153] logger.c: Looking for 3097 in sip_to_anywhere (domain pfdesenv.planetarium.com.br) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: build_route: Contact hop: Sep 6 16:39:31 VERBOSE[19153] logger.c: list_route: hop: Sep 6 16:39:31 VERBOSE[19153] logger.c: Transmitting (NAT) to 172.16.10.124:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKx3AWi2sgX4esQPyi;received=172.16.10.124 From: "3050" ;tag=OATvZpAyGXpN3cwZ To: "3097" Call-ID: wFJkP8nIZFcMprRv@172.16.10.124 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Sep 6 16:39:31 DEBUG[19137] chan_sip.c: Checking device state for peer 3050 Sep 6 16:39:31 DEBUG[19137] devicestate.c: Changing state for SIP/3050 - state 2 (In use) Sep 6 16:39:31 DEBUG[19163] pbx.c: Launching 'Queue' Sep 6 16:39:31 VERBOSE[19163] logger.c: -- Executing Queue("SIP/3050-097942f8", "myqueue|tr|||600") in new stack Sep 6 16:39:31 DEBUG[19163] app_queue.c: NO QUEUE_PRIO variable found. Using default. Sep 6 16:39:31 DEBUG[19163] app_queue.c: queue: myqueue, options: tr, url: , announce: , expires: 1157572171, priority: 0 Sep 6 16:39:31 DEBUG[19163] app_queue.c: Queue 'myqueue' Join, Channel 'SIP/3050-097942f8', Position '1' Sep 6 16:39:31 VERBOSE[19163] logger.c: Transmitting (NAT) to 172.16.10.124:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKx3AWi2sgX4esQPyi;received=172.16.10.124 From: "3050" ;tag=OATvZpAyGXpN3cwZ To: "3097" ;tag=as3d2e9a87 Call-ID: wFJkP8nIZFcMprRv@172.16.10.124 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Sep 6 16:39:31 DEBUG[19164] app_queue.c: Device 'SIP/3050' changed to state '2' (In use) Sep 6 16:39:31 DEBUG[19163] app_queue.c: It's our turn (SIP/3050-097942f8). Sep 6 16:39:31 DEBUG[19163] app_queue.c: SIP/3050-097942f8 is trying to call a queue member. Sep 6 16:39:31 DEBUG[19163] app_queue.c: Queue with URL=_ Sep 6 16:39:31 DEBUG[19163] app_queue.c: Trying 'Agent/3052' with metric 0 Sep 6 16:39:31 DEBUG[19163] channel.c: Not copying variable STACK-sip_to_anywhere-3097-1. Sep 6 16:39:31 DEBUG[19163] channel.c: Not copying variable SIPCALLID. Sep 6 16:39:31 DEBUG[19163] channel.c: Not copying variable SIPUSERAGENT. Sep 6 16:39:31 DEBUG[19163] channel.c: Not copying variable SIPDOMAIN. Sep 6 16:39:31 DEBUG[19163] channel.c: Not copying variable SIPURI. Sep 6 16:39:31 VERBOSE[19163] logger.c: -- outgoing agentcall, to agent '3052', on 'Local/3052@queue_to_agent-1e9b,1' Sep 6 16:39:31 VERBOSE[19163] logger.c: -- Called Agent/3052 Sep 6 16:39:31 DEBUG[19137] devicestate.c: Changing state for Local/3052@queue_to_agent - state 2 (In use) Sep 6 16:39:31 DEBUG[19165] pbx.c: Function result is '0' Sep 6 16:39:31 DEBUG[19165] pbx.c: Launching 'NoOp' Sep 6 16:39:31 VERBOSE[19165] logger.c: -- Executing NoOp("Local/3052@queue_to_agent-1e9b,2", "Group: 3052 Count: 0") in new stack Sep 6 16:39:31 DEBUG[19165] pbx.c: Function result is '0' Sep 6 16:39:31 DEBUG[19165] pbx.c: Expression result is '0' Sep 6 16:39:31 DEBUG[19165] pbx.c: Launching 'GotoIf' Sep 6 16:39:31 VERBOSE[19165] logger.c: -- Executing GotoIf("Local/3052@queue_to_agent-1e9b,2", "0 ? 1000") in new stack Sep 6 16:39:31 DEBUG[19165] pbx.c: Not taking any branch Sep 6 16:39:31 DEBUG[19165] pbx.c: Launching 'Dial' Sep 6 16:39:31 VERBOSE[19165] logger.c: -- Executing Dial("Local/3052@queue_to_agent-1e9b,2", "SIP/3052|15|T") in new stack Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Setting NAT on RTP to 0 Sep 6 16:39:31 DEBUG[19165] channel.c: Not copying variable STACK-queue_to_agent-3052-3. Sep 6 16:39:31 DEBUG[19165] channel.c: Not copying variable STACK-queue_to_agent-3052-2. Sep 6 16:39:31 DEBUG[19165] channel.c: Not copying variable STACK-queue_to_agent-3052-1. Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Outgoing Call for 3052 Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Updating call counter for outgoing call Sep 6 16:39:31 DEBUG[19166] app_queue.c: Device 'Local/3052@queue_to_agent' changed to state '2' (In use) but we don't care because they're not a member of any queue. Sep 6 16:39:31 VERBOSE[19165] logger.c: We're at 200.196.44.45 port 10012 Sep 6 16:39:31 VERBOSE[19165] logger.c: Adding codec 0x8 (alaw) to SDP Sep 6 16:39:31 VERBOSE[19165] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Header 0: INVITE sip:3052@172.16.10.141 SIP/2.0 (37) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK4b8adc63;rport (64) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Header 2: From: "Ramal Teste" ;tag=as2e2bac0a (59) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Header 3: To: (28) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Header 4: Contact: (33) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Header 5: Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 (55) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Header 6: CSeq: 102 INVITE (16) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Header 8: Max-Forwards: 70 (16) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Header 9: Date: Wed, 06 Sep 2006 19:39:31 GMT (35) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Header 11: Content-Type: application/sdp (29) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Header 12: Content-Length: 218 (19) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Header 13: (0) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Line: v=0 (3) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Line: o=root 19118 19118 IN IP4 200.196.44.45 (39) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Line: s=session (9) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Line: c=IN IP4 200.196.44.45 (22) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Line: t=0 0 (5) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Line: m=audio 10012 RTP/AVP 8 101 (27) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Line: a=fmtp:101 0-16 (15) Sep 6 16:39:31 DEBUG[19165] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Sep 6 16:39:31 VERBOSE[19165] logger.c: 13 headers, 10 lines Sep 6 16:39:31 VERBOSE[19165] logger.c: Reliably Transmitting (no NAT) to 172.16.10.141:5060: INVITE sip:3052@172.16.10.141 SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK4b8adc63;rport From: "Ramal Teste" ;tag=as2e2bac0a To: Contact: Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 06 Sep 2006 19:39:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 218 v=0 o=root 19118 19118 IN IP4 200.196.44.45 s=session c=IN IP4 200.196.44.45 t=0 0 m=audio 10012 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Sep 6 16:39:31 DEBUG[19165] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #19 Sep 6 16:39:31 VERBOSE[19165] logger.c: -- Called 3052 Sep 6 16:39:31 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.141:2217: SIP/2.0 100 Trying Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 CSeq: 102 INVITE From: "Ramal Teste" ;tag=as2e2bac0a To: ;tag=e949f769a006cca Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK4b8adc63;rport Content-Length: 0 User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 1: Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 (55) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 2: CSeq: 102 INVITE (16) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 3: From: "Ramal Teste" ;tag=as2e2bac0a (59) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 4: To: ;tag=e949f769a006cca (48) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 5: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK4b8adc63;rport (64) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 6: Content-Length: 0 (17) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 7: User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (58) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 8: (0) Sep 6 16:39:31 VERBOSE[19153] logger.c: --- (8 headers 0 lines)Sep 6 16:39:31 VERBOSE[19153] logger.c: --- (8 headers 0 lines)--- Sep 6 16:39:31 DEBUG[19153] chan_sip.c: *** SIP TIMER: Cancelling retransmission #19 - INVITE (got response) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7c610120743a79ec554afc953f5df4a1@200.196.44.45' Request 102: Found Sep 6 16:39:31 DEBUG[19153] chan_sip.c: SIP response 100 to standard invite Sep 6 16:39:31 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.141:2217: SIP/2.0 180 Ringing Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 CSeq: 102 INVITE From: "Ramal Teste" ;tag=as2e2bac0a To: ;tag=e949f769a006cca Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK4b8adc63;rport Content-Length: 0 Contact: User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 1: Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 (55) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 2: CSeq: 102 INVITE (16) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 3: From: "Ramal Teste" ;tag=as2e2bac0a (59) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 4: To: ;tag=e949f769a006cca (48) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 5: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK4b8adc63;rport (64) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 6: Content-Length: 0 (17) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 7: Contact: (33) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 8: User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (58) Sep 6 16:39:31 DEBUG[19153] chan_sip.c: Header 9: (0) Sep 6 16:39:31 VERBOSE[19153] logger.c: --- (9 headers 0 lines)Sep 6 16:39:31 VERBOSE[19153] logger.c: --- (9 headers 0 lines)--- Sep 6 16:39:31 DEBUG[19153] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7c610120743a79ec554afc953f5df4a1@200.196.44.45' Request 102: Found Sep 6 16:39:31 DEBUG[19153] chan_sip.c: SIP response 180 to standard invite Sep 6 16:39:31 VERBOSE[19165] logger.c: -- SIP/3052-0979ae40 is ringing Sep 6 16:39:31 DEBUG[19137] chan_sip.c: Checking device state for peer 3052 Sep 6 16:39:31 DEBUG[19137] devicestate.c: Changing state for SIP/3052 - state 6 (Ringing) Sep 6 16:39:31 DEBUG[19137] chan_sip.c: Checking device state for peer 3052 Sep 6 16:39:31 VERBOSE[19163] logger.c: -- Agent/3052 is ringing Sep 6 16:39:31 DEBUG[19167] app_queue.c: Device 'SIP/3052' changed to state '6' (Ringing) Sep 6 16:39:36 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.141:2217: SIP/2.0 200 OK Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 CSeq: 102 INVITE From: "Ramal Teste" ;tag=as2e2bac0a To: ;tag=e949f769a006cca Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK4b8adc63;rport Content-Length: 233 Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Content-Type: application/sdp Supported: replaces Contact: User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 1051082646 IN IP4 172.16.10.141 s=SIP Call c=IN IP4 172.16.10.141 t=0 0 m=audio 10004 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=silenceSupp:off - - - - Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 1: Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 (55) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 2: CSeq: 102 INVITE (16) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 3: From: "Ramal Teste" ;tag=as2e2bac0a (59) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 4: To: ;tag=e949f769a006cca (48) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 5: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK4b8adc63;rport (64) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 6: Content-Length: 233 (19) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 7: Allow: NOTIFY (13) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 8: Allow: REFER (12) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 9: Allow: OPTIONS (14) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 10: Allow: INVITE (13) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 11: Allow: ACK (10) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 12: Allow: CANCEL (13) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 13: Allow: BYE (10) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 14: Content-Type: application/sdp (29) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 15: Supported: replaces (19) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 16: Contact: (33) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 17: User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (58) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 18: (0) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Line: v=0 (3) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Line: o=MxSIP 0 1051082646 IN IP4 172.16.10.141 (41) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Line: s=SIP Call (10) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Line: c=IN IP4 172.16.10.141 (22) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Line: t=0 0 (5) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Line: m=audio 10004 RTP/AVP 8 101 (27) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Line: a=fmtp:101 0-15 (15) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Line: a=sendrecv (10) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Sep 6 16:39:36 VERBOSE[19153] logger.c: --- (18 headers 11 lines)Sep 6 16:39:36 VERBOSE[19153] logger.c: --- (18 headers 11 lines)--- Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Acked pending invite 102 Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Stopping retransmission on '7c610120743a79ec554afc953f5df4a1@200.196.44.45' of Request 102: Match Found Sep 6 16:39:36 DEBUG[19153] chan_sip.c: SIP response 200 to standard invite Sep 6 16:39:36 VERBOSE[19153] logger.c: Found RTP audio format 8 Sep 6 16:39:36 VERBOSE[19153] logger.c: Found RTP audio format 101 Sep 6 16:39:36 VERBOSE[19153] logger.c: Peer audio RTP is at port 172.16.10.141:10004 Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Peer audio RTP is at port 172.16.10.141:10004 Sep 6 16:39:36 VERBOSE[19153] logger.c: Found description format PCMA Sep 6 16:39:36 VERBOSE[19153] logger.c: Found description format telephone-event Sep 6 16:39:36 VERBOSE[19153] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Sep 6 16:39:36 VERBOSE[19153] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: build_route: Contact hop: Sep 6 16:39:36 VERBOSE[19153] logger.c: list_route: hop: Sep 6 16:39:36 VERBOSE[19153] logger.c: set_destination: Parsing for address/port to send to Sep 6 16:39:36 VERBOSE[19153] logger.c: set_destination: set destination to 172.16.10.141, port 5060 Sep 6 16:39:36 VERBOSE[19153] logger.c: Transmitting (no NAT) to 172.16.10.141:5060: ACK sip:3052@172.16.10.141 SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK05ddcd39;rport From: "Ramal Teste" ;tag=as2e2bac0a To: ;tag=e949f769a006cca Contact: Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Sep 6 16:39:36 VERBOSE[19165] logger.c: -- SIP/3052-0979ae40 answered Local/3052@queue_to_agent-1e9b,2 Sep 6 16:39:36 DEBUG[19137] chan_sip.c: Checking device state for peer 3052 Sep 6 16:39:36 DEBUG[19137] devicestate.c: Changing state for SIP/3052 - state 2 (In use) Sep 6 16:39:36 DEBUG[19137] chan_sip.c: Checking device state for peer 3052 Sep 6 16:39:36 DEBUG[19137] devicestate.c: Changing state for Local/3052@queue_to_agent - state 2 (In use) Sep 6 16:39:36 DEBUG[19163] app_queue.c: Dunno what to do with control type -1 Sep 6 16:39:36 VERBOSE[19163] logger.c: -- Agent/3052 answered SIP/3050-097942f8 Sep 6 16:39:36 DEBUG[19163] chan_sip.c: sip_answer(SIP/3050-097942f8) Sep 6 16:39:36 VERBOSE[19163] logger.c: We're at 200.196.44.45 port 16034 Sep 6 16:39:36 VERBOSE[19163] logger.c: Adding codec 0x8 (alaw) to SDP Sep 6 16:39:36 VERBOSE[19163] logger.c: Reliably Transmitting (NAT) to 172.16.10.124:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKx3AWi2sgX4esQPyi;received=172.16.10.124 From: "3050" ;tag=OATvZpAyGXpN3cwZ To: "3097" ;tag=as3d2e9a87 Call-ID: wFJkP8nIZFcMprRv@172.16.10.124 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 162 v=0 o=root 19118 19118 IN IP4 200.196.44.45 s=session c=IN IP4 200.196.44.45 t=0 0 m=audio 16034 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- Sep 6 16:39:36 DEBUG[19163] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #20 Sep 6 16:39:36 DEBUG[19169] app_queue.c: Device 'SIP/3052' changed to state '2' (In use) Sep 6 16:39:36 DEBUG[19170] app_queue.c: Device 'Local/3052@queue_to_agent' changed to state '2' (In use) but we don't care because they're not a member of any queue. Sep 6 16:39:36 DEBUG[19137] devicestate.c: Changing state for Local/3052@queue_to_agent - state 2 (In use) Sep 6 16:39:36 DEBUG[19137] devicestate.c: Changing state for Agent/3052 - state 3 (Busy) Sep 6 16:39:36 DEBUG[19137] chan_sip.c: Checking device state for peer 3050 Sep 6 16:39:36 DEBUG[19137] devicestate.c: Changing state for SIP/3050 - state 2 (In use) Sep 6 16:39:36 DEBUG[19171] app_queue.c: Device 'Local/3052@queue_to_agent' changed to state '2' (In use) but we don't care because they're not a member of any queue. Sep 6 16:39:36 DEBUG[19172] app_queue.c: Device 'Agent/3052' changed to state '3' (Busy) Sep 6 16:39:36 DEBUG[19173] app_queue.c: Device 'SIP/3050' changed to state '2' (In use) Sep 6 16:39:36 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.124:5060: ACK sip:3097@200.196.44.45 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKNrJyifD3RhX823Fy Max-Forwards: 70 User-Agent: SOYO.SIP V1.49.002 CFG0 From: "3050" ;tag=OATvZpAyGXpN3cwZ To: "3097" ;tag=as3d2e9a87 Call-ID: wFJkP8nIZFcMprRv@172.16.10.124 Contact: Proxy-Authorization: Digest username="3050", realm="asterisk", nonce="21fe649f", uri="sip:3097@200.196.44.45", response="96f9fcaf06e33e7bb461fca0488d51ac", algorithm=MD5 CSeq: 2 ACK Content-Length: 0 Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 0: ACK sip:3097@200.196.44.45 SIP/2.0 (34) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKNrJyifD3RhX823Fy (66) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 2: Max-Forwards: 70 (16) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 3: User-Agent: SOYO.SIP V1.49.002 CFG0 (36) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 4: From: "3050" ;tag=OATvZpAyGXpN3cwZ (72) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 5: To: "3097" ;tag=as3d2e9a87 (64) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 6: Call-ID: wFJkP8nIZFcMprRv@172.16.10.124 (39) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 7: Contact: (38) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 8: Proxy-Authorization: Digest username="3050", realm="asterisk", nonce="21fe649f", uri="sip:3097@200.196.44.45", response="96f9fcaf06e33e7bb461fca0488d51ac", algorithm=MD5 (169) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 9: CSeq: 2 ACK (11) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 10: Content-Length: 0 (17) Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Header 11: (0) Sep 6 16:39:36 VERBOSE[19153] logger.c: --- (11 headers 0 lines)Sep 6 16:39:36 VERBOSE[19153] logger.c: --- (11 headers 0 lines)--- Sep 6 16:39:36 DEBUG[19153] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Sep 6 16:39:36 DEBUG[19153] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #20 Sep 6 16:39:36 DEBUG[19153] chan_sip.c: Stopping retransmission on 'wFJkP8nIZFcMprRv@172.16.10.124' of Response 2: Match Found Sep 6 16:39:36 DEBUG[19165] channel.c: Planning to masquerade channel SIP/3052-0979ae40 into the structure of Local/3052@queue_to_agent-1e9b,1 Sep 6 16:39:36 DEBUG[19165] channel.c: Done planning to masquerade channel SIP/3052-0979ae40 into the structure of Local/3052@queue_to_agent-1e9b,1 Sep 6 16:39:36 DEBUG[19165] chan_local.c: Not posting to queue since already masked on 'Local/3052@queue_to_agent-1e9b,2' Sep 6 16:39:36 DEBUG[19163] channel.c: Actually Masquerading SIP/3052-0979ae40(6) into the structure of Local/3052@queue_to_agent-1e9b,1(6) Sep 6 16:39:36 DEBUG[19163] channel.c: Got clone lock for masquerade on 'SIP/3052-0979ae40' at 0x97a04ec Sep 6 16:39:36 DEBUG[19163] channel.c: Putting channel SIP/3052-0979ae40 in 8/8 formats Sep 6 16:39:36 DEBUG[19163] channel.c: Released clone lock on 'Local/3052@queue_to_agent-1e9b,1' Sep 6 16:39:36 DEBUG[19165] channel.c: Didn't get a frame from channel: Local/3052@queue_to_agent-1e9b,2 Sep 6 16:39:36 DEBUG[19165] channel.c: Bridge stops bridging channels Local/3052@queue_to_agent-1e9b,2 and Local/3052@queue_to_agent-1e9b,1 Sep 6 16:39:36 DEBUG[19163] channel.c: Done Masquerading SIP/3052-0979ae40 (6) Sep 6 16:39:36 DEBUG[19163] chan_agent.c: Bridge on 'SIP/3052-0979ae40' being set to 'Agent/3052' (3) Sep 6 16:39:36 DEBUG[19165] channel.c: Hanging up zombie 'Local/3052@queue_to_agent-1e9b,1' Sep 6 16:39:36 DEBUG[19165] app_dial.c: Exiting with DIALSTATUS=ANSWER. Sep 6 16:39:36 DEBUG[19165] pbx.c: Spawn extension (queue_to_agent,3052,3) exited non-zero on 'Local/3052@queue_to_agent-1e9b,2' Sep 6 16:39:36 DEBUG[19137] channel.c: Avoiding initial deadlock for 'Local/3052@queue_to_agent-1e9b,2' Sep 6 16:39:36 DEBUG[19137] channel.c: Avoiding initial deadlock for 'Local/3052@queue_to_agent-1e9b,2' Sep 6 16:39:36 DEBUG[19137] channel.c: Avoiding initial deadlock for 'Local/3052@queue_to_agent-1e9b,2' Sep 6 16:39:36 DEBUG[19137] channel.c: Avoiding initial deadlock for 'Local/3052@queue_to_agent-1e9b,2' Sep 6 16:39:36 DEBUG[19163] rtp.c: Ooh, format changed from unknown to alaw Sep 6 16:39:36 DEBUG[19137] channel.c: Avoiding initial deadlock for 'Local/3052@queue_to_agent-1e9b,2' Sep 6 16:39:36 DEBUG[19137] channel.c: Avoiding initial deadlock for 'Local/3052@queue_to_agent-1e9b,2' Sep 6 16:39:36 DEBUG[19137] channel.c: Avoiding initial deadlock for 'Local/3052@queue_to_agent-1e9b,2' Sep 6 16:39:36 DEBUG[19163] rtp.c: Ooh, format changed from unknown to alaw Sep 6 16:39:36 DEBUG[19137] channel.c: Avoiding initial deadlock for 'Local/3052@queue_to_agent-1e9b,2' Sep 6 16:39:36 DEBUG[19137] channel.c: Avoiding initial deadlock for 'Local/3052@queue_to_agent-1e9b,2' Sep 6 16:39:36 DEBUG[19137] channel.c: Avoiding initial deadlock for 'Local/3052@queue_to_agent-1e9b,2' Sep 6 16:39:36 WARNING[19137] channel.c: Avoided initial deadlock for '0x9799d88', 10 retries! Sep 6 16:39:36 DEBUG[19137] devicestate.c: Changing state for Local/3052@queue_to_agent - state 0 (Unknown) Sep 6 16:39:36 DEBUG[19174] app_queue.c: Device 'Local/3052@queue_to_agent' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. Sep 6 16:39:36 DEBUG[19165] channel.c: Hanging up channel 'Local/3052@queue_to_agent-1e9b,2' Sep 6 16:39:36 DEBUG[19137] devicestate.c: Changing state for Local/3052@queue_to_agent - state 0 (Unknown) Sep 6 16:39:36 DEBUG[19176] app_queue.c: Device 'Local/3052@queue_to_agent' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. Sep 6 16:39:37 DEBUG[19163] rtp.c: Got RTCP report of 84 bytes Sep 6 16:39:38 DEBUG[19163] rtp.c: Got RTCP report of 84 bytes Sep 6 16:39:38 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.141:2217: INVITE sip:3050@200.196.44.45 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bK28a82477f Max-Forwards: 70 Content-Length: 459 To: "Ramal Teste" ;tag=as2e2bac0a From: ;tag=e949f769a006cca Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 CSeq: 1686507440 INVITE Supported: timer Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Content-Type: application/sdp Contact: Supported: replaces User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 1051082647 IN IP4 172.16.10.141 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 10004 RTP/AVP 0 18 101 102 107 104 105 106 4 8 127 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 BV16/8000 a=rtpmap:102 BV32/16000 a=rtpmap:107 L16/16000 a=rtpmap:104 PCMU/16000 a=rtpmap:105 PCMA/16000 a=rtpmap:106 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 0: INVITE sip:3050@200.196.44.45 SIP/2.0 (37) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bK28a82477f (54) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 2: Max-Forwards: 70 (16) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 3: Content-Length: 459 (19) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 4: To: "Ramal Teste" ;tag=as2e2bac0a (57) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 5: From: ;tag=e949f769a006cca (50) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 6: Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 (55) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 7: CSeq: 1686507440 INVITE (23) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 8: Supported: timer (16) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 9: Allow: NOTIFY (13) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 10: Allow: REFER (12) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 11: Allow: OPTIONS (14) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 12: Allow: INVITE (13) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 13: Allow: ACK (10) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 14: Allow: CANCEL (13) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 15: Allow: BYE (10) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 16: Content-Type: application/sdp (29) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 17: Contact: (33) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 18: Supported: replaces (19) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 19: User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (58) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 20: (0) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Line: v=0 (3) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Line: o=MxSIP 0 1051082647 IN IP4 172.16.10.141 (41) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Line: s=SIP Call (10) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Line: c=IN IP4 0.0.0.0 (16) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Line: t=0 0 (5) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Line: m=audio 10004 RTP/AVP 0 18 101 102 107 104 105 106 4 8 127 (58) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Line: a=rtpmap:101 BV16/8000 (22) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Line: a=rtpmap:102 BV32/16000 (23) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Line: a=rtpmap:107 L16/16000 (22) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Line: a=rtpmap:104 PCMU/16000 (23) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Line: a=rtpmap:105 PCMA/16000 (23) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Line: a=rtpmap:106 L16/8000 (21) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Line: a=rtpmap:4 G723/8000 (20) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Line: a=rtpmap:127 telephone-event/8000 (33) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Line: a=fmtp:101 0-15 (15) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Sep 6 16:39:38 VERBOSE[19153] logger.c: --- (20 headers 19 lines)Sep 6 16:39:38 VERBOSE[19153] logger.c: --- (20 headers 19 lines)--- Sep 6 16:39:38 DEBUG[19153] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Begin: parsing SIP "Supported: timer" Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Found SIP option: -timer- Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Matched SIP option: timer Sep 6 16:39:38 DEBUG[19153] chan_sip.c: * SIP extension value: 4 for call 7c610120743a79ec554afc953f5df4a1@200.196.44.45 Sep 6 16:39:38 VERBOSE[19153] logger.c: Using INVITE request as basis request - 7c610120743a79ec554afc953f5df4a1@200.196.44.45 Sep 6 16:39:38 VERBOSE[19153] logger.c: Sending to 172.16.10.141 : 5060 (non-NAT) Sep 6 16:39:38 VERBOSE[19153] logger.c: Found RTP audio format 0 Sep 6 16:39:38 VERBOSE[19153] logger.c: Found RTP audio format 18 Sep 6 16:39:38 VERBOSE[19153] logger.c: Found RTP audio format 101 Sep 6 16:39:38 VERBOSE[19153] logger.c: Found RTP audio format 102 Sep 6 16:39:38 VERBOSE[19153] logger.c: Found RTP audio format 107 Sep 6 16:39:38 VERBOSE[19153] logger.c: Found RTP audio format 104 Sep 6 16:39:38 VERBOSE[19153] logger.c: Found RTP audio format 105 Sep 6 16:39:38 VERBOSE[19153] logger.c: Found RTP audio format 106 Sep 6 16:39:38 VERBOSE[19153] logger.c: Found RTP audio format 4 Sep 6 16:39:38 VERBOSE[19153] logger.c: Found RTP audio format 8 Sep 6 16:39:38 VERBOSE[19153] logger.c: Found RTP audio format 127 Sep 6 16:39:38 VERBOSE[19153] logger.c: Peer audio RTP is at port 0.0.0.0:10004 Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Peer audio RTP is at port 0.0.0.0:10004 Sep 6 16:39:38 VERBOSE[19153] logger.c: Found description format PCMU Sep 6 16:39:38 VERBOSE[19153] logger.c: Found description format G729 Sep 6 16:39:38 VERBOSE[19153] logger.c: Found description format BV16 Sep 6 16:39:38 VERBOSE[19153] logger.c: Found description format BV32 Sep 6 16:39:38 VERBOSE[19153] logger.c: Found description format L16 Sep 6 16:39:38 VERBOSE[19153] logger.c: Found description format PCMU Sep 6 16:39:38 VERBOSE[19153] logger.c: Found description format PCMA Sep 6 16:39:38 VERBOSE[19153] logger.c: Found description format L16 Sep 6 16:39:38 VERBOSE[19153] logger.c: Found description format G723 Sep 6 16:39:38 VERBOSE[19153] logger.c: Found description format PCMA Sep 6 16:39:38 VERBOSE[19153] logger.c: Found description format telephone-event Sep 6 16:39:38 VERBOSE[19153] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x14d (g723|ulaw|alaw|slin|g729)/video=0x0 (nothing), combined - 0x8 (alaw) Sep 6 16:39:38 VERBOSE[19153] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Sep 6 16:39:38 DEBUG[19153] chan_agent.c: Asked for bridged channel on 'SIP/3052-0979ae40'/'Agent/3052', returning 'SIP/3050-097942f8' Sep 6 16:39:38 DEBUG[19153] channel.c: Set channel SIP/3050-097942f8 to write format slin Sep 6 16:39:38 VERBOSE[19153] logger.c: -- Started music on hold, class 'planetarium', on channel 'SIP/3050-097942f8' Sep 6 16:39:38 DEBUG[19153] channel.c: Scheduling timer at 160 sample intervals Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Got a SIP re-invite for call 7c610120743a79ec554afc953f5df4a1@200.196.44.45 Sep 6 16:39:38 VERBOSE[19153] logger.c: We're at 200.196.44.45 port 10012 Sep 6 16:39:38 VERBOSE[19153] logger.c: Adding codec 0x8 (alaw) to SDP Sep 6 16:39:38 VERBOSE[19153] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Sep 6 16:39:38 VERBOSE[19153] logger.c: Reliably Transmitting (no NAT) to 172.16.10.141:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bK28a82477f;received=172.16.10.141 From: ;tag=e949f769a006cca To: "Ramal Teste" ;tag=as2e2bac0a Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 CSeq: 1686507440 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 218 v=0 o=root 19118 19119 IN IP4 200.196.44.45 s=session c=IN IP4 200.196.44.45 t=0 0 m=audio 10012 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Sep 6 16:39:38 DEBUG[19153] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #21 Sep 6 16:39:38 DEBUG[19163] channel.c: Generator got voice, switching to phase locked mode Sep 6 16:39:38 DEBUG[19163] channel.c: Scheduling timer at 0 sample intervals Sep 6 16:39:38 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.141:2217: ACK sip:3050@200.196.44.45 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bK9f348f468 Max-Forwards: 70 Content-Length: 0 To: "Ramal Teste" ;tag=as2e2bac0a From: ;tag=e949f769a006cca Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 CSeq: 1686507440 ACK Contact: User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 0: ACK sip:3050@200.196.44.45 SIP/2.0 (34) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bK9f348f468 (54) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 2: Max-Forwards: 70 (16) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 3: Content-Length: 0 (17) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 4: To: "Ramal Teste" ;tag=as2e2bac0a (57) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 5: From: ;tag=e949f769a006cca (50) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 6: Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 (55) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 7: CSeq: 1686507440 ACK (20) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 8: Contact: (33) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 9: User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (58) Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Header 10: (0) Sep 6 16:39:38 VERBOSE[19153] logger.c: --- (10 headers 0 lines)Sep 6 16:39:38 VERBOSE[19153] logger.c: --- (10 headers 0 lines)--- Sep 6 16:39:38 DEBUG[19153] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Sep 6 16:39:38 DEBUG[19153] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #21 Sep 6 16:39:38 DEBUG[19153] chan_sip.c: Stopping retransmission on '7c610120743a79ec554afc953f5df4a1@200.196.44.45' of Response 1686507440: Match Found Sep 6 16:39:41 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.141:2217: INVITE sip:3053@pfdesenv.planetarium.com.br:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bKfed4f7894 Max-Forwards: 70 Content-Length: 465 To: 3053 From: Leonardo Gomes 3052 ;tag=ebb31aee1a065cc Call-ID: a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 CSeq: 1040787166 INVITE Supported: timer Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Content-Type: application/sdp Contact: Leonardo Gomes 3052 Supported: replaces User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 2062842924 IN IP4 172.16.10.141 s=SIP Call c=IN IP4 172.16.10.141 t=0 0 m=audio 10006 RTP/AVP 0 18 101 102 107 104 105 106 4 8 127 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 BV16/8000 a=rtpmap:102 BV32/16000 a=rtpmap:107 L16/16000 a=rtpmap:104 PCMU/16000 a=rtpmap:105 PCMA/16000 a=rtpmap:106 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-15 a=silenceSupp:off - - - - Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 0: INVITE sip:3053@pfdesenv.planetarium.com.br:5060 SIP/2.0 (56) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bKfed4f7894 (54) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 2: Max-Forwards: 70 (16) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 3: Content-Length: 465 (19) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 4: To: 3053 (52) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 5: From: Leonardo Gomes 3052 ;tag=ebb31aee1a065cc (89) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 6: Call-ID: a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 (55) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 7: CSeq: 1040787166 INVITE (23) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 8: Supported: timer (16) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 9: Allow: NOTIFY (13) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 10: Allow: REFER (12) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 11: Allow: OPTIONS (14) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 12: Allow: INVITE (13) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 13: Allow: ACK (10) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 14: Allow: CANCEL (13) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 15: Allow: BYE (10) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 16: Content-Type: application/sdp (29) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 17: Contact: Leonardo Gomes 3052 (53) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 18: Supported: replaces (19) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 19: User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (58) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 20: (0) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: v=0 (3) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: o=MxSIP 0 2062842924 IN IP4 172.16.10.141 (41) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: s=SIP Call (10) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: c=IN IP4 172.16.10.141 (22) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: t=0 0 (5) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: m=audio 10006 RTP/AVP 0 18 101 102 107 104 105 106 4 8 127 (58) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:101 BV16/8000 (22) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:102 BV32/16000 (23) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:107 L16/16000 (22) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:104 PCMU/16000 (23) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:105 PCMA/16000 (23) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:106 L16/8000 (21) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:4 G723/8000 (20) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:127 telephone-event/8000 (33) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=fmtp:127 0-15 (15) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Sep 6 16:39:41 VERBOSE[19153] logger.c: --- (20 headers 19 lines)Sep 6 16:39:41 VERBOSE[19153] logger.c: --- (20 headers 19 lines)--- Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Allocating new SIP dialog for a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 - INVITE (With RTP) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Begin: parsing SIP "Supported: timer" Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Found SIP option: -timer- Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Matched SIP option: timer Sep 6 16:39:41 DEBUG[19153] chan_sip.c: * SIP extension value: 4 for call a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 Sep 6 16:39:41 VERBOSE[19153] logger.c: Using INVITE request as basis request - a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 Sep 6 16:39:41 VERBOSE[19153] logger.c: Sending to 172.16.10.141 : 5060 (non-NAT) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Setting NAT on RTP to 0 Sep 6 16:39:41 VERBOSE[19153] logger.c: Reliably Transmitting (no NAT) to 172.16.10.141:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bKfed4f7894;received=172.16.10.141 From: Leonardo Gomes 3052 ;tag=ebb31aee1a065cc To: 3053 ;tag=as19c8f3ff Call-ID: a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 CSeq: 1040787166 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1c34627e" Content-Length: 0 --- Sep 6 16:39:41 DEBUG[19153] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #22 Sep 6 16:39:41 VERBOSE[19153] logger.c: Scheduling destruction of call 'a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141' in 15000 ms Sep 6 16:39:41 VERBOSE[19153] logger.c: Found user '3052' Sep 6 16:39:41 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.141:2217: ACK sip:3053@pfdesenv.planetarium.com.br:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bKfed4f7894 Max-Forwards: 70 Content-Length: 0 To: 3053 ;tag=as19c8f3ff From: Leonardo Gomes 3052 ;tag=ebb31aee1a065cc Call-ID: a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 CSeq: 1040787166 ACK User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 0: ACK sip:3053@pfdesenv.planetarium.com.br:5060 SIP/2.0 (53) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bKfed4f7894 (54) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 2: Max-Forwards: 70 (16) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 3: Content-Length: 0 (17) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 4: To: 3053 ;tag=as19c8f3ff (67) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 5: From: Leonardo Gomes 3052 ;tag=ebb31aee1a065cc (89) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 6: Call-ID: a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 (55) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 7: CSeq: 1040787166 ACK (20) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 8: User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (58) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 9: (0) Sep 6 16:39:41 VERBOSE[19153] logger.c: --- (9 headers 0 lines)Sep 6 16:39:41 VERBOSE[19153] logger.c: --- (9 headers 0 lines)--- Sep 6 16:39:41 DEBUG[19153] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Sep 6 16:39:41 DEBUG[19153] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #22 Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Stopping retransmission on 'a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141' of Response 1040787166: Match Found Sep 6 16:39:41 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.141:2217: INVITE sip:3053@pfdesenv.planetarium.com.br:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bK12387c0b4 Max-Forwards: 70 Content-Length: 465 To: 3053 From: Leonardo Gomes 3052 ;tag=ebb31aee1a065cc Call-ID: a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 CSeq: 1040787167 INVITE Supported: timer Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Contact: Leonardo Gomes 3052 Content-Type: application/sdp Supported: replaces Proxy-Authorization:Digest response="a81c10353e67a685ab524b1fa565a605",username="3052",realm="asterisk",nonce="1c34627e",algorithm=MD5,uri="sip:3053@pfdesenv.planetarium.com.br:5060" User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 v=0 o=MxSIP 0 2062842924 IN IP4 172.16.10.141 s=SIP Call c=IN IP4 172.16.10.141 t=0 0 m=audio 10006 RTP/AVP 0 18 101 102 107 104 105 106 4 8 127 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 BV16/8000 a=rtpmap:102 BV32/16000 a=rtpmap:107 L16/16000 a=rtpmap:104 PCMU/16000 a=rtpmap:105 PCMA/16000 a=rtpmap:106 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-15 a=silenceSupp:off - - - - Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 0: INVITE sip:3053@pfdesenv.planetarium.com.br:5060 SIP/2.0 (56) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bK12387c0b4 (54) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 2: Max-Forwards: 70 (16) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 3: Content-Length: 465 (19) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 4: To: 3053 (52) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 5: From: Leonardo Gomes 3052 ;tag=ebb31aee1a065cc (89) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 6: Call-ID: a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 (55) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 7: CSeq: 1040787167 INVITE (23) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 8: Supported: timer (16) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 9: Allow: NOTIFY (13) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 10: Allow: REFER (12) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 11: Allow: OPTIONS (14) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 12: Allow: INVITE (13) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 13: Allow: ACK (10) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 14: Allow: CANCEL (13) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 15: Allow: BYE (10) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 16: Contact: Leonardo Gomes 3052 (53) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 17: Content-Type: application/sdp (29) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 18: Supported: replaces (19) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 19: Proxy-Authorization:Digest response="a81c10353e67a685ab524b1fa565a605",username="3052",realm="asterisk",nonce="1c34627e",algorithm=MD5,uri="sip:3053@pfdesenv.planetarium.com.br:5060" (182) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 20: User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (58) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Header 21: (0) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: v=0 (3) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: o=MxSIP 0 2062842924 IN IP4 172.16.10.141 (41) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: s=SIP Call (10) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: c=IN IP4 172.16.10.141 (22) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: t=0 0 (5) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: m=audio 10006 RTP/AVP 0 18 101 102 107 104 105 106 4 8 127 (58) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:101 BV16/8000 (22) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:102 BV32/16000 (23) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:107 L16/16000 (22) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:104 PCMU/16000 (23) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:105 PCMA/16000 (23) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:106 L16/8000 (21) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:4 G723/8000 (20) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=rtpmap:127 telephone-event/8000 (33) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=fmtp:127 0-15 (15) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Sep 6 16:39:41 VERBOSE[19153] logger.c: --- (21 headers 19 lines)Sep 6 16:39:41 VERBOSE[19153] logger.c: --- (21 headers 19 lines)--- Sep 6 16:39:41 DEBUG[19153] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Sep 6 16:39:41 VERBOSE[19153] logger.c: Using INVITE request as basis request - a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 Sep 6 16:39:41 VERBOSE[19153] logger.c: Sending to 172.16.10.141 : 5060 (non-NAT) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Setting NAT on RTP to 0 Sep 6 16:39:41 VERBOSE[19153] logger.c: Found user '3052' Sep 6 16:39:41 VERBOSE[19153] logger.c: Found RTP audio format 0 Sep 6 16:39:41 VERBOSE[19153] logger.c: Found RTP audio format 18 Sep 6 16:39:41 VERBOSE[19153] logger.c: Found RTP audio format 101 Sep 6 16:39:41 VERBOSE[19153] logger.c: Found RTP audio format 102 Sep 6 16:39:41 VERBOSE[19153] logger.c: Found RTP audio format 107 Sep 6 16:39:41 VERBOSE[19153] logger.c: Found RTP audio format 104 Sep 6 16:39:41 VERBOSE[19153] logger.c: Found RTP audio format 105 Sep 6 16:39:41 VERBOSE[19153] logger.c: Found RTP audio format 106 Sep 6 16:39:41 VERBOSE[19153] logger.c: Found RTP audio format 4 Sep 6 16:39:41 VERBOSE[19153] logger.c: Found RTP audio format 8 Sep 6 16:39:41 VERBOSE[19153] logger.c: Found RTP audio format 127 Sep 6 16:39:41 VERBOSE[19153] logger.c: Peer audio RTP is at port 172.16.10.141:10006 Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Peer audio RTP is at port 172.16.10.141:10006 Sep 6 16:39:41 VERBOSE[19153] logger.c: Found description format PCMU Sep 6 16:39:41 VERBOSE[19153] logger.c: Found description format G729 Sep 6 16:39:41 VERBOSE[19153] logger.c: Found description format BV16 Sep 6 16:39:41 VERBOSE[19153] logger.c: Found description format BV32 Sep 6 16:39:41 VERBOSE[19153] logger.c: Found description format L16 Sep 6 16:39:41 VERBOSE[19153] logger.c: Found description format PCMU Sep 6 16:39:41 VERBOSE[19153] logger.c: Found description format PCMA Sep 6 16:39:41 VERBOSE[19153] logger.c: Found description format L16 Sep 6 16:39:41 VERBOSE[19153] logger.c: Found description format G723 Sep 6 16:39:41 VERBOSE[19153] logger.c: Found description format PCMA Sep 6 16:39:41 VERBOSE[19153] logger.c: Found description format telephone-event Sep 6 16:39:41 VERBOSE[19153] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x14d (g723|ulaw|alaw|slin|g729)/video=0x0 (nothing), combined - 0x8 (alaw) Sep 6 16:39:41 VERBOSE[19153] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Checking SIP call limits for device 3052 Sep 6 16:39:41 DEBUG[19153] chan_sip.c: Updating call counter for incoming call Sep 6 16:39:41 VERBOSE[19153] logger.c: Looking for 3053 in sip_to_anywhere (domain pfdesenv.planetarium.com.br) Sep 6 16:39:41 DEBUG[19153] chan_sip.c: build_route: Contact hop: Leonardo Gomes 3052 Sep 6 16:39:41 VERBOSE[19153] logger.c: list_route: hop: Sep 6 16:39:41 VERBOSE[19153] logger.c: Transmitting (no NAT) to 172.16.10.141:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bK12387c0b4;received=172.16.10.141 From: Leonardo Gomes 3052 ;tag=ebb31aee1a065cc To: 3053 Call-ID: a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 CSeq: 1040787167 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Sep 6 16:39:41 DEBUG[19137] chan_sip.c: Checking device state for peer 3052 Sep 6 16:39:41 DEBUG[19137] devicestate.c: Changing state for SIP/3052 - state 2 (In use) Sep 6 16:39:41 DEBUG[19137] chan_sip.c: Checking device state for peer 3052 Sep 6 16:39:41 DEBUG[19177] pbx.c: Function result is '0' Sep 6 16:39:41 DEBUG[19177] pbx.c: Function result is '3052' Sep 6 16:39:41 DEBUG[19177] pbx.c: Launching 'NoOp' Sep 6 16:39:41 VERBOSE[19177] logger.c: -- Executing NoOp("SIP/3052-097a4608", "BEFORE SETGROUP: Group: 3052 Count: 0 Accountcode: 3052") in new stack Sep 6 16:39:41 DEBUG[19177] pbx.c: Function result is '0' Sep 6 16:39:41 DEBUG[19177] pbx.c: Launching 'NoOp' Sep 6 16:39:41 VERBOSE[19177] logger.c: -- Executing NoOp("SIP/3052-097a4608", "BEFORE SETOUTBOUNDGROUP: Group: 3053 Count: 0") in new stack Sep 6 16:39:41 DEBUG[19177] pbx.c: Function result is '3052' Sep 6 16:39:41 DEBUG[19177] pbx.c: Launching 'Set' Sep 6 16:39:41 VERBOSE[19177] logger.c: -- Executing Set("SIP/3052-097a4608", "GROUP()=3052") in new stack Sep 6 16:39:41 DEBUG[19177] pbx.c: Launching 'Set' Sep 6 16:39:41 VERBOSE[19177] logger.c: -- Executing Set("SIP/3052-097a4608", "OUTBOUND_GROUP=3053") in new stack Sep 6 16:39:41 DEBUG[19177] pbx.c: Function result is '1' Sep 6 16:39:41 DEBUG[19177] pbx.c: Launching 'NoOp' Sep 6 16:39:41 VERBOSE[19177] logger.c: -- Executing NoOp("SIP/3052-097a4608", "AFTER SETGROUP: Group: 3052 Count: 1") in new stack Sep 6 16:39:41 DEBUG[19177] pbx.c: Function result is '0' Sep 6 16:39:41 DEBUG[19177] pbx.c: Expression result is '0' Sep 6 16:39:41 DEBUG[19177] pbx.c: Launching 'GotoIf' Sep 6 16:39:41 VERBOSE[19177] logger.c: -- Executing GotoIf("SIP/3052-097a4608", "0 ? 1000") in new stack Sep 6 16:39:41 DEBUG[19177] pbx.c: Not taking any branch Sep 6 16:39:41 DEBUG[19177] pbx.c: Launching 'Dial' Sep 6 16:39:41 VERBOSE[19177] logger.c: -- Executing Dial("SIP/3052-097a4608", "SIP/3053|15|tT") in new stack Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Setting NAT on RTP to 0 Sep 6 16:39:41 DEBUG[19177] channel.c: Not copying variable STACK-sip_to_anywhere-3053-7. Sep 6 16:39:41 DEBUG[19177] channel.c: Not copying variable STACK-sip_to_anywhere-3053-6. Sep 6 16:39:41 DEBUG[19177] channel.c: Not copying variable STACK-sip_to_anywhere-3053-5. Sep 6 16:39:41 DEBUG[19177] channel.c: Not copying variable OUTBOUND_GROUP. Sep 6 16:39:41 DEBUG[19177] channel.c: Not copying variable STACK-sip_to_anywhere-3053-4. Sep 6 16:39:41 DEBUG[19177] channel.c: Not copying variable GROUP. Sep 6 16:39:41 DEBUG[19177] channel.c: Not copying variable STACK-sip_to_anywhere-3053-3. Sep 6 16:39:41 DEBUG[19177] channel.c: Not copying variable STACK-sip_to_anywhere-3053-2. Sep 6 16:39:41 DEBUG[19177] channel.c: Not copying variable STACK-sip_to_anywhere-3053-1. Sep 6 16:39:41 DEBUG[19177] channel.c: Not copying variable SIPCALLID. Sep 6 16:39:41 DEBUG[19177] channel.c: Not copying variable SIPUSERAGENT. Sep 6 16:39:41 DEBUG[19177] channel.c: Not copying variable SIPDOMAIN. Sep 6 16:39:41 DEBUG[19177] channel.c: Not copying variable SIPURI. Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Outgoing Call for 3053 Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Updating call counter for outgoing call Sep 6 16:39:41 VERBOSE[19177] logger.c: We're at 200.196.44.45 port 12878 Sep 6 16:39:41 VERBOSE[19177] logger.c: Adding codec 0x8 (alaw) to SDP Sep 6 16:39:41 VERBOSE[19177] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Header 0: INVITE sip:3053@172.16.10.135:5060 SIP/2.0 (42) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK046e0686;rport (64) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Header 2: From: "Leonardo Gomes" ;tag=as225fc9b2 (68) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Header 3: To: (33) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Header 4: Contact: (39) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Header 5: Call-ID: 3e42209b40751fb15a14d8a3097032d0@200.196.44.45 (55) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Header 6: CSeq: 102 INVITE (16) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Header 8: Max-Forwards: 70 (16) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Header 9: Date: Wed, 06 Sep 2006 19:39:41 GMT (35) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Header 11: Content-Type: application/sdp (29) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Header 12: Content-Length: 218 (19) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Header 13: (0) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Line: v=0 (3) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Line: o=root 19118 19118 IN IP4 200.196.44.45 (39) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Line: s=session (9) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Line: c=IN IP4 200.196.44.45 (22) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Line: t=0 0 (5) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Line: m=audio 12878 RTP/AVP 8 101 (27) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Line: a=fmtp:101 0-16 (15) Sep 6 16:39:41 DEBUG[19177] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Sep 6 16:39:41 VERBOSE[19177] logger.c: 13 headers, 10 lines Sep 6 16:39:41 VERBOSE[19177] logger.c: Reliably Transmitting (no NAT) to 172.16.10.135:5060: INVITE sip:3053@172.16.10.135:5060 SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK046e0686;rport From: "Leonardo Gomes" ;tag=as225fc9b2 To: Contact: Call-ID: 3e42209b40751fb15a14d8a3097032d0@200.196.44.45 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 06 Sep 2006 19:39:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 218 v=0 o=root 19118 19118 IN IP4 200.196.44.45 s=session c=IN IP4 200.196.44.45 t=0 0 m=audio 12878 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Sep 6 16:39:41 DEBUG[19177] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #24 Sep 6 16:39:41 VERBOSE[19177] logger.c: -- Called 3053 Sep 6 16:39:41 DEBUG[19178] app_queue.c: Device 'SIP/3052' changed to state '2' (In use) Sep 6 16:39:42 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.135:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 200.196.44.45:5060;rport=5060;received=200.196.44.45;branch=z9hG4bK046e0686 From: "Leonardo Gomes" ;tag=as225fc9b2 To: Contact: Call-ID: 3e42209b40751fb15a14d8a3097032d0@200.196.44.45 CSeq: 102 INVITE Content-Length: 0 Sep 6 16:39:42 DEBUG[19153] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) Sep 6 16:39:42 DEBUG[19153] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;rport=5060;received=200.196.44.45;branch=z9hG4bK046e0686 (92) Sep 6 16:39:42 DEBUG[19153] chan_sip.c: Header 2: From: "Leonardo Gomes" ;tag=as225fc9b2 (68) Sep 6 16:39:42 DEBUG[19153] chan_sip.c: Header 3: To: (33) Sep 6 16:39:42 DEBUG[19153] chan_sip.c: Header 4: Contact: (38) Sep 6 16:39:42 DEBUG[19153] chan_sip.c: Header 5: Call-ID: 3e42209b40751fb15a14d8a3097032d0@200.196.44.45 (55) Sep 6 16:39:42 DEBUG[19153] chan_sip.c: Header 6: CSeq: 102 INVITE (16) Sep 6 16:39:42 DEBUG[19153] chan_sip.c: Header 7: Content-Length: 0 (17) Sep 6 16:39:42 DEBUG[19153] chan_sip.c: Header 8: (0) Sep 6 16:39:42 VERBOSE[19153] logger.c: --- (8 headers 0 lines)Sep 6 16:39:42 VERBOSE[19153] logger.c: --- (8 headers 0 lines)--- Sep 6 16:39:42 DEBUG[19153] chan_sip.c: *** SIP TIMER: Cancelling retransmission #24 - INVITE (got response) Sep 6 16:39:42 DEBUG[19153] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3e42209b40751fb15a14d8a3097032d0@200.196.44.45' Request 102: Found Sep 6 16:39:42 DEBUG[19153] chan_sip.c: SIP response 100 to standard invite Sep 6 16:39:42 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.135:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 200.196.44.45:5060;rport=5060;received=200.196.44.45;branch=z9hG4bK046e0686 From: "Leonardo Gomes" ;tag=as225fc9b2 To: ;tag=600b8f56 Contact: Call-ID: 3e42209b40751fb15a14d8a3097032d0@200.196.44.45 CSeq: 102 INVITE Content-Length: 0 Sep 6 16:39:42 DEBUG[19153] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) Sep 6 16:39:42 DEBUG[19153] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;rport=5060;received=200.196.44.45;branch=z9hG4bK046e0686 (92) Sep 6 16:39:42 DEBUG[19153] chan_sip.c: Header 2: From: "Leonardo Gomes" ;tag=as225fc9b2 (68) Sep 6 16:39:42 DEBUG[19153] chan_sip.c: Header 3: To: ;tag=600b8f56 (46) Sep 6 16:39:42 DEBUG[19153] chan_sip.c: Header 4: Contact: (38) Sep 6 16:39:42 DEBUG[19153] chan_sip.c: Header 5: Call-ID: 3e42209b40751fb15a14d8a3097032d0@200.196.44.45 (55) Sep 6 16:39:42 DEBUG[19153] chan_sip.c: Header 6: CSeq: 102 INVITE (16) Sep 6 16:39:42 DEBUG[19153] chan_sip.c: Header 7: Content-Length: 0 (17) Sep 6 16:39:42 DEBUG[19153] chan_sip.c: Header 8: (0) Sep 6 16:39:42 VERBOSE[19153] logger.c: --- (8 headers 0 lines)Sep 6 16:39:42 VERBOSE[19153] logger.c: --- (8 headers 0 lines)--- Sep 6 16:39:42 DEBUG[19153] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3e42209b40751fb15a14d8a3097032d0@200.196.44.45' Request 102: Found Sep 6 16:39:42 DEBUG[19153] chan_sip.c: SIP response 180 to standard invite Sep 6 16:39:42 VERBOSE[19177] logger.c: -- SIP/3053-097abe20 is ringing Sep 6 16:39:42 VERBOSE[19177] logger.c: Transmitting (no NAT) to 172.16.10.141:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bK12387c0b4;received=172.16.10.141 From: Leonardo Gomes 3052 ;tag=ebb31aee1a065cc To: 3053 ;tag=as29a2fca0 Call-ID: a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 CSeq: 1040787167 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Sep 6 16:39:42 DEBUG[19137] chan_sip.c: Checking device state for peer 3053 Sep 6 16:39:42 DEBUG[19137] devicestate.c: Changing state for SIP/3053 - state 6 (Ringing) Sep 6 16:39:42 DEBUG[19179] app_queue.c: Device 'SIP/3053' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. Sep 6 16:39:43 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.135:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.196.44.45:5060;rport=5060;received=200.196.44.45;branch=z9hG4bK046e0686 From: "Leonardo Gomes" ;tag=as225fc9b2 To: ;tag=600b8f56 Contact: Call-ID: 3e42209b40751fb15a14d8a3097032d0@200.196.44.45 CSeq: 102 INVITE Content-Type: application/sdp Content-Length: 197 v=0 o=- 41307 0 IN IP4 172.16.10.135 s=SIP CALL c=IN IP4 172.16.10.135 t=0 0 m=audio 10000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;rport=5060;received=200.196.44.45;branch=z9hG4bK046e0686 (92) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 2: From: "Leonardo Gomes" ;tag=as225fc9b2 (68) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 3: To: ;tag=600b8f56 (46) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 4: Contact: (38) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 5: Call-ID: 3e42209b40751fb15a14d8a3097032d0@200.196.44.45 (55) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 6: CSeq: 102 INVITE (16) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 7: Content-Type: application/sdp (29) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 8: Content-Length: 197 (19) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 9: (0) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Line: v=0 (3) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Line: o=- 41307 0 IN IP4 172.16.10.135 (32) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Line: s=SIP CALL (10) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Line: c=IN IP4 172.16.10.135 (22) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Line: t=0 0 (5) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Line: m=audio 10000 RTP/AVP 8 101 (27) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Line: a=fmtp:101 0-15 (15) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Line: a=sendrecv (10) Sep 6 16:39:43 VERBOSE[19153] logger.c: --- (9 headers 10 lines)Sep 6 16:39:43 VERBOSE[19153] logger.c: --- (9 headers 10 lines)--- Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Acked pending invite 102 Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Stopping retransmission on '3e42209b40751fb15a14d8a3097032d0@200.196.44.45' of Request 102: Match Found Sep 6 16:39:43 DEBUG[19153] chan_sip.c: SIP response 200 to standard invite Sep 6 16:39:43 VERBOSE[19153] logger.c: Found RTP audio format 8 Sep 6 16:39:43 VERBOSE[19153] logger.c: Found RTP audio format 101 Sep 6 16:39:43 VERBOSE[19153] logger.c: Peer audio RTP is at port 172.16.10.135:10000 Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Peer audio RTP is at port 172.16.10.135:10000 Sep 6 16:39:43 VERBOSE[19153] logger.c: Found description format PCMA Sep 6 16:39:43 VERBOSE[19153] logger.c: Found description format telephone-event Sep 6 16:39:43 VERBOSE[19153] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Sep 6 16:39:43 VERBOSE[19153] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: build_route: Contact hop: Sep 6 16:39:43 VERBOSE[19153] logger.c: list_route: hop: Sep 6 16:39:43 VERBOSE[19153] logger.c: set_destination: Parsing for address/port to send to Sep 6 16:39:43 VERBOSE[19153] logger.c: set_destination: set destination to 172.16.10.135, port 5060 Sep 6 16:39:43 VERBOSE[19153] logger.c: Transmitting (no NAT) to 172.16.10.135:5060: ACK sip:3053@172.16.10.135:5060 SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK13abc4ef;rport From: "Leonardo Gomes" ;tag=as225fc9b2 To: ;tag=600b8f56 Contact: Call-ID: 3e42209b40751fb15a14d8a3097032d0@200.196.44.45 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Sep 6 16:39:43 VERBOSE[19177] logger.c: -- SIP/3053-097abe20 answered SIP/3052-097a4608 Sep 6 16:39:43 DEBUG[19177] chan_sip.c: sip_answer(SIP/3052-097a4608) Sep 6 16:39:43 VERBOSE[19177] logger.c: We're at 200.196.44.45 port 16474 Sep 6 16:39:43 VERBOSE[19177] logger.c: Adding codec 0x8 (alaw) to SDP Sep 6 16:39:43 VERBOSE[19177] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Sep 6 16:39:43 VERBOSE[19177] logger.c: Reliably Transmitting (no NAT) to 172.16.10.141:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bK12387c0b4;received=172.16.10.141 From: Leonardo Gomes 3052 ;tag=ebb31aee1a065cc To: 3053 ;tag=as29a2fca0 Call-ID: a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 CSeq: 1040787167 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 218 v=0 o=root 19118 19118 IN IP4 200.196.44.45 s=session c=IN IP4 200.196.44.45 t=0 0 m=audio 16474 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Sep 6 16:39:43 DEBUG[19177] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #25 Sep 6 16:39:43 DEBUG[19137] chan_sip.c: Checking device state for peer 3053 Sep 6 16:39:43 DEBUG[19137] devicestate.c: Changing state for SIP/3053 - state 2 (In use) Sep 6 16:39:43 DEBUG[19137] chan_sip.c: Checking device state for peer 3052 Sep 6 16:39:43 DEBUG[19137] devicestate.c: Changing state for SIP/3052 - state 2 (In use) Sep 6 16:39:43 DEBUG[19137] chan_sip.c: Checking device state for peer 3052 Sep 6 16:39:43 DEBUG[19180] app_queue.c: Device 'SIP/3053' changed to state '2' (In use) but we don't care because they're not a member of any queue. Sep 6 16:39:43 DEBUG[19181] app_queue.c: Device 'SIP/3052' changed to state '2' (In use) Sep 6 16:39:43 DEBUG[19177] rtp.c: Ooh, format changed from unknown to alaw Sep 6 16:39:43 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.141:2217: ACK sip:3053@200.196.44.45 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bKe1e7de4a2 Max-Forwards: 70 Content-Length: 0 To: 3053 ;tag=as29a2fca0 From: Leonardo Gomes 3052 ;tag=ebb31aee1a065cc Call-ID: a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 CSeq: 1040787167 ACK Contact: Leonardo Gomes 3052 Proxy-Authorization:Digest response="52fcd39b4cf3c39542ee6b565779e909",username="3052",realm="asterisk",nonce="1c34627e",algorithm=MD5,uri="sip:3053@200.196.44.45" User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 0: ACK sip:3053@200.196.44.45 SIP/2.0 (34) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bKe1e7de4a2 (54) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 2: Max-Forwards: 70 (16) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 3: Content-Length: 0 (17) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 4: To: 3053 ;tag=as29a2fca0 (67) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 5: From: Leonardo Gomes 3052 ;tag=ebb31aee1a065cc (89) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 6: Call-ID: a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 (55) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 7: CSeq: 1040787167 ACK (20) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 8: Contact: Leonardo Gomes 3052 (53) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 9: Proxy-Authorization:Digest response="52fcd39b4cf3c39542ee6b565779e909",username="3052",realm="asterisk",nonce="1c34627e",algorithm=MD5,uri="sip:3053@200.196.44.45" (163) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 10: User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (58) Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Header 11: (0) Sep 6 16:39:43 VERBOSE[19153] logger.c: --- (11 headers 0 lines)Sep 6 16:39:43 VERBOSE[19153] logger.c: --- (11 headers 0 lines)--- Sep 6 16:39:43 DEBUG[19153] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Sep 6 16:39:43 DEBUG[19153] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #25 Sep 6 16:39:43 DEBUG[19153] chan_sip.c: Stopping retransmission on 'a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141' of Response 1040787167: Match Found Sep 6 16:39:43 DEBUG[19177] rtp.c: Ooh, format changed from unknown to alaw Sep 6 16:39:46 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.135:5060: Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 0: (0) Sep 6 16:39:46 VERBOSE[19153] logger.c: --- (0 headers 0 lines)Sep 6 16:39:46 VERBOSE[19153] logger.c: --- (0 headers 0 lines) Nat keepalive Sep 6 16:39:46 VERBOSE[19153] logger.c: --- (0 headers 0 lines) Nat keepalive --- Sep 6 16:39:46 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.141:2217: REFER sip:3050@200.196.44.45 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bKb6190a830 Max-Forwards: 70 Content-Length: 0 To: "Ramal Teste" ;tag=as2e2bac0a From: ;tag=e949f769a006cca Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 CSeq: 1686507441 REFER Supported: timer Contact: Refer-To: sip:3053@pfdesenv.planetarium.com.br:5060?Replaces=a0c76ddcaa74e5de68d55f3b3398d0bf%40172.16.10.141%3bto-tag%3das29a2fca0%3bfrom-tag%3debb31aee1a065cc Referred-By: User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 0: REFER sip:3050@200.196.44.45 SIP/2.0 (36) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bKb6190a830 (54) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 2: Max-Forwards: 70 (16) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 3: Content-Length: 0 (17) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 4: To: "Ramal Teste" ;tag=as2e2bac0a (57) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 5: From: ;tag=e949f769a006cca (50) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 6: Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 (55) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 7: CSeq: 1686507441 REFER (22) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 8: Supported: timer (16) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 9: Contact: (33) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 10: Refer-To: sip:3053@pfdesenv.planetarium.com.br:5060?Replaces=a0c76ddcaa74e5de68d55f3b3398d0bf%40172.16.10.141%3bto-tag%3das29a2fca0%3bfrom-tag%3debb31aee1a065cc (160) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 11: Referred-By: (37) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 12: User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (58) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 13: (0) Sep 6 16:39:46 VERBOSE[19153] logger.c: --- (13 headers 0 lines)Sep 6 16:39:46 VERBOSE[19153] logger.c: --- (13 headers 0 lines)--- Sep 6 16:39:46 DEBUG[19153] chan_sip.c: **** Received REFER (9) - Command in SIP REFER Sep 6 16:39:46 DEBUG[19153] chan_sip.c: SIP call transfer received for call 7c610120743a79ec554afc953f5df4a1@200.196.44.45 (REFER)! Sep 6 16:39:46 VERBOSE[19153] logger.c: Transfer to 3053 in sip_to_anywhere Sep 6 16:39:46 VERBOSE[19153] logger.c: Transfer from 3052 in sip_to_anywhere Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Assigning Replace-Call-ID Info a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 to REPLACE_CALL_ID Sep 6 16:39:46 DEBUG[19153] chan_sip.c: 202 Accepted (supervised) Sep 6 16:39:46 DEBUG[19153] chan_agent.c: Asked for bridged channel on 'SIP/3052-0979ae40'/'Agent/3052', returning 'SIP/3050-097942f8' Sep 6 16:39:46 DEBUG[19153] channel.c: Set channel SIP/3050-097942f8 to write format alaw Sep 6 16:39:46 VERBOSE[19153] logger.c: -- Stopped music on hold on SIP/3050-097942f8 Sep 6 16:39:46 DEBUG[19153] channel.c: Scheduling timer at 0 sample intervals Sep 6 16:39:46 DEBUG[19153] chan_agent.c: Asked for bridged channel on 'SIP/3050-097942f8'/'Agent/3052', returning 'SIP/3052-0979ae40' Sep 6 16:39:46 DEBUG[19153] channel.c: Planning to masquerade channel Agent/3052 into the structure of SIP/3052-097a4608 Sep 6 16:39:46 DEBUG[19153] channel.c: Done planning to masquerade channel Agent/3052 into the structure of SIP/3052-097a4608 Sep 6 16:39:46 VERBOSE[19153] logger.c: Transmitting (no NAT) to 172.16.10.141:5060: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 172.16.10.141;branch=z9hG4bKb6190a830;received=172.16.10.141 From: ;tag=e949f769a006cca To: "Ramal Teste" ;tag=as2e2bac0a Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 CSeq: 1686507441 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Sep 6 16:39:46 VERBOSE[19153] logger.c: set_destination: Parsing for address/port to send to Sep 6 16:39:46 VERBOSE[19153] logger.c: set_destination: set destination to 172.16.10.141, port 5060 Sep 6 16:39:46 VERBOSE[19153] logger.c: Reliably Transmitting (no NAT) to 172.16.10.141:5060: NOTIFY sip:3052@172.16.10.141 SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK265d4c71;rport From: "Ramal Teste" ;tag=as2e2bac0a To: ;tag=e949f769a006cca Contact: Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=1686507441 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- Sep 6 16:39:46 DEBUG[19153] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #26 Sep 6 16:39:46 VERBOSE[19153] logger.c: set_destination: Parsing for address/port to send to Sep 6 16:39:46 VERBOSE[19153] logger.c: set_destination: set destination to 172.16.10.141, port 5060 Sep 6 16:39:46 VERBOSE[19153] logger.c: Reliably Transmitting (no NAT) to 172.16.10.141:5060: BYE sip:3052@172.16.10.141 SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK071cad7c;rport From: "Ramal Teste" ;tag=as2e2bac0a To: ;tag=e949f769a006cca Contact: Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Sep 6 16:39:46 DEBUG[19153] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #27 Sep 6 16:39:46 DEBUG[19177] channel.c: Actually Masquerading Agent/3052(6) into the structure of SIP/3052-097a4608(6) Sep 6 16:39:46 DEBUG[19177] channel.c: Got clone lock for masquerade on 'Agent/3052' at 0x979a304 Sep 6 16:39:46 DEBUG[19177] chan_sip.c: Hangup call Agent/3052, SIP callid a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141) Sep 6 16:39:46 DEBUG[19177] chan_sip.c: update_call_counter(3052) - decrement call limit counter Sep 6 16:39:46 DEBUG[19177] chan_sip.c: Updating call counter for incoming call Sep 6 16:39:46 VERBOSE[19177] logger.c: Scheduling destruction of call 'a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141' in 32000 ms Sep 6 16:39:46 VERBOSE[19177] logger.c: set_destination: Parsing for address/port to send to Sep 6 16:39:46 VERBOSE[19177] logger.c: set_destination: set destination to 172.16.10.141, port 5060 Sep 6 16:39:46 VERBOSE[19177] logger.c: Reliably Transmitting (no NAT) to 172.16.10.141:5060: BYE sip:3052@172.16.10.141 SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK1783ed58;rport From: 3053 ;tag=as29a2fca0 To: Leonardo Gomes 3052 ;tag=ebb31aee1a065cc Contact: Call-ID: a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Sep 6 16:39:46 DEBUG[19177] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #29 Sep 6 16:39:46 DEBUG[19177] channel.c: Putting channel Agent/3052 in 8/8 formats Sep 6 16:39:46 DEBUG[19177] channel.c: Released clone lock on 'SIP/3052-097a4608' Sep 6 16:39:46 DEBUG[19177] channel.c: Done Masquerading Agent/3052 (6) Sep 6 16:39:46 DEBUG[19163] channel.c: Didn't get a frame from channel: SIP/3052-097a4608 Sep 6 16:39:46 DEBUG[19163] channel.c: Bridge stops bridging channels SIP/3050-097942f8 and SIP/3052-097a4608 Sep 6 16:39:46 DEBUG[19163] channel.c: Hanging up zombie 'SIP/3052-097a4608' Sep 6 16:39:46 DEBUG[19163] pbx.c: Spawn extension (sip_to_anywhere,3097,1) exited non-zero on 'SIP/3050-097942f8' Sep 6 16:39:46 DEBUG[19163] channel.c: Hanging up channel 'SIP/3050-097942f8' Sep 6 16:39:46 DEBUG[19163] chan_sip.c: Hangup call SIP/3050-097942f8, SIP callid wFJkP8nIZFcMprRv@172.16.10.124) Sep 6 16:39:46 DEBUG[19163] chan_sip.c: update_call_counter(3050) - decrement call limit counter Sep 6 16:39:46 DEBUG[19163] chan_sip.c: Updating call counter for incoming call Sep 6 16:39:46 VERBOSE[19163] logger.c: Scheduling destruction of call 'wFJkP8nIZFcMprRv@172.16.10.124' in 32000 ms Sep 6 16:39:46 VERBOSE[19163] logger.c: set_destination: Parsing for address/port to send to Sep 6 16:39:46 VERBOSE[19163] logger.c: set_destination: set destination to 172.16.10.124, port 5060 Sep 6 16:39:46 VERBOSE[19163] logger.c: Reliably Transmitting (NAT) to 172.16.10.124:5060: BYE sip:3050@172.16.10.124:5060 SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK19cae55c;rport From: "3097" ;tag=as3d2e9a87 To: "3050" ;tag=OATvZpAyGXpN3cwZ Contact: Call-ID: wFJkP8nIZFcMprRv@172.16.10.124 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Sep 6 16:39:46 DEBUG[19163] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #31 Sep 6 16:39:46 DEBUG[19137] chan_sip.c: Checking device state for peer 3052 Sep 6 16:39:46 DEBUG[19137] devicestate.c: Changing state for SIP/3052 - state 2 (In use) Sep 6 16:39:46 DEBUG[19137] chan_sip.c: Checking device state for peer 3052 Sep 6 16:39:46 DEBUG[19137] chan_sip.c: Checking device state for peer 3050 Sep 6 16:39:46 DEBUG[19137] devicestate.c: Changing state for SIP/3050 - state 1 (Not in use) Sep 6 16:39:46 DEBUG[19182] app_queue.c: Device 'SIP/3052' changed to state '2' (In use) Sep 6 16:39:46 DEBUG[19183] app_queue.c: Device 'SIP/3050' changed to state '1' (Not in use) Sep 6 16:39:46 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.124:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK19cae55c;rport Call-ID: wFJkP8nIZFcMprRv@172.16.10.124 CSeq: 102 BYE From: "3097" ;tag=as3d2e9a87 To: "3050" ;tag=OATvZpAyGXpN3cwZ Contact: Content-Length: 0 Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK19cae55c;rport (64) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 2: Call-ID: wFJkP8nIZFcMprRv@172.16.10.124 (39) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 3: CSeq: 102 BYE (13) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 4: From: "3097" ;tag=as3d2e9a87 (66) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 5: To: "3050" ;tag=OATvZpAyGXpN3cwZ (70) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 6: Contact: (38) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 7: Content-Length: 0 (17) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 8: (0) Sep 6 16:39:46 VERBOSE[19153] logger.c: --- (8 headers 0 lines)Sep 6 16:39:46 VERBOSE[19153] logger.c: --- (8 headers 0 lines)--- Sep 6 16:39:46 DEBUG[19153] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #31 Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Stopping retransmission on 'wFJkP8nIZFcMprRv@172.16.10.124' of Request 102: Match Found Sep 6 16:39:46 VERBOSE[19153] logger.c: Destroying call 'wFJkP8nIZFcMprRv@172.16.10.124' Sep 6 16:39:46 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.141:2217: SIP/2.0 200 OK Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 CSeq: 103 NOTIFY From: "Ramal Teste" ;tag=as2e2bac0a To: ;tag=e949f769a006cca Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK265d4c71;rport Content-Length: 0 Supported: replaces User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 1: Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 (55) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 2: CSeq: 103 NOTIFY (16) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 3: From: "Ramal Teste" ;tag=as2e2bac0a (59) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 4: To: ;tag=e949f769a006cca (48) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 5: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK265d4c71;rport (64) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 6: Content-Length: 0 (17) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 7: Supported: replaces (19) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 8: User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (58) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 9: (0) Sep 6 16:39:46 VERBOSE[19153] logger.c: --- (9 headers 0 lines)Sep 6 16:39:46 VERBOSE[19153] logger.c: --- (9 headers 0 lines)--- Sep 6 16:39:46 DEBUG[19153] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #26 Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Stopping retransmission on '7c610120743a79ec554afc953f5df4a1@200.196.44.45' of Request 103: Match Found Sep 6 16:39:46 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.141:2217: SIP/2.0 200 OK Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 CSeq: 104 BYE From: "Ramal Teste" ;tag=as2e2bac0a To: ;tag=e949f769a006cca Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK071cad7c;rport Content-Length: 0 Supported: replaces User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 1: Call-ID: 7c610120743a79ec554afc953f5df4a1@200.196.44.45 (55) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 2: CSeq: 104 BYE (13) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 3: From: "Ramal Teste" ;tag=as2e2bac0a (59) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 4: To: ;tag=e949f769a006cca (48) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 5: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK071cad7c;rport (64) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 6: Content-Length: 0 (17) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 7: Supported: replaces (19) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 8: User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (58) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 9: (0) Sep 6 16:39:46 VERBOSE[19153] logger.c: --- (9 headers 0 lines)Sep 6 16:39:46 VERBOSE[19153] logger.c: --- (9 headers 0 lines)--- Sep 6 16:39:46 DEBUG[19153] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #27 Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Stopping retransmission on '7c610120743a79ec554afc953f5df4a1@200.196.44.45' of Request 104: Match Found Sep 6 16:39:46 VERBOSE[19153] logger.c: <-- SIP read from 172.16.10.141:2217: SIP/2.0 200 OK Call-ID: a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 CSeq: 102 BYE From: 3053 ;tag=as29a2fca0 To: Leonardo Gomes 3052 ;tag=ebb31aee1a065cc Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK1783ed58;rport Content-Length: 0 Supported: replaces User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 1: Call-ID: a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141 (55) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 2: CSeq: 102 BYE (13) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 3: From: 3053 ;tag=as29a2fca0 (69) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 4: To: Leonardo Gomes 3052 ;tag=ebb31aee1a065cc (87) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 5: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK1783ed58;rport (64) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 6: Content-Length: 0 (17) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 7: Supported: replaces (19) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 8: User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 (58) Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Header 9: (0) Sep 6 16:39:46 VERBOSE[19153] logger.c: --- (9 headers 0 lines)Sep 6 16:39:46 VERBOSE[19153] logger.c: --- (9 headers 0 lines)--- Sep 6 16:39:46 DEBUG[19153] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #29 Sep 6 16:39:46 DEBUG[19153] chan_sip.c: Stopping retransmission on 'a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141' of Request 102: Match Found Sep 6 16:39:46 VERBOSE[19153] logger.c: Destroying call 'a0c76ddcaa74e5de68d55f3b3398d0bf@172.16.10.141' Sep 6 16:39:54 DEBUG[19156] chan_agent.c: Asked for bridged channel on 'SIP/3053-097abe20'/'Agent/3052', returning 'SIP/3052-0979ae40'