Jun 21 17:59:33 VERBOSE[25164] logger.c: Asterisk Event Logger restarted Jun 21 17:59:33 VERBOSE[25164] logger.c: Asterisk Queue Logger restarted Jun 21 17:59:35 DEBUG[25145] chan_iax2.c: Checking device state for device 3051 Jun 21 17:59:35 DEBUG[25145] chan_iax2.c: iax2_devicestate(3051): Found peer. What's device state of 3051? addr=1896485036, defaddr=0 maxms=2000, lastms=8 Jun 21 17:59:35 DEBUG[25145] devicestate.c: Changing state for IAX2/3051 - state 1 (Not in use) Jun 21 17:59:35 DEBUG[25167] app_queue.c: Device 'IAX2/3051' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Jun 21 17:59:35 DEBUG[25157] db.c: Unable to find key 'si-000fd3000a41' in family 'iax/provisioning/cache' Jun 21 17:59:35 DEBUG[25145] chan_iax2.c: Checking device state for device 3051 Jun 21 17:59:35 DEBUG[25145] chan_iax2.c: iax2_devicestate(3051): Found peer. What's device state of 3051? addr=1896485036, defaddr=0 maxms=2000, lastms=8 Jun 21 17:59:35 DEBUG[25145] devicestate.c: Changing state for IAX2/3051 - state 1 (Not in use) Jun 21 17:59:35 DEBUG[25168] app_queue.c: Device 'IAX2/3051' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Jun 21 17:59:35 DEBUG[25157] iax2-provision.c: Unable to create provisioning packet for 'si-000fd3000a41' Jun 21 17:59:50 VERBOSE[25160] 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=z9hG4bK07szc4jf6ibgHwgv Max-Forwards: 70 User-Agent: SOYO.SIP V1.49.002 CFG0 From: "3050" ;tag=WAxzvQzTfYiYK7En To: "3097" Call-ID: VASi7D6EAXbgM4Gn@172.16.10.124 Contact: CSeq: 1 INVITE Supported: replaces Content-Type: application/sdp Content-Length: 235 v=0 o=- 51848067 93470807 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 Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 0: INVITE sip:3097@pfdesenv.planetarium.com.br SIP/2.0 (51) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bK07szc4jf6ibgHwgv (66) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 2: Max-Forwards: 70 (16) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 3: User-Agent: SOYO.SIP V1.49.002 CFG0 (36) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 4: From: "3050" ;tag=WAxzvQzTfYiYK7En (72) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 5: To: "3097" (49) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 6: Call-ID: VASi7D6EAXbgM4Gn@172.16.10.124 (39) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 7: Contact: (38) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 8: CSeq: 1 INVITE (14) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 9: Supported: replaces (19) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 10: Content-Type: application/sdp (29) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 11: Content-Length: 235 (19) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 12: (0) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: v=0 (3) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: o=- 51848067 93470807 IN IP4 172.16.10.124 (42) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: s=SIP CALL (10) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: c=IN IP4 172.16.10.124 (22) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: t=0 0 (5) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: m=audio 1722 RTP/AVP 18 4 0 8 3 (31) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: a=rtpmap:4 G723/8000 (20) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) Jun 21 17:59:50 VERBOSE[25160] logger.c: --- (12 headers 11 lines)Jun 21 17:59:50 VERBOSE[25160] logger.c: --- (12 headers 11 lines)--- Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Allocating new SIP dialog for VASi7D6EAXbgM4Gn@172.16.10.124 - INVITE (With RTP) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Begin: parsing SIP "Supported: replaces" Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Found SIP option: -replaces- Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Matched SIP option: replaces Jun 21 17:59:50 DEBUG[25160] chan_sip.c: * SIP extension value: 1 for call VASi7D6EAXbgM4Gn@172.16.10.124 Jun 21 17:59:50 VERBOSE[25160] logger.c: Using INVITE request as basis request - VASi7D6EAXbgM4Gn@172.16.10.124 Jun 21 17:59:50 VERBOSE[25160] logger.c: Sending to 172.16.10.124 : 5060 (non-NAT) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Setting NAT on RTP to 524288 Jun 21 17:59:50 VERBOSE[25160] 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=z9hG4bK07szc4jf6ibgHwgv;received=172.16.10.124 From: "3050" ;tag=WAxzvQzTfYiYK7En To: "3097" ;tag=as0e11bdcd Call-ID: VASi7D6EAXbgM4Gn@172.16.10.124 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="4c27f730" Content-Length: 0 --- Jun 21 17:59:50 DEBUG[25160] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #23 Jun 21 17:59:50 VERBOSE[25160] logger.c: Scheduling destruction of call 'VASi7D6EAXbgM4Gn@172.16.10.124' in 15000 ms Jun 21 17:59:50 VERBOSE[25160] logger.c: Found user '3050' Jun 21 17:59:50 VERBOSE[25160] 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=z9hG4bK07szc4jf6ibgHwgv Max-Forwards: 70 User-Agent: SOYO.SIP V1.49.002 CFG0 From: "3050" ;tag=WAxzvQzTfYiYK7En To: "3097" ;tag=as0e11bdcd Call-ID: VASi7D6EAXbgM4Gn@172.16.10.124 Contact: CSeq: 1 ACK Content-Length: 0 Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 0: ACK sip:3097@pfdesenv.planetarium.com.br SIP/2.0 (48) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bK07szc4jf6ibgHwgv (66) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 2: Max-Forwards: 70 (16) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 3: User-Agent: SOYO.SIP V1.49.002 CFG0 (36) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 4: From: "3050" ;tag=WAxzvQzTfYiYK7En (72) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 5: To: "3097" ;tag=as0e11bdcd (64) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 6: Call-ID: VASi7D6EAXbgM4Gn@172.16.10.124 (39) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 7: Contact: (38) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 8: CSeq: 1 ACK (11) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 9: Content-Length: 0 (17) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 10: (0) Jun 21 17:59:50 VERBOSE[25160] logger.c: --- (10 headers 0 lines)Jun 21 17:59:50 VERBOSE[25160] logger.c: --- (10 headers 0 lines)--- Jun 21 17:59:50 DEBUG[25160] chan_sip.c: = Found Their Call ID: VASi7D6EAXbgM4Gn@172.16.10.124 Their Tag WAxzvQzTfYiYK7En Our tag: as0e11bdcd Jun 21 17:59:50 DEBUG[25160] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Jun 21 17:59:50 DEBUG[25160] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #23 Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Stopping retransmission on 'VASi7D6EAXbgM4Gn@172.16.10.124' of Response 1: Match Found Jun 21 17:59:50 VERBOSE[25160] 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=z9hG4bK635FXxWKxuqGnRMd Max-Forwards: 70 User-Agent: SOYO.SIP V1.49.002 CFG0 From: "3050" ;tag=WAxzvQzTfYiYK7En To: "3097" Call-ID: VASi7D6EAXbgM4Gn@172.16.10.124 Contact: Proxy-Authorization: Digest username="3050", realm="asterisk", nonce="4c27f730", uri="sip:3097@pfdesenv.planetarium.com.br", response="930eda3854ee777770a18139cd6538be", algorithm=MD5 CSeq: 2 INVITE Supported: replaces Content-Type: application/sdp Content-Length: 235 v=0 o=- 83479301 17448528 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 Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 0: INVITE sip:3097@pfdesenv.planetarium.com.br SIP/2.0 (51) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bK635FXxWKxuqGnRMd (66) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 2: Max-Forwards: 70 (16) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 3: User-Agent: SOYO.SIP V1.49.002 CFG0 (36) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 4: From: "3050" ;tag=WAxzvQzTfYiYK7En (72) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 5: To: "3097" (49) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 6: Call-ID: VASi7D6EAXbgM4Gn@172.16.10.124 (39) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 7: Contact: (38) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 8: Proxy-Authorization: Digest username="3050", realm="asterisk", nonce="4c27f730", uri="sip:3097@pfdesenv.planetarium.com.br", response="930eda3854ee777770a18139cd6538be", algorithm=MD5 (183) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 9: CSeq: 2 INVITE (14) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 10: Supported: replaces (19) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 11: Content-Type: application/sdp (29) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 12: Content-Length: 235 (19) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Header 13: (0) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: v=0 (3) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: o=- 83479301 17448528 IN IP4 172.16.10.124 (42) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: s=SIP CALL (10) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: c=IN IP4 172.16.10.124 (22) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: t=0 0 (5) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: m=audio 1722 RTP/AVP 18 4 0 8 3 (31) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: a=rtpmap:4 G723/8000 (20) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) Jun 21 17:59:50 VERBOSE[25160] logger.c: --- (13 headers 11 lines)Jun 21 17:59:50 VERBOSE[25160] logger.c: --- (13 headers 11 lines)--- Jun 21 17:59:50 DEBUG[25160] chan_sip.c: = Found Their Call ID: VASi7D6EAXbgM4Gn@172.16.10.124 Their Tag WAxzvQzTfYiYK7En Our tag: as0e11bdcd Jun 21 17:59:50 DEBUG[25160] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Jun 21 17:59:50 VERBOSE[25160] logger.c: Using INVITE request as basis request - VASi7D6EAXbgM4Gn@172.16.10.124 Jun 21 17:59:50 VERBOSE[25160] logger.c: Sending to 172.16.10.124 : 5060 (NAT) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Setting NAT on RTP to 524288 Jun 21 17:59:50 VERBOSE[25160] logger.c: Found user '3050' Jun 21 17:59:50 VERBOSE[25160] logger.c: Found RTP audio format 18 Jun 21 17:59:50 VERBOSE[25160] logger.c: Found RTP audio format 4 Jun 21 17:59:50 VERBOSE[25160] logger.c: Found RTP audio format 0 Jun 21 17:59:50 VERBOSE[25160] logger.c: Found RTP audio format 8 Jun 21 17:59:50 VERBOSE[25160] logger.c: Found RTP audio format 3 Jun 21 17:59:50 VERBOSE[25160] logger.c: Peer audio RTP is at port 172.16.10.124:1722 Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Peer audio RTP is at port 172.16.10.124:1722 Jun 21 17:59:50 VERBOSE[25160] logger.c: Found description format G729 Jun 21 17:59:50 VERBOSE[25160] logger.c: Found description format G723 Jun 21 17:59:50 VERBOSE[25160] logger.c: Found description format PCMU Jun 21 17:59:50 VERBOSE[25160] logger.c: Found description format PCMA Jun 21 17:59:50 VERBOSE[25160] logger.c: Found description format GSM Jun 21 17:59:50 VERBOSE[25160] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x8 (alaw) Jun 21 17:59:50 VERBOSE[25160] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Checking SIP call limits for device 3050 Jun 21 17:59:50 DEBUG[25160] chan_sip.c: Updating call counter for incoming call Jun 21 17:59:50 VERBOSE[25160] logger.c: Looking for 3097 in 3050_aux (domain pfdesenv.planetarium.com.br) Jun 21 17:59:50 DEBUG[25160] chan_sip.c: build_route: Contact hop: Jun 21 17:59:50 VERBOSE[25160] logger.c: list_route: hop: Jun 21 17:59:50 VERBOSE[25160] 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=z9hG4bK635FXxWKxuqGnRMd;received=172.16.10.124 From: "3050" ;tag=WAxzvQzTfYiYK7En To: "3097" Call-ID: VASi7D6EAXbgM4Gn@172.16.10.124 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jun 21 17:59:50 DEBUG[25145] chan_sip.c: Checking device state for peer 3050 Jun 21 17:59:50 DEBUG[25145] devicestate.c: Changing state for SIP/3050 - state 2 (In use) Jun 21 17:59:50 DEBUG[25173] pbx.c: Expression result is '0' Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'GotoIf' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing GotoIf("SIP/3050-312e", "0 ? 1000") in new stack Jun 21 17:59:50 DEBUG[25173] pbx.c: Not taking any branch Jun 21 17:59:50 DEBUG[25173] pbx.c: Expression result is '1' Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'GotoIf' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing GotoIf("SIP/3050-312e", "1 ? 200:400") in new stack Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Goto (3050_aux,3097,200) Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'DBget' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing DBget("SIP/3050-312e", "ramalbloqueado=BLOQUEIORAMAL/3050") in new stack Jun 21 17:59:50 WARNING[25173] app_db.c: This application has been deprecated, please use the ${DB(family/key)} function instead. Jun 21 17:59:50 VERBOSE[25173] logger.c: -- DBget: varname=ramalbloqueado, family=BLOQUEIORAMAL, key=3050 Jun 21 17:59:50 DEBUG[25173] db.c: Unable to find key '3050' in family 'BLOQUEIORAMAL' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- DBget: Value not found in database. Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'Goto' Jun 21 17:59:50 DEBUG[25174] app_queue.c: Device 'SIP/3050' changed to state '2' (In use) Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing Goto("SIP/3050-312e", "400") in new stack Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Goto (3050_aux,3097,400) Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'SetVar' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing SetVar("SIP/3050-312e", "NUMERODISCADO=3097") in new stack Jun 21 17:59:50 WARNING[25173] pbx.c: SetVar is deprecated, please use Set instead. Jun 21 17:59:50 DEBUG[25173] pbx.c: Expression result is '0' Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'GotoIf' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing GotoIf("SIP/3050-312e", "0 ? 404") in new stack Jun 21 17:59:50 DEBUG[25173] pbx.c: Not taking any branch Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'SetVar' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing SetVar("SIP/3050-312e", "TRANSFER_CONTEXT=3050_aux") in new stack Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'Set' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing Set("SIP/3050-312e", "GROUP=3050") in new stack Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'SetVar' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing SetVar("SIP/3050-312e", "LIMIT_WARNING_FILE=beep") in new stack Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'SetVar' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing SetVar("SIP/3050-312e", "LIMIT_TIMEOUT_FILE=beep5") in new stack Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'Goto' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing Goto("SIP/3050-312e", "500") in new stack Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Goto (3050_aux,3097,500) Jun 21 17:59:50 DEBUG[25173] pbx.c: Expression result is '0' Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'GotoIf' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing GotoIf("SIP/3050-312e", "0 ? 700") in new stack Jun 21 17:59:50 DEBUG[25173] pbx.c: Not taking any branch Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'DBget' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing DBget("SIP/3050-312e", "GRAVACAOCHAMADASAIDA=GRAVACAOCHAMADASAIDA/3050") in new stack Jun 21 17:59:50 VERBOSE[25173] logger.c: -- DBget: varname=GRAVACAOCHAMADASAIDA, family=GRAVACAOCHAMADASAIDA, key=3050 Jun 21 17:59:50 DEBUG[25173] db.c: Unable to find key '3050' in family 'GRAVACAOCHAMADASAIDA' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- DBget: Value not found in database. Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'Goto' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing Goto("SIP/3050-312e", "700") in new stack Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Goto (3050_aux,3097,700) Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'Goto' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing Goto("SIP/3050-312e", "3050_out|3097|1") in new stack Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Goto (3050_out,3097,1) Jun 21 17:59:50 DEBUG[25173] pbx.c: Expression result is '1' Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'GotoIf' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing GotoIf("SIP/3050-312e", "1 ? 2:5") in new stack Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Goto (3050_out,3097,2) Jun 21 17:59:50 DEBUG[25173] pbx.c: Expression result is '1' Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'GotoIf' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing GotoIf("SIP/3050-312e", "1 ? 3:5") in new stack Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Goto (3050_out,3097,3) Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'SetVar' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing SetVar("SIP/3050-312e", "CHAMADAINTERNA=T") in new stack Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'SetCIDNum' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing SetCIDNum("SIP/3050-312e", "3050") in new stack Jun 21 17:59:50 WARNING[25173] app_setcidnum.c: SetCIDNum is deprecated, please use Set(CALLERID(number)=value) instead. Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'Goto' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing Goto("SIP/3050-312e", "disca|3097|1") in new stack Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Goto (disca,3097,1) Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'Goto' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing Goto("SIP/3050-312e", "fila_desenvolvimento|3097|1") in new stack Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Goto (fila_desenvolvimento,3097,1) Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'SetLanguage' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing SetLanguage("SIP/3050-312e", "br") in new stack Jun 21 17:59:50 WARNING[25173] pbx.c: SetLanguage is deprecated, please use Set(LANGUAGE()=language) instead. Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'Goto' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing Goto("SIP/3050-312e", "3") in new stack Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Goto (fila_desenvolvimento,3097,3) Jun 21 17:59:50 DEBUG[25173] pbx.c: Launching 'Wait' Jun 21 17:59:50 VERBOSE[25173] logger.c: -- Executing Wait("SIP/3050-312e", "1") in new stack Jun 21 17:59:51 DEBUG[25173] pbx.c: Launching 'SetCIDName' Jun 21 17:59:51 VERBOSE[25173] logger.c: -- Executing SetCIDName("SIP/3050-312e", "Fila:desenvolvimento:Ramal Teste") in new stack Jun 21 17:59:51 WARNING[25173] app_setcidname.c: SetCIDName is deprecated, please use Set(CALLERID(name)=value) instead. Jun 21 17:59:51 DEBUG[25173] pbx.c: Launching 'SetVar' Jun 21 17:59:51 VERBOSE[25173] logger.c: -- Executing SetVar("SIP/3050-312e", "NOMEFILA=desenvolvimento") in new stack Jun 21 17:59:51 DEBUG[25173] pbx.c: Launching 'SetVar' Jun 21 17:59:51 VERBOSE[25173] logger.c: -- Executing SetVar("SIP/3050-312e", "MONITOR_FILENAME=1150923590.0-filadesenvolvimento-3097-in-20060621-175951-3050") in new stack Jun 21 17:59:51 DEBUG[25173] pbx.c: Launching 'Goto' Jun 21 17:59:51 VERBOSE[25173] logger.c: -- Executing Goto("SIP/3050-312e", "8") in new stack Jun 21 17:59:51 VERBOSE[25173] logger.c: -- Goto (fila_desenvolvimento,3097,8) Jun 21 17:59:51 DEBUG[25173] pbx.c: Launching 'SetVar' Jun 21 17:59:51 VERBOSE[25173] logger.c: -- Executing SetVar("SIP/3050-312e", "DATAENTRADAFILA=1150923591") in new stack Jun 21 17:59:51 DEBUG[25173] pbx.c: Launching 'SetVar' Jun 21 17:59:51 VERBOSE[25173] logger.c: -- Executing SetVar("SIP/3050-312e", "NUMEROCLIENTE=666") in new stack Jun 21 17:59:51 DEBUG[25173] pbx.c: Launching 'SetVar' Jun 21 17:59:51 VERBOSE[25173] logger.c: -- Executing SetVar("SIP/3050-312e", "IDTSOLICITANTE=C") in new stack Jun 21 17:59:51 DEBUG[25173] pbx.c: Launching 'SetVar' Jun 21 17:59:51 VERBOSE[25173] logger.c: -- Executing SetVar("SIP/3050-312e", "NUMEROTELEFCHAMADO=3097") in new stack Jun 21 17:59:51 DEBUG[25173] pbx.c: Launching 'Queue' Jun 21 17:59:51 VERBOSE[25173] logger.c: -- Executing Queue("SIP/3050-312e", "desenvolvimento|tr|||600") in new stack Jun 21 17:59:51 DEBUG[25173] app_queue.c: NO QUEUE_PRIO variable found. Using default. Jun 21 17:59:51 DEBUG[25173] app_queue.c: queue: desenvolvimento, options: tr, url: , announce: , expires: 1150924191, priority: 0 Jun 21 17:59:51 DEBUG[25173] app_queue.c: Queue 'desenvolvimento' Join, Channel 'SIP/3050-312e', Position '1' Jun 21 17:59:51 VERBOSE[25173] 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=z9hG4bK635FXxWKxuqGnRMd;received=172.16.10.124 From: "3050" ;tag=WAxzvQzTfYiYK7En To: "3097" ;tag=as7e8ef3b7 Call-ID: VASi7D6EAXbgM4Gn@172.16.10.124 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jun 21 17:59:51 DEBUG[25173] app_queue.c: It's our turn (SIP/3050-312e). Jun 21 17:59:51 DEBUG[25173] app_queue.c: SIP/3050-312e is trying to call a queue member. Jun 21 17:59:51 DEBUG[25173] app_queue.c: Queue with URL=_ Jun 21 17:59:51 DEBUG[25173] app_queue.c: Queue with URL=_ Jun 21 17:59:51 DEBUG[25173] app_queue.c: Queue with URL=_ Jun 21 17:59:51 DEBUG[25173] app_queue.c: Queue with URL=_ Jun 21 17:59:51 DEBUG[25173] app_queue.c: Queue with URL=_ Jun 21 17:59:51 DEBUG[25173] app_queue.c: Queue with URL=_ Jun 21 17:59:51 DEBUG[25173] app_queue.c: Trying 'Agent/8003' with metric 0 Jun 21 17:59:51 DEBUG[25173] app_queue.c: Trying 'Agent/3051' with metric 0 Jun 21 17:59:51 DEBUG[25173] app_queue.c: Trying 'Agent/3053' with metric 0 Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-12. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable NUMEROTELEFCHAMADO. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-11. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable IDTSOLICITANTE. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-10. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable NUMEROCLIENTE. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-9. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable DATAENTRADAFILA. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-8. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-7. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable MONITOR_FILENAME. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-6. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable NOMEFILA. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-5. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-4. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-3. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-2. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-1. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-disca-3097-1. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_out-3097-5. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_out-3097-4. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable CHAMADAINTERNA. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_out-3097-3. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_out-3097-2. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_out-3097-1. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_aux-3097-700. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_aux-3097-602. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable DBGETSTATUS. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_aux-3097-501. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_aux-3097-500. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_aux-3097-406. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable LIMIT_TIMEOUT_FILE. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_aux-3097-405. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable LIMIT_WARNING_FILE. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_aux-3097-404. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable GROUP. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_aux-3097-403. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable TRANSFER_CONTEXT. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_aux-3097-402. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_aux-3097-401. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable NUMERODISCADO. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_aux-3097-400. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_aux-3097-301. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_aux-3097-200. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_aux-3097-2. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable STACK-3050_aux-3097-1. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable SIPCALLID. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable SIPUSERAGENT. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable SIPDOMAIN. Jun 21 17:59:51 DEBUG[25173] channel.c: Not copying variable SIPURI. Jun 21 17:59:51 VERBOSE[25173] logger.c: -- outgoing agentcall, to agent '3053', on 'Local/3053@dac_suporte-5ff0,1' Jun 21 17:59:51 VERBOSE[25173] logger.c: -- Called Agent/3053 Jun 21 17:59:51 DEBUG[25145] devicestate.c: Changing state for Local/3053@dac_suporte - state 2 (In use) Jun 21 17:59:51 DEBUG[25175] pbx.c: Launching 'Set' Jun 21 17:59:51 VERBOSE[25175] logger.c: -- Executing Set("Local/3053@dac_suporte-5ff0,2", "GROUP=3053") in new stack Jun 21 17:59:51 DEBUG[25175] pbx.c: Function result is '1' Jun 21 17:59:51 DEBUG[25175] pbx.c: Launching 'NoOp' Jun 21 17:59:51 VERBOSE[25175] logger.c: -- Executing NoOp("Local/3053@dac_suporte-5ff0,2", "Group: 3053 Count: 1") in new stack Jun 21 17:59:51 DEBUG[25175] pbx.c: Function result is '1' Jun 21 17:59:51 DEBUG[25175] pbx.c: Expression result is '0' Jun 21 17:59:51 DEBUG[25175] pbx.c: Launching 'GotoIf' Jun 21 17:59:51 VERBOSE[25175] logger.c: -- Executing GotoIf("Local/3053@dac_suporte-5ff0,2", "0 ? 1000") in new stack Jun 21 17:59:51 DEBUG[25175] pbx.c: Not taking any branch Jun 21 17:59:51 DEBUG[25175] pbx.c: Launching 'Goto' Jun 21 17:59:51 VERBOSE[25175] logger.c: -- Executing Goto("Local/3053@dac_suporte-5ff0,2", "200") in new stack Jun 21 17:59:51 VERBOSE[25175] logger.c: -- Goto (dac_suporte,3053,200) Jun 21 17:59:51 DEBUG[25175] pbx.c: Launching 'SetVar' Jun 21 17:59:51 VERBOSE[25175] logger.c: -- Executing SetVar("Local/3053@dac_suporte-5ff0,2", "CHAMADAINTERNA=T") in new stack Jun 21 17:59:51 DEBUG[25175] pbx.c: Launching 'Dial' Jun 21 17:59:51 VERBOSE[25175] logger.c: -- Executing Dial("Local/3053@dac_suporte-5ff0,2", "SIP/3053|15|T") in new stack Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Setting NAT on RTP to 0 Jun 21 17:59:51 DEBUG[25175] channel.c: Not copying variable STACK-dac_suporte-3053-201. Jun 21 17:59:51 DEBUG[25175] channel.c: Not copying variable CHAMADAINTERNA. Jun 21 17:59:51 DEBUG[25175] channel.c: Not copying variable STACK-dac_suporte-3053-200. Jun 21 17:59:51 DEBUG[25175] channel.c: Not copying variable STACK-dac_suporte-3053-4. Jun 21 17:59:51 DEBUG[25175] channel.c: Not copying variable STACK-dac_suporte-3053-3. Jun 21 17:59:51 DEBUG[25175] channel.c: Not copying variable STACK-dac_suporte-3053-2. Jun 21 17:59:51 DEBUG[25175] channel.c: Not copying variable GROUP. Jun 21 17:59:51 DEBUG[25175] channel.c: Not copying variable STACK-dac_suporte-3053-1. Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Outgoing Call for 3053 Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Updating call counter for outgoing call Jun 21 17:59:51 VERBOSE[25175] logger.c: We're at 200.196.44.45 port 10606 Jun 21 17:59:51 VERBOSE[25175] logger.c: Adding codec 0x8 (alaw) to SDP Jun 21 17:59:51 VERBOSE[25175] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Header 0: INVITE sip:3053@172.16.10.112:5060 SIP/2.0 (42) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK7b7a94b2;rport (64) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Header 2: From: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 (80) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Header 3: To: (33) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Header 4: Contact: (33) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Header 5: Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 (55) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Header 6: CSeq: 102 INVITE (16) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Header 8: Max-Forwards: 70 (16) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Header 9: Date: Wed, 21 Jun 2006 20:59:51 GMT (35) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Header 11: Content-Type: application/sdp (29) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Header 12: Content-Length: 218 (19) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Header 13: (0) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Line: v=0 (3) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Line: o=root 25128 25128 IN IP4 200.196.44.45 (39) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Line: s=session (9) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Line: c=IN IP4 200.196.44.45 (22) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Line: t=0 0 (5) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Line: m=audio 10606 RTP/AVP 8 101 (27) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Line: a=fmtp:101 0-16 (15) Jun 21 17:59:51 DEBUG[25175] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Jun 21 17:59:51 VERBOSE[25175] logger.c: 13 headers, 10 lines Jun 21 17:59:51 VERBOSE[25175] logger.c: Reliably Transmitting (no NAT) to 172.16.10.112:5060: INVITE sip:3053@172.16.10.112:5060 SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK7b7a94b2;rport From: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 To: Contact: Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Jun 2006 20:59:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 218 v=0 o=root 25128 25128 IN IP4 200.196.44.45 s=session c=IN IP4 200.196.44.45 t=0 0 m=audio 10606 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jun 21 17:59:51 DEBUG[25175] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #25 Jun 21 17:59:51 VERBOSE[25175] logger.c: -- Called 3053 Jun 21 17:59:51 DEBUG[25175] channel.c: Set channel SIP/3053-7a88 to read format alaw Jun 21 17:59:51 DEBUG[25175] channel.c: Set channel Local/3053@dac_suporte-5ff0,2 to write format alaw Jun 21 17:59:51 DEBUG[25175] channel.c: Set channel Local/3053@dac_suporte-5ff0,2 to read format alaw Jun 21 17:59:51 DEBUG[25175] channel.c: Set channel SIP/3053-7a88 to write format alaw Jun 21 17:59:51 DEBUG[25176] app_queue.c: Device 'Local/3053@dac_suporte' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jun 21 17:59:51 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.112:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK7b7a94b2;rport From: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 To: ;tag=590337CD-1BADB706 CSeq: 102 INVITE Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Contact: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 Content-Length: 0 Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK7b7a94b2;rport (64) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 2: From: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 (80) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 3: To: ;tag=590337CD-1BADB706 (55) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 4: CSeq: 102 INVITE (16) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 5: Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 (55) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 6: Contact: (38) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 7: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 (54) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 8: Content-Length: 0 (17) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 9: (0) Jun 21 17:59:51 VERBOSE[25160] logger.c: --- (9 headers 0 lines)Jun 21 17:59:51 VERBOSE[25160] logger.c: --- (9 headers 0 lines)--- Jun 21 17:59:51 DEBUG[25160] chan_sip.c: = Found Their Call ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Their Tag Our tag: as12a46264 Jun 21 17:59:51 DEBUG[25160] chan_sip.c: *** SIP TIMER: Cancelling retransmission #25 - INVITE (got response) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '119ce4a9105ce0a33433067e7132ca34@200.196.44.45' Request 102: Found Jun 21 17:59:51 DEBUG[25160] chan_sip.c: SIP response 100 to standard invite Jun 21 17:59:51 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.112:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK7b7a94b2;rport From: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 To: ;tag=590337CD-1BADB706 CSeq: 102 INVITE Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Contact: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 Allow-Events: talk,hold,conference Content-Length: 0 Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK7b7a94b2;rport (64) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 2: From: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 (80) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 3: To: ;tag=590337CD-1BADB706 (55) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 4: CSeq: 102 INVITE (16) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 5: Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 (55) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 6: Contact: (38) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 7: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 (54) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 8: Allow-Events: talk,hold,conference (34) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 9: Content-Length: 0 (17) Jun 21 17:59:51 DEBUG[25160] chan_sip.c: Header 10: (0) Jun 21 17:59:51 VERBOSE[25160] logger.c: --- (10 headers 0 lines)Jun 21 17:59:51 VERBOSE[25160] logger.c: --- (10 headers 0 lines)--- Jun 21 17:59:51 DEBUG[25160] chan_sip.c: = Found Their Call ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Their Tag 590337CD-1BADB706 Our tag: as12a46264 Jun 21 17:59:51 DEBUG[25160] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '119ce4a9105ce0a33433067e7132ca34@200.196.44.45' Request 102: Found Jun 21 17:59:51 DEBUG[25160] chan_sip.c: SIP response 180 to standard invite Jun 21 17:59:51 VERBOSE[25175] logger.c: -- SIP/3053-7a88 is ringing Jun 21 17:59:51 DEBUG[25145] chan_sip.c: Checking device state for peer 3053 Jun 21 17:59:51 DEBUG[25145] devicestate.c: Changing state for SIP/3053 - state 6 (Ringing) Jun 21 17:59:51 VERBOSE[25173] logger.c: -- Agent/3053 is ringing Jun 21 17:59:51 DEBUG[25177] 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. Jun 21 17:59:53 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.112:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK7b7a94b2;rport From: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 To: ;tag=590337CD-1BADB706 CSeq: 102 INVITE Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 Content-Type: application/sdp Content-Length: 187 v=0 o=- 979455113 979455113 IN IP4 172.16.10.112 s=Polycom IP Phone c=IN IP4 172.16.10.112 t=0 0 m=audio 2224 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK7b7a94b2;rport (64) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 2: From: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 (80) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 3: To: ;tag=590337CD-1BADB706 (55) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 4: CSeq: 102 INVITE (16) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 5: Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 (55) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 6: Contact: (38) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 (54) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 9: Content-Type: application/sdp (29) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 10: Content-Length: 187 (19) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 11: (0) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Line: v=0 (3) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Line: o=- 979455113 979455113 IN IP4 172.16.10.112 (44) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Line: s=Polycom IP Phone (18) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Line: c=IN IP4 172.16.10.112 (22) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Line: t=0 0 (5) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Line: m=audio 2224 RTP/AVP 8 101 (26) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Jun 21 17:59:53 VERBOSE[25160] logger.c: --- (11 headers 8 lines)Jun 21 17:59:53 VERBOSE[25160] logger.c: --- (11 headers 8 lines)--- Jun 21 17:59:53 DEBUG[25160] chan_sip.c: = Found Their Call ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Their Tag 590337CD-1BADB706 Our tag: as12a46264 Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Acked pending invite 102 Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Stopping retransmission on '119ce4a9105ce0a33433067e7132ca34@200.196.44.45' of Request 102: Match Found Jun 21 17:59:53 DEBUG[25160] chan_sip.c: SIP response 200 to standard invite Jun 21 17:59:53 VERBOSE[25160] logger.c: Found RTP audio format 8 Jun 21 17:59:53 VERBOSE[25160] logger.c: Found RTP audio format 101 Jun 21 17:59:53 VERBOSE[25160] logger.c: Peer audio RTP is at port 172.16.10.112:2224 Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Peer audio RTP is at port 172.16.10.112:2224 Jun 21 17:59:53 VERBOSE[25160] logger.c: Found description format PCMA Jun 21 17:59:53 VERBOSE[25160] logger.c: Found description format telephone-event Jun 21 17:59:53 VERBOSE[25160] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Jun 21 17:59:53 VERBOSE[25160] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: build_route: Contact hop: Jun 21 17:59:53 VERBOSE[25160] logger.c: list_route: hop: Jun 21 17:59:53 VERBOSE[25160] logger.c: set_destination: Parsing for address/port to send to Jun 21 17:59:53 VERBOSE[25160] logger.c: set_destination: set destination to 172.16.10.112, port 5060 Jun 21 17:59:53 VERBOSE[25160] logger.c: Transmitting (no NAT) to 172.16.10.112:5060: ACK sip:3053@172.16.10.112:5060 SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK55d3c813;rport From: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 To: ;tag=590337CD-1BADB706 Contact: Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jun 21 17:59:53 VERBOSE[25175] logger.c: -- SIP/3053-7a88 answered Local/3053@dac_suporte-5ff0,2 Jun 21 17:59:53 DEBUG[25175] channel.c: Set channel Local/3053@dac_suporte-5ff0,2 to read format alaw Jun 21 17:59:53 DEBUG[25175] channel.c: Set channel SIP/3053-7a88 to write format alaw Jun 21 17:59:53 DEBUG[25175] channel.c: Set channel SIP/3053-7a88 to read format alaw Jun 21 17:59:53 DEBUG[25175] channel.c: Set channel Local/3053@dac_suporte-5ff0,2 to write format alaw Jun 21 17:59:53 DEBUG[25145] chan_sip.c: Checking device state for peer 3053 Jun 21 17:59:53 DEBUG[25145] devicestate.c: Changing state for SIP/3053 - state 2 (In use) Jun 21 17:59:53 DEBUG[25145] devicestate.c: Changing state for Local/3053@dac_suporte - state 2 (In use) Jun 21 17:59:53 DEBUG[25173] app_queue.c: Dunno what to do with control type -1 Jun 21 17:59:53 VERBOSE[25173] logger.c: -- Agent/3053 answered SIP/3050-312e Jun 21 17:59:53 DEBUG[25173] channel.c: Set channel SIP/3050-312e to read format alaw Jun 21 17:59:53 DEBUG[25173] channel.c: Set channel Agent/3053 to write format alaw Jun 21 17:59:53 DEBUG[25173] channel.c: Set channel Agent/3053 to read format alaw Jun 21 17:59:53 DEBUG[25173] channel.c: Set channel SIP/3050-312e to write format alaw Jun 21 17:59:53 DEBUG[25173] chan_sip.c: sip_answer(SIP/3050-312e) Jun 21 17:59:53 VERBOSE[25173] logger.c: We're at 200.196.44.45 port 11576 Jun 21 17:59:53 VERBOSE[25173] logger.c: Adding codec 0x8 (alaw) to SDP Jun 21 17:59:53 VERBOSE[25173] 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=z9hG4bK635FXxWKxuqGnRMd;received=172.16.10.124 From: "3050" ;tag=WAxzvQzTfYiYK7En To: "3097" ;tag=as7e8ef3b7 Call-ID: VASi7D6EAXbgM4Gn@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 25128 25128 IN IP4 200.196.44.45 s=session c=IN IP4 200.196.44.45 t=0 0 m=audio 11576 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- Jun 21 17:59:53 DEBUG[25173] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #26 Jun 21 17:59:53 DEBUG[25178] 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. Jun 21 17:59:53 DEBUG[25179] app_queue.c: Device 'Local/3053@dac_suporte' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jun 21 17:59:53 DEBUG[25145] devicestate.c: Changing state for Local/3053@dac_suporte - state 2 (In use) Jun 21 17:59:53 DEBUG[25145] devicestate.c: Changing state for Agent/3053 - state 3 (Busy) Jun 21 17:59:53 DEBUG[25145] chan_sip.c: Checking device state for peer 3050 Jun 21 17:59:53 DEBUG[25145] devicestate.c: Changing state for SIP/3050 - state 2 (In use) Jun 21 17:59:53 DEBUG[25180] app_queue.c: Device 'Local/3053@dac_suporte' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jun 21 17:59:53 DEBUG[25181] app_queue.c: Device 'Agent/3053' changed to state '3' (Busy) Jun 21 17:59:53 DEBUG[25182] app_queue.c: Device 'SIP/3050' changed to state '2' (In use) Jun 21 17:59:53 DEBUG[25175] channel.c: Planning to masquerade channel SIP/3053-7a88 into the structure of Local/3053@dac_suporte-5ff0,1 Jun 21 17:59:53 DEBUG[25175] channel.c: Done planning to masquerade channel SIP/3053-7a88 into the structure of Local/3053@dac_suporte-5ff0,1 Jun 21 17:59:53 DEBUG[25175] chan_local.c: Not posting to queue since already masked on 'Local/3053@dac_suporte-5ff0,2' Jun 21 17:59:53 DEBUG[25173] channel.c: Actually Masquerading SIP/3053-7a88(6) into the structure of Local/3053@dac_suporte-5ff0,1(6) Jun 21 17:59:53 DEBUG[25173] channel.c: Got clone lock for masquerade on 'SIP/3053-7a88' at 0x8e9d75c Jun 21 17:59:53 DEBUG[25173] channel.c: Set channel SIP/3053-7a88 to write format alaw Jun 21 17:59:53 DEBUG[25173] channel.c: Set channel SIP/3053-7a88 to read format alaw Jun 21 17:59:53 DEBUG[25173] channel.c: Putting channel SIP/3053-7a88 in 8/8 formats Jun 21 17:59:53 DEBUG[25173] channel.c: Released clone lock on 'Local/3053@dac_suporte-5ff0,1' Jun 21 17:59:53 DEBUG[25173] channel.c: Done Masquerading SIP/3053-7a88 (6) Jun 21 17:59:53 DEBUG[25173] chan_agent.c: Bridge on 'SIP/3053-7a88' being set to 'Agent/3053' (3) Jun 21 17:59:53 DEBUG[25175] channel.c: Didn't get a frame from channel: Local/3053@dac_suporte-5ff0,1 Jun 21 17:59:53 DEBUG[25175] channel.c: Bridge stops bridging channels Local/3053@dac_suporte-5ff0,2 and Local/3053@dac_suporte-5ff0,1 Jun 21 17:59:53 DEBUG[25175] channel.c: Hanging up zombie 'Local/3053@dac_suporte-5ff0,1' Jun 21 17:59:53 DEBUG[25175] app_dial.c: Exiting with DIALSTATUS=ANSWER. Jun 21 17:59:53 DEBUG[25175] pbx.c: Spawn extension (dac_suporte,3053,201) exited non-zero on 'Local/3053@dac_suporte-5ff0,2' Jun 21 17:59:53 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3053@dac_suporte-5ff0,2' Jun 21 17:59:53 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3053@dac_suporte-5ff0,2' Jun 21 17:59:53 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3053@dac_suporte-5ff0,2' Jun 21 17:59:53 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3053@dac_suporte-5ff0,2' Jun 21 17:59:53 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3053@dac_suporte-5ff0,2' Jun 21 17:59:53 DEBUG[25173] rtp.c: Ooh, format changed from unknown to alaw Jun 21 17:59:53 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3053@dac_suporte-5ff0,2' Jun 21 17:59:53 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3053@dac_suporte-5ff0,2' Jun 21 17:59:53 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3053@dac_suporte-5ff0,2' Jun 21 17:59:53 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3053@dac_suporte-5ff0,2' Jun 21 17:59:53 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3053@dac_suporte-5ff0,2' Jun 21 17:59:53 WARNING[25145] channel.c: Avoided initial deadlock for '0x8e96d08', 10 retries! Jun 21 17:59:53 DEBUG[25145] devicestate.c: Changing state for Local/3053@dac_suporte - state 0 (Unknown) Jun 21 17:59:53 DEBUG[25183] app_queue.c: Device 'Local/3053@dac_suporte' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. Jun 21 17:59:53 VERBOSE[25160] 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=z9hG4bKW2DmliTcnOmLgPBO Max-Forwards: 70 User-Agent: SOYO.SIP V1.49.002 CFG0 From: "3050" ;tag=WAxzvQzTfYiYK7En To: "3097" ;tag=as7e8ef3b7 Call-ID: VASi7D6EAXbgM4Gn@172.16.10.124 Contact: Proxy-Authorization: Digest username="3050", realm="asterisk", nonce="4c27f730", uri="sip:3097@200.196.44.45", response="607540311ff02036746070ef0103b98f", algorithm=MD5 CSeq: 2 ACK Content-Length: 0 Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 0: ACK sip:3097@200.196.44.45 SIP/2.0 (34) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKW2DmliTcnOmLgPBO (66) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 2: Max-Forwards: 70 (16) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 3: User-Agent: SOYO.SIP V1.49.002 CFG0 (36) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 4: From: "3050" ;tag=WAxzvQzTfYiYK7En (72) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 5: To: "3097" ;tag=as7e8ef3b7 (64) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 6: Call-ID: VASi7D6EAXbgM4Gn@172.16.10.124 (39) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 7: Contact: (38) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 8: Proxy-Authorization: Digest username="3050", realm="asterisk", nonce="4c27f730", uri="sip:3097@200.196.44.45", response="607540311ff02036746070ef0103b98f", algorithm=MD5 (169) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 9: CSeq: 2 ACK (11) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 10: Content-Length: 0 (17) Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Header 11: (0) Jun 21 17:59:53 VERBOSE[25160] logger.c: --- (11 headers 0 lines)Jun 21 17:59:53 VERBOSE[25160] logger.c: --- (11 headers 0 lines)--- Jun 21 17:59:53 DEBUG[25160] chan_sip.c: = No match Their Call ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Their Tag 590337CD-1BADB706 Our tag: as12a46264 Jun 21 17:59:53 DEBUG[25160] chan_sip.c: = Found Their Call ID: VASi7D6EAXbgM4Gn@172.16.10.124 Their Tag WAxzvQzTfYiYK7En Our tag: as7e8ef3b7 Jun 21 17:59:53 DEBUG[25160] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Jun 21 17:59:53 DEBUG[25160] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #26 Jun 21 17:59:53 DEBUG[25160] chan_sip.c: Stopping retransmission on 'VASi7D6EAXbgM4Gn@172.16.10.124' of Response 2: Match Found Jun 21 17:59:53 DEBUG[25175] channel.c: Hanging up channel 'Local/3053@dac_suporte-5ff0,2' Jun 21 17:59:53 DEBUG[25145] devicestate.c: Changing state for Local/3053@dac_suporte - state 0 (Unknown) Jun 21 17:59:53 DEBUG[25184] app_queue.c: Device 'Local/3053@dac_suporte' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. Jun 21 17:59:53 DEBUG[25173] rtp.c: Ooh, format changed from unknown to alaw Jun 21 17:59:54 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.112:5060: INVITE sip:3050@200.196.44.45 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bKa81cf6f8E2ECB4F7 From: ;tag=590337CD-1BADB706 To: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 CSeq: 1 INVITE Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 181 v=0 o=- 979455113 979455114 IN IP4 172.16.10.112 s=Polycom IP Phone c=IN IP4 0.0.0.0 t=0 0 m=audio 2224 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 0: INVITE sip:3050@200.196.44.45 SIP/2.0 (37) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bKa81cf6f8E2ECB4F7 (66) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 2: From: ;tag=590337CD-1BADB706 (57) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 3: To: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 (78) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 4: CSeq: 1 INVITE (14) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 5: Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 (55) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 6: Contact: (38) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 (54) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 9: Supported: 100rel,replaces (26) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 10: Allow-Events: talk,hold,conference (34) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 11: Max-Forwards: 70 (16) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 12: Content-Type: application/sdp (29) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 13: Content-Length: 181 (19) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 14: (0) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Line: v=0 (3) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Line: o=- 979455113 979455114 IN IP4 172.16.10.112 (44) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Line: s=Polycom IP Phone (18) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Line: c=IN IP4 0.0.0.0 (16) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Line: t=0 0 (5) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Line: m=audio 2224 RTP/AVP 8 101 (26) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Jun 21 17:59:54 VERBOSE[25160] logger.c: --- (14 headers 8 lines)Jun 21 17:59:54 VERBOSE[25160] logger.c: --- (14 headers 8 lines)--- Jun 21 17:59:54 DEBUG[25160] chan_sip.c: = Found Their Call ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Their Tag 590337CD-1BADB706 Our tag: as12a46264 Jun 21 17:59:54 DEBUG[25160] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Begin: parsing SIP "Supported: 100rel,replaces" Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Found SIP option: -100rel- Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Matched SIP option: 100rel Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Found SIP option: -replaces- Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Matched SIP option: replaces Jun 21 17:59:54 DEBUG[25160] chan_sip.c: * SIP extension value: 3 for call 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Jun 21 17:59:54 VERBOSE[25160] logger.c: Using INVITE request as basis request - 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Jun 21 17:59:54 VERBOSE[25160] logger.c: Sending to 172.16.10.112 : 5060 (non-NAT) Jun 21 17:59:54 VERBOSE[25160] logger.c: Found RTP audio format 8 Jun 21 17:59:54 VERBOSE[25160] logger.c: Found RTP audio format 101 Jun 21 17:59:54 VERBOSE[25160] logger.c: Peer audio RTP is at port 0.0.0.0:2224 Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Peer audio RTP is at port 0.0.0.0:2224 Jun 21 17:59:54 VERBOSE[25160] logger.c: Found description format PCMA Jun 21 17:59:54 VERBOSE[25160] logger.c: Found description format telephone-event Jun 21 17:59:54 VERBOSE[25160] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Jun 21 17:59:54 VERBOSE[25160] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jun 21 17:59:54 DEBUG[25160] chan_agent.c: Asked for bridged channel on 'SIP/3053-7a88'/'Agent/3053', returning 'SIP/3050-312e' Jun 21 17:59:54 DEBUG[25160] channel.c: Set channel SIP/3050-312e to write format slin Jun 21 17:59:54 VERBOSE[25160] logger.c: -- Started music on hold, class 'planetarium', on channel 'SIP/3050-312e' Jun 21 17:59:54 DEBUG[25160] channel.c: Scheduling timer at 160 sample intervals Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Got a SIP re-invite for call 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Jun 21 17:59:54 VERBOSE[25160] logger.c: We're at 200.196.44.45 port 10606 Jun 21 17:59:54 VERBOSE[25160] logger.c: Adding codec 0x8 (alaw) to SDP Jun 21 17:59:54 VERBOSE[25160] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Jun 21 17:59:54 VERBOSE[25160] logger.c: Reliably Transmitting (no NAT) to 172.16.10.112:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bKa81cf6f8E2ECB4F7;received=172.16.10.112 From: ;tag=590337CD-1BADB706 To: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 CSeq: 1 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 25128 25129 IN IP4 200.196.44.45 s=session c=IN IP4 200.196.44.45 t=0 0 m=audio 10606 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jun 21 17:59:54 DEBUG[25160] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #27 Jun 21 17:59:54 DEBUG[25173] channel.c: Generator got voice, switching to phase locked mode Jun 21 17:59:54 DEBUG[25173] channel.c: Scheduling timer at 0 sample intervals Jun 21 17:59:54 DEBUG[25173] rtp.c: Difference is 952, ms is 139 Jun 21 17:59:54 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.112:5060: ACK sip:3050@200.196.44.45 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK8343487a4077E571 From: ;tag=590337CD-1BADB706 To: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 CSeq: 1 ACK Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 Max-Forwards: 70 Content-Length: 0 Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 0: ACK sip:3050@200.196.44.45 SIP/2.0 (34) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK8343487a4077E571 (66) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 2: From: ;tag=590337CD-1BADB706 (57) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 3: To: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 (78) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 4: CSeq: 1 ACK (11) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 5: Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 (55) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 6: Contact: (38) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 (54) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 9: Max-Forwards: 70 (16) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 10: Content-Length: 0 (17) Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Header 11: (0) Jun 21 17:59:54 VERBOSE[25160] logger.c: --- (11 headers 0 lines)Jun 21 17:59:54 VERBOSE[25160] logger.c: --- (11 headers 0 lines)--- Jun 21 17:59:54 DEBUG[25160] chan_sip.c: = Found Their Call ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Their Tag 590337CD-1BADB706 Our tag: as12a46264 Jun 21 17:59:54 DEBUG[25160] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Jun 21 17:59:54 DEBUG[25160] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #27 Jun 21 17:59:54 DEBUG[25160] chan_sip.c: Stopping retransmission on '119ce4a9105ce0a33433067e7132ca34@200.196.44.45' of Response 1: Match Found Jun 21 17:59:57 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.112:5060: INVITE sip:3054@pfdesenv.planetarium.com.br:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK68f3c80c49CAB3BB From: "Teste IP300" ;tag=274E1422-E9CA97BF To: CSeq: 1 INVITE Call-ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 223 v=0 o=- 979455117 979455117 IN IP4 172.16.10.112 s=Polycom IP Phone c=IN IP4 172.16.10.112 t=0 0 a=sendrecv m=audio 2222 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 0: INVITE sip:3054@pfdesenv.planetarium.com.br:5060;user=phone SIP/2.0 (67) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK68f3c80c49CAB3BB (66) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 2: From: "Teste IP300" ;tag=274E1422-E9CA97BF (80) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 3: To: (53) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 4: CSeq: 1 INVITE (14) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 5: Call-ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 (49) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 6: Contact: (38) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 (54) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 9: Supported: 100rel,replaces (26) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 10: Allow-Events: talk,hold,conference (34) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 11: Max-Forwards: 70 (16) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 12: Content-Type: application/sdp (29) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 13: Content-Length: 223 (19) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 14: (0) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: v=0 (3) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: o=- 979455117 979455117 IN IP4 172.16.10.112 (44) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: s=Polycom IP Phone (18) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: c=IN IP4 172.16.10.112 (22) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: t=0 0 (5) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: a=sendrecv (10) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: m=audio 2222 RTP/AVP 0 8 101 (28) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Jun 21 17:59:57 VERBOSE[25160] logger.c: --- (14 headers 10 lines)Jun 21 17:59:57 VERBOSE[25160] logger.c: --- (14 headers 10 lines)--- Jun 21 17:59:57 DEBUG[25160] chan_sip.c: = No match Their Call ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Their Tag 590337CD-1BADB706 Our tag: as12a46264 Jun 21 17:59:57 DEBUG[25160] chan_sip.c: = No match Their Call ID: VASi7D6EAXbgM4Gn@172.16.10.124 Their Tag WAxzvQzTfYiYK7En Our tag: as7e8ef3b7 Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Allocating new SIP dialog for d7fef62e-79c1d760-3e4cde55@172.16.10.112 - INVITE (With RTP) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Begin: parsing SIP "Supported: 100rel,replaces" Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Found SIP option: -100rel- Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Matched SIP option: 100rel Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Found SIP option: -replaces- Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Matched SIP option: replaces Jun 21 17:59:57 DEBUG[25160] chan_sip.c: * SIP extension value: 3 for call d7fef62e-79c1d760-3e4cde55@172.16.10.112 Jun 21 17:59:57 VERBOSE[25160] logger.c: Using INVITE request as basis request - d7fef62e-79c1d760-3e4cde55@172.16.10.112 Jun 21 17:59:57 VERBOSE[25160] logger.c: Sending to 172.16.10.112 : 5060 (non-NAT) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Setting NAT on RTP to 0 Jun 21 17:59:57 VERBOSE[25160] logger.c: Reliably Transmitting (no NAT) to 172.16.10.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK68f3c80c49CAB3BB;received=172.16.10.112 From: "Teste IP300" ;tag=274E1422-E9CA97BF To: ;tag=as6018bc76 Call-ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="2619cacc" Content-Length: 0 --- Jun 21 17:59:57 DEBUG[25160] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #28 Jun 21 17:59:57 VERBOSE[25160] logger.c: Scheduling destruction of call 'd7fef62e-79c1d760-3e4cde55@172.16.10.112' in 15000 ms Jun 21 17:59:57 VERBOSE[25160] logger.c: Found user '3053' Jun 21 17:59:57 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.112:5060: ACK sip:3054@pfdesenv.planetarium.com.br:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK68f3c80c49CAB3BB From: "Teste IP300" ;tag=274E1422-E9CA97BF To: ;tag=as6018bc76 CSeq: 1 ACK Call-ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 Max-Forwards: 70 Content-Length: 0 Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 0: ACK sip:3054@pfdesenv.planetarium.com.br:5060 SIP/2.0 (53) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK68f3c80c49CAB3BB (66) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 2: From: "Teste IP300" ;tag=274E1422-E9CA97BF (80) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 3: To: ;tag=as6018bc76 (68) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 4: CSeq: 1 ACK (11) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 5: Call-ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 (49) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 6: Contact: (38) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 (54) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 9: Max-Forwards: 70 (16) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 10: Content-Length: 0 (17) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 11: (0) Jun 21 17:59:57 VERBOSE[25160] logger.c: --- (11 headers 0 lines)Jun 21 17:59:57 VERBOSE[25160] logger.c: --- (11 headers 0 lines)--- Jun 21 17:59:57 DEBUG[25160] chan_sip.c: = Found Their Call ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 Their Tag 274E1422-E9CA97BF Our tag: as6018bc76 Jun 21 17:59:57 DEBUG[25160] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Jun 21 17:59:57 DEBUG[25160] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #28 Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Stopping retransmission on 'd7fef62e-79c1d760-3e4cde55@172.16.10.112' of Response 1: Match Found Jun 21 17:59:57 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.112:5060: INVITE sip:3054@pfdesenv.planetarium.com.br:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK9f4c2679A1F318F4 From: "Teste IP300" ;tag=274E1422-E9CA97BF To: CSeq: 2 INVITE Call-ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="3053", realm="asterisk", nonce="2619cacc", uri="sip:3054@pfdesenv.planetarium.com.br:5060;user=phone", response="74931ba2c9721e42a43421e8878a9a56", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 223 v=0 o=- 979455117 979455117 IN IP4 172.16.10.112 s=Polycom IP Phone c=IN IP4 172.16.10.112 t=0 0 a=sendrecv m=audio 2222 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 0: INVITE sip:3054@pfdesenv.planetarium.com.br:5060;user=phone SIP/2.0 (67) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK9f4c2679A1F318F4 (66) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 2: From: "Teste IP300" ;tag=274E1422-E9CA97BF (80) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 3: To: (53) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 4: CSeq: 2 INVITE (14) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 5: Call-ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 (49) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 6: Contact: (38) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 (54) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 9: Supported: 100rel,replaces (26) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 10: Allow-Events: talk,hold,conference (34) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 11: Proxy-Authorization: Digest username="3053", realm="asterisk", nonce="2619cacc", uri="sip:3054@pfdesenv.planetarium.com.br:5060;user=phone", response="74931ba2c9721e42a43421e8878a9a56", algorithm=MD5 (199) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 12: Max-Forwards: 70 (16) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 13: Content-Type: application/sdp (29) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 14: Content-Length: 223 (19) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 15: (0) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: v=0 (3) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: o=- 979455117 979455117 IN IP4 172.16.10.112 (44) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: s=Polycom IP Phone (18) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: c=IN IP4 172.16.10.112 (22) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: t=0 0 (5) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: a=sendrecv (10) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: m=audio 2222 RTP/AVP 0 8 101 (28) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Jun 21 17:59:57 VERBOSE[25160] logger.c: --- (15 headers 10 lines)Jun 21 17:59:57 VERBOSE[25160] logger.c: --- (15 headers 10 lines)--- Jun 21 17:59:57 DEBUG[25160] chan_sip.c: = Found Their Call ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 Their Tag 274E1422-E9CA97BF Our tag: as6018bc76 Jun 21 17:59:57 DEBUG[25160] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Jun 21 17:59:57 VERBOSE[25160] logger.c: Using INVITE request as basis request - d7fef62e-79c1d760-3e4cde55@172.16.10.112 Jun 21 17:59:57 VERBOSE[25160] logger.c: Sending to 172.16.10.112 : 5060 (non-NAT) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Setting NAT on RTP to 0 Jun 21 17:59:57 VERBOSE[25160] logger.c: Found user '3053' Jun 21 17:59:57 VERBOSE[25160] logger.c: Found RTP audio format 0 Jun 21 17:59:57 VERBOSE[25160] logger.c: Found RTP audio format 8 Jun 21 17:59:57 VERBOSE[25160] logger.c: Found RTP audio format 101 Jun 21 17:59:57 VERBOSE[25160] logger.c: Peer audio RTP is at port 172.16.10.112:2222 Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Peer audio RTP is at port 172.16.10.112:2222 Jun 21 17:59:57 VERBOSE[25160] logger.c: Found description format PCMU Jun 21 17:59:57 VERBOSE[25160] logger.c: Found description format PCMA Jun 21 17:59:57 VERBOSE[25160] logger.c: Found description format telephone-event Jun 21 17:59:57 VERBOSE[25160] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Jun 21 17:59:57 VERBOSE[25160] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Checking SIP call limits for device 3053 Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Updating call counter for incoming call Jun 21 17:59:57 VERBOSE[25160] logger.c: Looking for 3054 in 3053_aux (domain pfdesenv.planetarium.com.br) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: build_route: Contact hop: Jun 21 17:59:57 VERBOSE[25160] logger.c: list_route: hop: Jun 21 17:59:57 VERBOSE[25160] logger.c: Transmitting (no NAT) to 172.16.10.112:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK9f4c2679A1F318F4;received=172.16.10.112 From: "Teste IP300" ;tag=274E1422-E9CA97BF To: Call-ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jun 21 17:59:57 DEBUG[25145] chan_sip.c: Checking device state for peer 3053 Jun 21 17:59:57 DEBUG[25145] devicestate.c: Changing state for SIP/3053 - state 2 (In use) Jun 21 17:59:57 DEBUG[25185] pbx.c: Expression result is '0' Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'GotoIf' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing GotoIf("SIP/3053-3155", "0 ? 1000") in new stack Jun 21 17:59:57 DEBUG[25185] pbx.c: Not taking any branch Jun 21 17:59:57 DEBUG[25185] pbx.c: Expression result is '1' Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'GotoIf' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing GotoIf("SIP/3053-3155", "1 ? 200:400") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Goto (3053_aux,3054,200) Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'DBget' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing DBget("SIP/3053-3155", "ramalbloqueado=BLOQUEIORAMAL/3053") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- DBget: varname=ramalbloqueado, family=BLOQUEIORAMAL, key=3053 Jun 21 17:59:57 DEBUG[25185] db.c: Unable to find key '3053' in family 'BLOQUEIORAMAL' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- DBget: Value not found in database. Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'Goto' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing Goto("SIP/3053-3155", "400") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Goto (3053_aux,3054,400) Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'SetVar' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing SetVar("SIP/3053-3155", "NUMERODISCADO=3054") in new stack Jun 21 17:59:57 DEBUG[25185] pbx.c: Expression result is '0' Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'GotoIf' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing GotoIf("SIP/3053-3155", "0 ? 404") in new stack Jun 21 17:59:57 DEBUG[25185] pbx.c: Not taking any branch Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'SetVar' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing SetVar("SIP/3053-3155", "TRANSFER_CONTEXT=3053_aux") in new stack Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'Set' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing Set("SIP/3053-3155", "GROUP=3053") in new stack Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'SetVar' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing SetVar("SIP/3053-3155", "LIMIT_WARNING_FILE=beep") in new stack Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'SetVar' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing SetVar("SIP/3053-3155", "LIMIT_TIMEOUT_FILE=beep5") in new stack Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'Goto' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing Goto("SIP/3053-3155", "500") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Goto (3053_aux,3054,500) Jun 21 17:59:57 DEBUG[25185] pbx.c: Expression result is '0' Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'GotoIf' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing GotoIf("SIP/3053-3155", "0 ? 700") in new stack Jun 21 17:59:57 DEBUG[25185] pbx.c: Not taking any branch Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'DBget' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing DBget("SIP/3053-3155", "GRAVACAOCHAMADASAIDA=GRAVACAOCHAMADASAIDA/3053") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- DBget: varname=GRAVACAOCHAMADASAIDA, family=GRAVACAOCHAMADASAIDA, key=3053 Jun 21 17:59:57 DEBUG[25185] db.c: Unable to find key '3053' in family 'GRAVACAOCHAMADASAIDA' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- DBget: Value not found in database. Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'Goto' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing Goto("SIP/3053-3155", "700") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Goto (3053_aux,3054,700) Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'Goto' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing Goto("SIP/3053-3155", "3053_out|3054|1") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Goto (3053_out,3054,1) Jun 21 17:59:57 DEBUG[25185] pbx.c: Expression result is '1' Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'GotoIf' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing GotoIf("SIP/3053-3155", "1 ? 2:5") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Goto (3053_out,3054,2) Jun 21 17:59:57 DEBUG[25185] pbx.c: Expression result is '1' Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'GotoIf' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing GotoIf("SIP/3053-3155", "1 ? 3:5") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Goto (3053_out,3054,3) Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'SetVar' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing SetVar("SIP/3053-3155", "CHAMADAINTERNA=T") in new stack Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'SetCIDNum' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing SetCIDNum("SIP/3053-3155", "3053") in new stack Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'Goto' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing Goto("SIP/3053-3155", "disca|3054|1") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Goto (disca,3054,1) Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'Goto' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing Goto("SIP/3053-3155", "3054_in|3054|1") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Goto (3054_in,3054,1) Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'Goto' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing Goto("SIP/3053-3155", "1000") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Goto (3054_in,3054,1000) Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'Macro' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing Macro("SIP/3053-3155", "atende|SIP/3054|3054") in new stack Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'SetLanguage' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing SetLanguage("SIP/3053-3155", "br") in new stack Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'SetVar' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing SetVar("SIP/3053-3155", "DIALTECHRAMAL=SIP/3054") in new stack Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'Goto' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing Goto("SIP/3053-3155", "100") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Goto (macro-atende,s,100) Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'DBget' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing DBget("SIP/3053-3155", "RAMALREDIR=SIGAME/3054") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- DBget: varname=RAMALREDIR, family=SIGAME, key=3054 Jun 21 17:59:57 DEBUG[25185] db.c: Unable to find key '3054' in family 'SIGAME' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- DBget: Value not found in database. Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'Goto' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing Goto("SIP/3053-3155", "s|150") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Goto (macro-atende,s,150) Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'DBget' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing DBget("SIP/3053-3155", "RAMALREDIR=SIGAMEEXTERNO/3054") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- DBget: varname=RAMALREDIR, family=SIGAMEEXTERNO, key=3054 Jun 21 17:59:57 DEBUG[25185] db.c: Unable to find key '3054' in family 'SIGAMEEXTERNO' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- DBget: Value not found in database. Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'Goto' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing Goto("SIP/3053-3155", "s|300") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Goto (macro-atende,s,300) Jun 21 17:59:57 DEBUG[25185] pbx.c: Expression result is '1' Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'GotoIf' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing GotoIf("SIP/3053-3155", "1 ? 301:302") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Goto (macro-atende,s,301) Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'SetVar' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing SetVar("SIP/3053-3155", "RAMALFINAL=3054") in new stack Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'DBput' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing DBput("SIP/3053-3155", "ULTIMOCHAMADO/3054=3053") in new stack Jun 21 17:59:57 WARNING[25185] app_db.c: This application has been deprecated, please use the ${DB(family/key)} function instead. Jun 21 17:59:57 VERBOSE[25185] logger.c: -- DBput: family=ULTIMOCHAMADO, key=3054, value=3053 Jun 21 17:59:57 DEBUG[25186] 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. Jun 21 17:59:57 DEBUG[25185] pbx.c: Expression result is '0' Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'GotoIf' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing GotoIf("SIP/3053-3155", "0 ? 304:305") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Goto (macro-atende,s,305) Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'Goto' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing Goto("SIP/3053-3155", "350") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Goto (macro-atende,s,350) Jun 21 17:59:57 DEBUG[25185] pbx.c: Expression result is '1' Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'GotoIf' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing GotoIf("SIP/3053-3155", "1 ? 500") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Goto (macro-atende,s,500) Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'Set' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing Set("SIP/3053-3155", "OUTBOUND_GROUP=3054") in new stack Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'DBget' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing DBget("SIP/3053-3155", "MAXLIGACOES=MAXLIGACOES/3054") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- DBget: varname=MAXLIGACOES, family=MAXLIGACOES, key=3054 Jun 21 17:59:57 DEBUG[25185] db.c: Unable to find key '3054' in family 'MAXLIGACOES' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- DBget: Value not found in database. Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'Goto' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing Goto("SIP/3053-3155", "s|700") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Goto (macro-atende,s,700) Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'StopMonitor' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing StopMonitor("SIP/3053-3155", "") in new stack Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'DBget' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing DBget("SIP/3053-3155", "GRAVACAOCHAMADA=GRAVACAOCHAMADA/3054") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- DBget: varname=GRAVACAOCHAMADA, family=GRAVACAOCHAMADA, key=3054 Jun 21 17:59:57 DEBUG[25185] db.c: Unable to find key '3054' in family 'GRAVACAOCHAMADA' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- DBget: Value not found in database. Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'Goto' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing Goto("SIP/3053-3155", "s|900") in new stack Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Goto (macro-atende,s,900) Jun 21 17:59:57 DEBUG[25185] pbx.c: Launching 'Dial' Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Executing Dial("SIP/3053-3155", "SIP/3054|30|tT") in new stack Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Setting NAT on RTP to 524288 Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-900. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable MACRO_DEPTH. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-802. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable DBGETSTATUS. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-701. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-700. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-602. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-501. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable OUTBOUND_GROUP. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-500. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-350. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-305. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-303. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-302. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable RAMALFINAL. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-301. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-300. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-251. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-150. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-201. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-100. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-3. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable DIALTECHRAMAL. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-2. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-macro-atende-s-1. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable ARG2. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable ARG1. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable MACRO_PRIORITY. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable MACRO_CONTEXT. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable MACRO_EXTEN. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3054_in-3054-1000. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3054_in-3054-1. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-disca-3054-1. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_out-3054-5. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_out-3054-4. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable CHAMADAINTERNA. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_out-3054-3. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_out-3054-2. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_out-3054-1. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_aux-3054-700. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_aux-3054-602. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_aux-3054-501. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_aux-3054-500. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_aux-3054-406. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable LIMIT_TIMEOUT_FILE. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_aux-3054-405. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable LIMIT_WARNING_FILE. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_aux-3054-404. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable GROUP. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_aux-3054-403. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable TRANSFER_CONTEXT. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_aux-3054-402. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_aux-3054-401. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable NUMERODISCADO. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_aux-3054-400. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_aux-3054-301. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_aux-3054-200. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_aux-3054-2. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable STACK-3053_aux-3054-1. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable SIPCALLID. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable SIPUSERAGENT. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable SIPDOMAIN. Jun 21 17:59:57 DEBUG[25185] channel.c: Not copying variable SIPURI. Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Outgoing Call for 3054 Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Updating call counter for outgoing call Jun 21 17:59:57 VERBOSE[25185] logger.c: We're at 200.196.44.45 port 18108 Jun 21 17:59:57 VERBOSE[25185] logger.c: Adding codec 0x8 (alaw) to SDP Jun 21 17:59:57 VERBOSE[25185] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Header 0: INVITE sip:3054@172.16.10.130;user=phone SIP/2.0 (48) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK4c013b9d;rport (64) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Header 2: From: "Teste Desenv Polycom" ;tag=as56128416 (68) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Header 3: To: (39) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Header 4: Contact: (33) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Header 5: Call-ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 (55) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Header 6: CSeq: 102 INVITE (16) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Header 8: Max-Forwards: 70 (16) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Header 9: Date: Wed, 21 Jun 2006 20:59:57 GMT (35) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Header 11: Content-Type: application/sdp (29) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Header 12: Content-Length: 218 (19) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Header 13: (0) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Line: v=0 (3) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Line: o=root 25128 25128 IN IP4 200.196.44.45 (39) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Line: s=session (9) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Line: c=IN IP4 200.196.44.45 (22) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Line: t=0 0 (5) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Line: m=audio 18108 RTP/AVP 8 101 (27) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Line: a=fmtp:101 0-16 (15) Jun 21 17:59:57 DEBUG[25185] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Jun 21 17:59:57 VERBOSE[25185] logger.c: 13 headers, 10 lines Jun 21 17:59:57 VERBOSE[25185] logger.c: Reliably Transmitting (NAT) to 172.16.10.130:5060: INVITE sip:3054@172.16.10.130;user=phone SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK4c013b9d;rport From: "Teste Desenv Polycom" ;tag=as56128416 To: Contact: Call-ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Jun 2006 20:59:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 218 v=0 o=root 25128 25128 IN IP4 200.196.44.45 s=session c=IN IP4 200.196.44.45 t=0 0 m=audio 18108 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jun 21 17:59:57 DEBUG[25185] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #30 Jun 21 17:59:57 VERBOSE[25185] logger.c: -- Called 3054 Jun 21 17:59:57 DEBUG[25185] channel.c: Set channel SIP/3054-ef3c to read format alaw Jun 21 17:59:57 DEBUG[25185] channel.c: Set channel SIP/3053-3155 to write format alaw Jun 21 17:59:57 DEBUG[25185] channel.c: Set channel SIP/3053-3155 to read format alaw Jun 21 17:59:57 DEBUG[25185] channel.c: Set channel SIP/3054-ef3c to write format alaw Jun 21 17:59:57 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.130:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK4c013b9d;rport From: "Teste Desenv Polycom" ;tag=as56128416 To: Call-ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 CSeq: 102 INVITE User-Agent: Grandstream BT100 1.0.6.7 Content-Length: 0 Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK4c013b9d;rport (64) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 2: From: "Teste Desenv Polycom" ;tag=as56128416 (68) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 3: To: (39) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 4: Call-ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 (55) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 5: CSeq: 102 INVITE (16) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 6: User-Agent: Grandstream BT100 1.0.6.7 (37) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 7: Content-Length: 0 (17) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 8: (0) Jun 21 17:59:57 VERBOSE[25160] logger.c: --- (8 headers 0 lines)Jun 21 17:59:57 VERBOSE[25160] logger.c: --- (8 headers 0 lines)--- Jun 21 17:59:57 DEBUG[25160] chan_sip.c: = Found Their Call ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 Their Tag Our tag: as56128416 Jun 21 17:59:57 DEBUG[25160] chan_sip.c: *** SIP TIMER: Cancelling retransmission #30 - INVITE (got response) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6cc9df0942d9ab2a25810b8033722227@200.196.44.45' Request 102: Found Jun 21 17:59:57 DEBUG[25160] chan_sip.c: SIP response 100 to standard invite Jun 21 17:59:57 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.130:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK4c013b9d;rport From: "Teste Desenv Polycom" ;tag=as56128416 To: ;tag=2df1a0c4b2f73fe4 Call-ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 CSeq: 102 INVITE User-Agent: Grandstream BT100 1.0.6.7 Content-Length: 0 Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK4c013b9d;rport (64) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 2: From: "Teste Desenv Polycom" ;tag=as56128416 (68) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 3: To: ;tag=2df1a0c4b2f73fe4 (60) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 4: Call-ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 (55) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 5: CSeq: 102 INVITE (16) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 6: User-Agent: Grandstream BT100 1.0.6.7 (37) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 7: Content-Length: 0 (17) Jun 21 17:59:57 DEBUG[25160] chan_sip.c: Header 8: (0) Jun 21 17:59:57 VERBOSE[25160] logger.c: --- (8 headers 0 lines)Jun 21 17:59:57 VERBOSE[25160] logger.c: --- (8 headers 0 lines)--- Jun 21 17:59:57 DEBUG[25160] chan_sip.c: = Found Their Call ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 Their Tag Our tag: as56128416 Jun 21 17:59:57 DEBUG[25160] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6cc9df0942d9ab2a25810b8033722227@200.196.44.45' Request 102: Found Jun 21 17:59:57 DEBUG[25160] chan_sip.c: SIP response 180 to standard invite Jun 21 17:59:57 VERBOSE[25185] logger.c: -- SIP/3054-ef3c is ringing Jun 21 17:59:57 VERBOSE[25185] logger.c: Transmitting (no NAT) to 172.16.10.112:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK9f4c2679A1F318F4;received=172.16.10.112 From: "Teste IP300" ;tag=274E1422-E9CA97BF To: ;tag=as1cfc2349 Call-ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jun 21 17:59:57 DEBUG[25145] chan_sip.c: Checking device state for peer 3054 Jun 21 17:59:57 DEBUG[25145] devicestate.c: Changing state for SIP/3054 - state 6 (Ringing) Jun 21 17:59:57 DEBUG[25187] app_queue.c: Device 'SIP/3054' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. Jun 21 17:59:58 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.130:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK4c013b9d;rport From: "Teste Desenv Polycom" ;tag=as56128416 To: ;tag=2df1a0c4b2f73fe4 Call-ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 CSeq: 102 INVITE User-Agent: Grandstream BT100 1.0.6.7 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Supported: replaces Content-Length: 158 v=0 o=3054 8000 8000 IN IP4 172.16.10.130 s=SIP Call c=IN IP4 172.16.10.130 t=0 0 m=audio 10000 RTP/AVP 8 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK4c013b9d;rport (64) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 2: From: "Teste Desenv Polycom" ;tag=as56128416 (68) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 3: To: ;tag=2df1a0c4b2f73fe4 (60) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 4: Call-ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 (55) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 5: CSeq: 102 INVITE (16) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 6: User-Agent: Grandstream BT100 1.0.6.7 (37) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 7: Contact: (44) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 9: Content-Type: application/sdp (29) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 10: Supported: replaces (19) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 11: Content-Length: 158 (19) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 12: (0) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Line: v=0 (3) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Line: o=3054 8000 8000 IN IP4 172.16.10.130 (37) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Line: s=SIP Call (10) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Line: c=IN IP4 172.16.10.130 (22) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Line: t=0 0 (5) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Line: m=audio 10000 RTP/AVP 8 (23) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Line: a=sendrecv (10) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Line: a=ptime:20 (10) Jun 21 17:59:58 VERBOSE[25160] logger.c: --- (12 headers 9 lines)Jun 21 17:59:58 VERBOSE[25160] logger.c: --- (12 headers 9 lines)--- Jun 21 17:59:58 DEBUG[25160] chan_sip.c: = Found Their Call ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 Their Tag 2df1a0c4b2f73fe4 Our tag: as56128416 Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Acked pending invite 102 Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Stopping retransmission on '6cc9df0942d9ab2a25810b8033722227@200.196.44.45' of Request 102: Match Found Jun 21 17:59:58 DEBUG[25160] chan_sip.c: SIP response 200 to standard invite Jun 21 17:59:58 VERBOSE[25160] logger.c: Found RTP audio format 8 Jun 21 17:59:58 VERBOSE[25160] logger.c: Peer audio RTP is at port 172.16.10.130:10000 Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Peer audio RTP is at port 172.16.10.130:10000 Jun 21 17:59:58 VERBOSE[25160] logger.c: Found description format PCMA Jun 21 17:59:58 VERBOSE[25160] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Jun 21 17:59:58 VERBOSE[25160] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: build_route: Contact hop: Jun 21 17:59:58 VERBOSE[25160] logger.c: list_route: hop: Jun 21 17:59:58 VERBOSE[25160] logger.c: set_destination: Parsing for address/port to send to Jun 21 17:59:58 VERBOSE[25160] logger.c: set_destination: set destination to 172.16.10.130, port 5060 Jun 21 17:59:58 VERBOSE[25160] logger.c: Transmitting (NAT) to 172.16.10.130:5060: ACK sip:3054@172.16.10.130;user=phone SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK5778d5e0;rport From: "Teste Desenv Polycom" ;tag=as56128416 To: ;tag=2df1a0c4b2f73fe4 Contact: Call-ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jun 21 17:59:58 VERBOSE[25185] logger.c: -- SIP/3054-ef3c answered SIP/3053-3155 Jun 21 17:59:58 DEBUG[25185] channel.c: Set channel SIP/3053-3155 to read format alaw Jun 21 17:59:58 DEBUG[25185] channel.c: Set channel SIP/3054-ef3c to write format alaw Jun 21 17:59:58 DEBUG[25185] channel.c: Set channel SIP/3054-ef3c to read format alaw Jun 21 17:59:58 DEBUG[25185] channel.c: Set channel SIP/3053-3155 to write format alaw Jun 21 17:59:58 DEBUG[25185] chan_sip.c: sip_answer(SIP/3053-3155) Jun 21 17:59:58 VERBOSE[25185] logger.c: We're at 200.196.44.45 port 19336 Jun 21 17:59:58 VERBOSE[25185] logger.c: Adding codec 0x8 (alaw) to SDP Jun 21 17:59:58 VERBOSE[25185] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Jun 21 17:59:58 VERBOSE[25185] logger.c: Reliably Transmitting (no NAT) to 172.16.10.112:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK9f4c2679A1F318F4;received=172.16.10.112 From: "Teste IP300" ;tag=274E1422-E9CA97BF To: ;tag=as1cfc2349 Call-ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 CSeq: 2 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 25128 25128 IN IP4 200.196.44.45 s=session c=IN IP4 200.196.44.45 t=0 0 m=audio 19336 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jun 21 17:59:58 DEBUG[25185] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #32 Jun 21 17:59:58 VERBOSE[25185] logger.c: -- Attempting native bridge of SIP/3053-3155 and SIP/3054-ef3c Jun 21 17:59:58 DEBUG[25145] chan_sip.c: Checking device state for peer 3054 Jun 21 17:59:58 DEBUG[25145] devicestate.c: Changing state for SIP/3054 - state 2 (In use) Jun 21 17:59:58 DEBUG[25145] chan_sip.c: Checking device state for peer 3053 Jun 21 17:59:58 DEBUG[25145] devicestate.c: Changing state for SIP/3053 - state 2 (In use) Jun 21 17:59:58 DEBUG[25188] app_queue.c: Device 'SIP/3054' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jun 21 17:59:58 DEBUG[25189] 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. Jun 21 17:59:58 DEBUG[25185] rtp.c: Ooh, format changed from unknown to alaw Jun 21 17:59:58 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.112:5060: ACK sip:3054@200.196.44.45 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK1057c0c8B1BDF87 From: "Teste IP300" ;tag=274E1422-E9CA97BF To: ;tag=as1cfc2349 CSeq: 2 ACK Call-ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 Proxy-Authorization: Digest username="3053", realm="asterisk", nonce="2619cacc", uri="sip:3054@pfdesenv.planetarium.com.br:5060;user=phone", response="74931ba2c9721e42a43421e8878a9a56", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 0: ACK sip:3054@200.196.44.45 SIP/2.0 (34) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK1057c0c8B1BDF87 (65) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 2: From: "Teste IP300" ;tag=274E1422-E9CA97BF (80) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 3: To: ;tag=as1cfc2349 (68) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 4: CSeq: 2 ACK (11) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 5: Call-ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 (49) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 6: Contact: (38) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 (54) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 9: Proxy-Authorization: Digest username="3053", realm="asterisk", nonce="2619cacc", uri="sip:3054@pfdesenv.planetarium.com.br:5060;user=phone", response="74931ba2c9721e42a43421e8878a9a56", algorithm=MD5 (199) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 10: Max-Forwards: 70 (16) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 11: Content-Length: 0 (17) Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Header 12: (0) Jun 21 17:59:58 VERBOSE[25160] logger.c: --- (12 headers 0 lines)Jun 21 17:59:58 VERBOSE[25160] logger.c: --- (12 headers 0 lines)--- Jun 21 17:59:58 DEBUG[25160] chan_sip.c: = No match Their Call ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 Their Tag 2df1a0c4b2f73fe4 Our tag: as56128416 Jun 21 17:59:58 DEBUG[25160] chan_sip.c: = Found Their Call ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 Their Tag 274E1422-E9CA97BF Our tag: as1cfc2349 Jun 21 17:59:58 DEBUG[25160] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Jun 21 17:59:58 DEBUG[25160] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #32 Jun 21 17:59:58 DEBUG[25160] chan_sip.c: Stopping retransmission on 'd7fef62e-79c1d760-3e4cde55@172.16.10.112' of Response 2: Match Found Jun 21 17:59:58 DEBUG[25185] rtp.c: Ooh, format changed from unknown to alaw Jun 21 18:00:00 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.112:5060: REFER sip:3050@200.196.44.45 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK286b30caC8E3F481 From: ;tag=590337CD-1BADB706 To: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 CSeq: 2 REFER Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Contact: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 Refer-To: Referred-By: Max-Forwards: 70 Content-Length: 0 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 0: REFER sip:3050@200.196.44.45 SIP/2.0 (36) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK286b30caC8E3F481 (66) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 2: From: ;tag=590337CD-1BADB706 (57) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 3: To: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 (78) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 4: CSeq: 2 REFER (13) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 5: Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 (55) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 6: Contact: (38) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 7: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 (54) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 8: Refer-To: (164) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 9: Referred-By: (42) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 10: Max-Forwards: 70 (16) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 11: Content-Length: 0 (17) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 12: (0) Jun 21 18:00:00 VERBOSE[25160] logger.c: --- (12 headers 0 lines)Jun 21 18:00:00 VERBOSE[25160] logger.c: --- (12 headers 0 lines)--- Jun 21 18:00:00 DEBUG[25160] chan_sip.c: = No match Their Call ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 Their Tag 2df1a0c4b2f73fe4 Our tag: as56128416 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: = No match Their Call ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 Their Tag 274E1422-E9CA97BF Our tag: as1cfc2349 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: = Found Their Call ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Their Tag 590337CD-1BADB706 Our tag: as12a46264 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: **** Received REFER (9) - Command in SIP REFER Jun 21 18:00:00 DEBUG[25160] chan_sip.c: SIP call transfer received for call 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 (REFER)! Jun 21 18:00:00 VERBOSE[25160] logger.c: Transfer to 3054 in 3053_aux Jun 21 18:00:00 VERBOSE[25160] logger.c: Transfer from 3053 in 3053_aux Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Assigning Replace-Call-ID Info d7fef62e-79c1d760-3e4cde55@172.16.10.112 to REPLACE_CALL_ID Jun 21 18:00:00 DEBUG[25160] chan_sip.c: 202 Accepted (supervised) Jun 21 18:00:00 DEBUG[25160] chan_agent.c: Asked for bridged channel on 'SIP/3053-7a88'/'Agent/3053', returning 'SIP/3050-312e' Jun 21 18:00:00 DEBUG[25160] channel.c: Set channel SIP/3050-312e to write format alaw Jun 21 18:00:00 VERBOSE[25160] logger.c: -- Stopped music on hold on SIP/3050-312e Jun 21 18:00:00 DEBUG[25160] channel.c: Scheduling timer at 0 sample intervals Jun 21 18:00:00 DEBUG[25160] chan_agent.c: Asked for bridged channel on 'SIP/3050-312e'/'Agent/3053', returning 'SIP/3053-7a88' Jun 21 18:00:00 DEBUG[25160] channel.c: Planning to masquerade channel Agent/3053 into the structure of SIP/3053-3155 Jun 21 18:00:00 DEBUG[25160] channel.c: Done planning to masquerade channel Agent/3053 into the structure of SIP/3053-3155 Jun 21 18:00:00 VERBOSE[25160] logger.c: Transmitting (no NAT) to 172.16.10.112:5060: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK286b30caC8E3F481;received=172.16.10.112 From: ;tag=590337CD-1BADB706 To: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 CSeq: 2 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jun 21 18:00:00 VERBOSE[25160] logger.c: set_destination: Parsing for address/port to send to Jun 21 18:00:00 VERBOSE[25160] logger.c: set_destination: set destination to 172.16.10.112, port 5060 Jun 21 18:00:00 VERBOSE[25160] logger.c: Reliably Transmitting (no NAT) to 172.16.10.112:5060: NOTIFY sip:3053@172.16.10.112:5060 SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK655f1bf6;rport From: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 To: ;tag=590337CD-1BADB706 Contact: Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- Jun 21 18:00:00 DEBUG[25160] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #33 Jun 21 18:00:00 VERBOSE[25160] logger.c: set_destination: Parsing for address/port to send to Jun 21 18:00:00 VERBOSE[25160] logger.c: set_destination: set destination to 172.16.10.112, port 5060 Jun 21 18:00:00 VERBOSE[25160] logger.c: Reliably Transmitting (no NAT) to 172.16.10.112:5060: BYE sip:3053@172.16.10.112:5060 SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK43a7925c;rport From: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 To: ;tag=590337CD-1BADB706 Contact: Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jun 21 18:00:00 DEBUG[25160] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #34 Jun 21 18:00:00 DEBUG[25185] channel.c: Actually Masquerading Agent/3053(6) into the structure of SIP/3053-3155(6) Jun 21 18:00:00 DEBUG[25185] channel.c: Got clone lock for masquerade on 'Agent/3053' at 0x8e97234 Jun 21 18:00:00 DEBUG[25185] chan_sip.c: Hangup call Agent/3053, SIP callid d7fef62e-79c1d760-3e4cde55@172.16.10.112) Jun 21 18:00:00 DEBUG[25185] chan_sip.c: update_call_counter(3053) - decrement call limit counter Jun 21 18:00:00 DEBUG[25185] chan_sip.c: Updating call counter for incoming call Jun 21 18:00:00 VERBOSE[25185] logger.c: set_destination: Parsing for address/port to send to Jun 21 18:00:00 VERBOSE[25185] logger.c: set_destination: set destination to 172.16.10.112, port 5060 Jun 21 18:00:00 VERBOSE[25185] logger.c: Reliably Transmitting (no NAT) to 172.16.10.112:5060: BYE sip:3053@172.16.10.112:5060 SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK69f470c0;rport From: ;tag=as1cfc2349 To: "Teste IP300" ;tag=274E1422-E9CA97BF Contact: Call-ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jun 21 18:00:00 DEBUG[25185] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #35 Jun 21 18:00:00 DEBUG[25185] channel.c: Set channel Agent/3053 to write format alaw Jun 21 18:00:00 DEBUG[25185] channel.c: Set channel Agent/3053 to read format alaw Jun 21 18:00:00 DEBUG[25185] channel.c: Putting channel Agent/3053 in 8/8 formats Jun 21 18:00:00 DEBUG[25185] channel.c: Released clone lock on 'SIP/3053-3155' Jun 21 18:00:00 DEBUG[25185] channel.c: Done Masquerading Agent/3053 (6) Jun 21 18:00:00 DEBUG[25173] channel.c: Didn't get a frame from channel: SIP/3053-3155 Jun 21 18:00:00 DEBUG[25173] channel.c: Bridge stops bridging channels SIP/3050-312e and SIP/3053-3155 Jun 21 18:00:00 DEBUG[25173] channel.c: Hanging up zombie 'SIP/3053-3155' Jun 21 18:00:00 DEBUG[25173] pbx.c: Spawn extension (fila_desenvolvimento,3097,12) exited non-zero on 'SIP/3050-312e' Jun 21 18:00:00 DEBUG[25173] channel.c: Hanging up channel 'SIP/3050-312e' Jun 21 18:00:00 DEBUG[25173] chan_sip.c: Hangup call SIP/3050-312e, SIP callid VASi7D6EAXbgM4Gn@172.16.10.124) Jun 21 18:00:00 DEBUG[25173] chan_sip.c: update_call_counter(3050) - decrement call limit counter Jun 21 18:00:00 DEBUG[25173] chan_sip.c: Updating call counter for incoming call Jun 21 18:00:00 VERBOSE[25173] logger.c: set_destination: Parsing for address/port to send to Jun 21 18:00:00 VERBOSE[25173] logger.c: set_destination: set destination to 172.16.10.124, port 5060 Jun 21 18:00:00 VERBOSE[25173] 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=z9hG4bK693d749a;rport From: "3097" ;tag=as7e8ef3b7 To: "3050" ;tag=WAxzvQzTfYiYK7En Contact: Call-ID: VASi7D6EAXbgM4Gn@172.16.10.124 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jun 21 18:00:00 DEBUG[25173] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #36 Jun 21 18:00:00 DEBUG[25173] res_monitor.c: monitor executing "/bin/nice -n 40 /usr/local/sbin/gc.registragravacao" "/var/spool/asterisk/monitor/1150923590.0-filadesenvolvimento-3097-in-20060621-175951-3050-in.WAV" "/var/spool/asterisk/monitor/1150923590.0-filadesenvolvimento-3097-in-20060621-175951-3050-out.WAV" "/var/spool/asterisk/monitor/1150923590.0-filadesenvolvimento-3097-in-20060621-175951-3050.WAV" & Jun 21 18:00:00 DEBUG[25145] chan_sip.c: Checking device state for peer 3053 Jun 21 18:00:00 DEBUG[25145] devicestate.c: Changing state for SIP/3053 - state 2 (In use) Jun 21 18:00:00 DEBUG[25145] chan_sip.c: Checking device state for peer 3050 Jun 21 18:00:00 DEBUG[25145] devicestate.c: Changing state for SIP/3050 - state 1 (Not in use) Jun 21 18:00:00 DEBUG[25192] 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. Jun 21 18:00:00 DEBUG[25193] app_queue.c: Device 'SIP/3050' changed to state '1' (Not in use) Jun 21 18:00:00 VERBOSE[25160] 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=z9hG4bK693d749a;rport Call-ID: VASi7D6EAXbgM4Gn@172.16.10.124 CSeq: 102 BYE From: "3097" ;tag=as7e8ef3b7 To: "3050" ;tag=WAxzvQzTfYiYK7En Contact: Content-Length: 0 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK693d749a;rport (64) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 2: Call-ID: VASi7D6EAXbgM4Gn@172.16.10.124 (39) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 3: CSeq: 102 BYE (13) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 4: From: "3097" ;tag=as7e8ef3b7 (66) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 5: To: "3050" ;tag=WAxzvQzTfYiYK7En (70) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 6: Contact: (38) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 7: Content-Length: 0 (17) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 8: (0) Jun 21 18:00:00 VERBOSE[25160] logger.c: --- (8 headers 0 lines)Jun 21 18:00:00 VERBOSE[25160] logger.c: --- (8 headers 0 lines)--- Jun 21 18:00:00 DEBUG[25160] chan_sip.c: = No match Their Call ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 Their Tag 2df1a0c4b2f73fe4 Our tag: as56128416 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: = No match Their Call ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 Their Tag 274E1422-E9CA97BF Our tag: as1cfc2349 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: = No match Their Call ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Their Tag 590337CD-1BADB706 Our tag: as12a46264 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: = Found Their Call ID: VASi7D6EAXbgM4Gn@172.16.10.124 Their Tag WAxzvQzTfYiYK7En Our tag: as7e8ef3b7 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #36 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Stopping retransmission on 'VASi7D6EAXbgM4Gn@172.16.10.124' of Request 102: Match Found Jun 21 18:00:00 VERBOSE[25160] logger.c: Destroying call 'VASi7D6EAXbgM4Gn@172.16.10.124' Jun 21 18:00:00 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.112:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK655f1bf6;rport From: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 To: ;tag=590337CD-1BADB706 CSeq: 103 NOTIFY Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Contact: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 Content-Length: 0 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK655f1bf6;rport (64) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 2: From: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 (80) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 3: To: ;tag=590337CD-1BADB706 (55) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 4: CSeq: 103 NOTIFY (16) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 5: Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 (55) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 6: Contact: (38) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 7: Event: refer;id=2 (17) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 (54) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 9: Content-Length: 0 (17) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 10: (0) Jun 21 18:00:00 VERBOSE[25160] logger.c: --- (10 headers 0 lines)Jun 21 18:00:00 VERBOSE[25160] logger.c: --- (10 headers 0 lines)--- Jun 21 18:00:00 DEBUG[25160] chan_sip.c: = No match Their Call ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 Their Tag 2df1a0c4b2f73fe4 Our tag: as56128416 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: = No match Their Call ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 Their Tag 274E1422-E9CA97BF Our tag: as1cfc2349 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: = Found Their Call ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Their Tag 590337CD-1BADB706 Our tag: as12a46264 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #33 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Stopping retransmission on '119ce4a9105ce0a33433067e7132ca34@200.196.44.45' of Request 103: Match Found Jun 21 18:00:00 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.112:5060: BYE sip:3050@200.196.44.45 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK3ef842dc17074B4B From: ;tag=590337CD-1BADB706 To: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 CSeq: 3 BYE Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Contact: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 Max-Forwards: 70 Content-Length: 0 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 0: BYE sip:3050@200.196.44.45 SIP/2.0 (34) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK3ef842dc17074B4B (66) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 2: From: ;tag=590337CD-1BADB706 (57) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 3: To: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 (78) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 4: CSeq: 3 BYE (11) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 5: Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 (55) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 6: Contact: (38) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 7: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 (54) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 8: Max-Forwards: 70 (16) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 9: Content-Length: 0 (17) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 10: (0) Jun 21 18:00:00 VERBOSE[25160] logger.c: --- (10 headers 0 lines)Jun 21 18:00:00 VERBOSE[25160] logger.c: --- (10 headers 0 lines)--- Jun 21 18:00:00 DEBUG[25160] chan_sip.c: = No match Their Call ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 Their Tag 2df1a0c4b2f73fe4 Our tag: as56128416 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: = No match Their Call ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 Their Tag 274E1422-E9CA97BF Our tag: as1cfc2349 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: = Found Their Call ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Their Tag 590337CD-1BADB706 Our tag: as12a46264 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: **** Received BYE (8) - Command in SIP BYE Jun 21 18:00:00 VERBOSE[25160] logger.c: Sending to 172.16.10.112 : 5060 (non-NAT) Jun 21 18:00:00 VERBOSE[25160] logger.c: Transmitting (no NAT) to 172.16.10.112:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.10.112:5060;branch=z9hG4bK3ef842dc17074B4B;received=172.16.10.112 From: ;tag=590337CD-1BADB706 To: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- Jun 21 18:00:00 DEBUG[25185] chan_agent.c: Bridge on 'SIP/3053-7a88' being cleared (2) Jun 21 18:00:00 DEBUG[25185] channel.c: Hanging up channel 'SIP/3053-7a88' Jun 21 18:00:00 DEBUG[25185] chan_sip.c: Hangup call SIP/3053-7a88, SIP callid 119ce4a9105ce0a33433067e7132ca34@200.196.44.45) Jun 21 18:00:00 DEBUG[25185] chan_sip.c: update_call_counter(3053) - decrement call limit counter Jun 21 18:00:00 DEBUG[25185] chan_sip.c: Updating call counter for incoming call Jun 21 18:00:00 DEBUG[25185] channel.c: Didn't get a frame from channel: Agent/3053 Jun 21 18:00:00 DEBUG[25185] channel.c: Bridge stops bridging channels Agent/3053 and SIP/3054-ef3c Jun 21 18:00:00 DEBUG[25185] channel.c: Hanging up channel 'SIP/3054-ef3c' Jun 21 18:00:00 DEBUG[25185] chan_sip.c: Hangup call SIP/3054-ef3c, SIP callid 6cc9df0942d9ab2a25810b8033722227@200.196.44.45) Jun 21 18:00:00 DEBUG[25185] chan_sip.c: update_call_counter(3054) - decrement call limit counter Jun 21 18:00:00 DEBUG[25185] chan_sip.c: Updating call counter for outgoing call Jun 21 18:00:00 VERBOSE[25185] logger.c: set_destination: Parsing for address/port to send to Jun 21 18:00:00 VERBOSE[25185] logger.c: set_destination: set destination to 172.16.10.130, port 5060 Jun 21 18:00:00 VERBOSE[25185] logger.c: Reliably Transmitting (NAT) to 172.16.10.130:5060: BYE sip:3054@172.16.10.130;user=phone SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK747ed1d3;rport From: "Teste Desenv Polycom" ;tag=as56128416 To: ;tag=2df1a0c4b2f73fe4 Contact: Call-ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jun 21 18:00:00 DEBUG[25185] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #37 Jun 21 18:00:00 DEBUG[25185] app_dial.c: Exiting with DIALSTATUS=ANSWER. Jun 21 18:00:00 DEBUG[25185] app_macro.c: Spawn extension (macro-atende,s,900) exited non-zero on 'Agent/3053' in macro 'atende' Jun 21 18:00:00 DEBUG[25185] pbx.c: Spawn extension (macro-atende,s,900) exited non-zero on 'Agent/3053' Jun 21 18:00:00 DEBUG[25185] pbx.c: Launching 'NoOp' Jun 21 18:00:00 VERBOSE[25185] logger.c: -- Executing NoOp("Agent/3053", "Fila:desenvolvimento:Ramal Teste 3050 2 3054") in new stack Jun 21 18:00:00 DEBUG[25185] pbx.c: Launching 'Hangup' Jun 21 18:00:00 VERBOSE[25185] logger.c: -- Executing Hangup("Agent/3053", "") in new stack Jun 21 18:00:00 DEBUG[25185] pbx.c: Spawn extension (macro-atende,h,2) exited non-zero on 'Agent/3053' Jun 21 18:00:00 DEBUG[25145] chan_sip.c: Checking device state for peer 3053 Jun 21 18:00:00 DEBUG[25145] devicestate.c: Changing state for SIP/3053 - state 1 (Not in use) Jun 21 18:00:00 DEBUG[25145] chan_sip.c: Checking device state for peer 3054 Jun 21 18:00:00 DEBUG[25145] devicestate.c: Changing state for SIP/3054 - state 1 (Not in use) Jun 21 18:00:00 DEBUG[25194] app_queue.c: Device 'SIP/3053' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Jun 21 18:00:00 DEBUG[25195] app_queue.c: Device 'SIP/3054' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Jun 21 18:00:00 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.130:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK747ed1d3;rport From: "Teste Desenv Polycom" ;tag=as56128416 To: ;tag=2df1a0c4b2f73fe4 Call-ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 CSeq: 103 BYE User-Agent: Grandstream BT100 1.0.6.7 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Supported: replaces Content-Length: 0 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK747ed1d3;rport (64) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 2: From: "Teste Desenv Polycom" ;tag=as56128416 (68) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 3: To: ;tag=2df1a0c4b2f73fe4 (60) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 4: Call-ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 (55) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 5: CSeq: 103 BYE (13) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 6: User-Agent: Grandstream BT100 1.0.6.7 (37) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 7: Contact: (44) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 9: Supported: replaces (19) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 10: Content-Length: 0 (17) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 11: (0) Jun 21 18:00:00 VERBOSE[25160] logger.c: --- (11 headers 0 lines)Jun 21 18:00:00 VERBOSE[25160] logger.c: --- (11 headers 0 lines)--- Jun 21 18:00:00 DEBUG[25160] chan_sip.c: = Found Their Call ID: 6cc9df0942d9ab2a25810b8033722227@200.196.44.45 Their Tag 2df1a0c4b2f73fe4 Our tag: as56128416 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #37 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Stopping retransmission on '6cc9df0942d9ab2a25810b8033722227@200.196.44.45' of Request 103: Match Found Jun 21 18:00:00 VERBOSE[25160] logger.c: Destroying call '6cc9df0942d9ab2a25810b8033722227@200.196.44.45' Jun 21 18:00:00 DEBUG[25185] channel.c: Hanging up channel 'Agent/3053' Jun 21 18:00:00 DEBUG[25185] chan_agent.c: Hangup called for state Up Jun 21 18:00:00 DEBUG[25145] devicestate.c: Changing state for Agent/3053 - state 1 (Not in use) Jun 21 18:00:00 DEBUG[25145] devicestate.c: Changing state for Agent/3053 - state 1 (Not in use) Jun 21 18:00:00 DEBUG[25196] app_queue.c: Device 'Agent/3053' changed to state '1' (Not in use) Jun 21 18:00:00 DEBUG[25197] app_queue.c: Device 'Agent/3053' changed to state '1' (Not in use) Jun 21 18:00:00 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.112:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK43a7925c;rport From: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 To: ;tag=590337CD-1BADB706 CSeq: 104 BYE Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Contact: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 Content-Length: 0 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 0: SIP/2.0 500 Internal Server Error (33) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK43a7925c;rport (64) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 2: From: "Fila:desenvolvimento:Ramal Teste" ;tag=as12a46264 (80) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 3: To: ;tag=590337CD-1BADB706 (55) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 4: CSeq: 104 BYE (13) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 5: Call-ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 (55) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 6: Contact: (38) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 7: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 (54) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 8: Content-Length: 0 (17) Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Header 9: (0) Jun 21 18:00:00 VERBOSE[25160] logger.c: --- (9 headers 0 lines)Jun 21 18:00:00 VERBOSE[25160] logger.c: --- (9 headers 0 lines)--- Jun 21 18:00:00 DEBUG[25160] chan_sip.c: = No match Their Call ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 Their Tag 274E1422-E9CA97BF Our tag: as1cfc2349 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: = Found Their Call ID: 119ce4a9105ce0a33433067e7132ca34@200.196.44.45 Their Tag 590337CD-1BADB706 Our tag: as12a46264 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #34 Jun 21 18:00:00 DEBUG[25160] chan_sip.c: Stopping retransmission on '119ce4a9105ce0a33433067e7132ca34@200.196.44.45' of Request 104: Match Found Jun 21 18:00:00 VERBOSE[25160] logger.c: -- Incoming call: Got SIP response 500 "Internal Server Error" back from 172.16.10.112 Jun 21 18:00:00 VERBOSE[25160] logger.c: Destroying call '119ce4a9105ce0a33433067e7132ca34@200.196.44.45' Jun 21 18:00:01 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.112:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK69f470c0;rport From: ;tag=as1cfc2349 To: "Teste IP300" ;tag=274E1422-E9CA97BF CSeq: 102 BYE Call-ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 Contact: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 Content-Length: 0 Jun 21 18:00:01 DEBUG[25160] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Jun 21 18:00:01 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK69f470c0;rport (64) Jun 21 18:00:01 DEBUG[25160] chan_sip.c: Header 2: From: ;tag=as1cfc2349 (70) Jun 21 18:00:01 DEBUG[25160] chan_sip.c: Header 3: To: "Teste IP300" ;tag=274E1422-E9CA97BF (78) Jun 21 18:00:01 DEBUG[25160] chan_sip.c: Header 4: CSeq: 102 BYE (13) Jun 21 18:00:01 DEBUG[25160] chan_sip.c: Header 5: Call-ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 (49) Jun 21 18:00:01 DEBUG[25160] chan_sip.c: Header 6: Contact: (38) Jun 21 18:00:01 DEBUG[25160] chan_sip.c: Header 7: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.6.0036 (54) Jun 21 18:00:01 DEBUG[25160] chan_sip.c: Header 8: Content-Length: 0 (17) Jun 21 18:00:01 DEBUG[25160] chan_sip.c: Header 9: (0) Jun 21 18:00:01 VERBOSE[25160] logger.c: --- (9 headers 0 lines)Jun 21 18:00:01 VERBOSE[25160] logger.c: --- (9 headers 0 lines)--- Jun 21 18:00:01 DEBUG[25160] chan_sip.c: = Found Their Call ID: d7fef62e-79c1d760-3e4cde55@172.16.10.112 Their Tag 274E1422-E9CA97BF Our tag: as1cfc2349 Jun 21 18:00:01 DEBUG[25160] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #35 Jun 21 18:00:01 DEBUG[25160] chan_sip.c: Stopping retransmission on 'd7fef62e-79c1d760-3e4cde55@172.16.10.112' of Request 102: Match Found Jun 21 18:00:01 VERBOSE[25160] logger.c: Destroying call 'd7fef62e-79c1d760-3e4cde55@172.16.10.112' Jun 21 18:00:06 DEBUG[25157] chan_iax2.c: Allocate call number Jun 21 18:00:06 DEBUG[25157] chan_iax2.c: Registration created on call 7 Jun 21 18:00:07 VERBOSE[25160] 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=z9hG4bKR443AGMIQ17CJcUA Max-Forwards: 70 User-Agent: SOYO.SIP V1.49.002 CFG0 From: "3050" ;tag=dMcJkgjfMV4RpGyS To: "3097" Call-ID: QKXGJdVPDLiqjXYU@172.16.10.124 Contact: CSeq: 1 INVITE Supported: replaces Content-Type: application/sdp Content-Length: 235 v=0 o=- 18279790 13906630 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 Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 0: INVITE sip:3097@pfdesenv.planetarium.com.br SIP/2.0 (51) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKR443AGMIQ17CJcUA (66) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 2: Max-Forwards: 70 (16) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 3: User-Agent: SOYO.SIP V1.49.002 CFG0 (36) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 4: From: "3050" ;tag=dMcJkgjfMV4RpGyS (72) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 5: To: "3097" (49) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 6: Call-ID: QKXGJdVPDLiqjXYU@172.16.10.124 (39) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 7: Contact: (38) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 8: CSeq: 1 INVITE (14) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 9: Supported: replaces (19) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 10: Content-Type: application/sdp (29) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 11: Content-Length: 235 (19) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 12: (0) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: v=0 (3) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: o=- 18279790 13906630 IN IP4 172.16.10.124 (42) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: s=SIP CALL (10) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: c=IN IP4 172.16.10.124 (22) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: t=0 0 (5) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: m=audio 1722 RTP/AVP 18 4 0 8 3 (31) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: a=rtpmap:4 G723/8000 (20) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) Jun 21 18:00:07 VERBOSE[25160] logger.c: --- (12 headers 11 lines)Jun 21 18:00:07 VERBOSE[25160] logger.c: --- (12 headers 11 lines)--- Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Allocating new SIP dialog for QKXGJdVPDLiqjXYU@172.16.10.124 - INVITE (With RTP) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Begin: parsing SIP "Supported: replaces" Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Found SIP option: -replaces- Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Matched SIP option: replaces Jun 21 18:00:07 DEBUG[25160] chan_sip.c: * SIP extension value: 1 for call QKXGJdVPDLiqjXYU@172.16.10.124 Jun 21 18:00:07 VERBOSE[25160] logger.c: Using INVITE request as basis request - QKXGJdVPDLiqjXYU@172.16.10.124 Jun 21 18:00:07 VERBOSE[25160] logger.c: Sending to 172.16.10.124 : 5060 (non-NAT) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Setting NAT on RTP to 524288 Jun 21 18:00:07 VERBOSE[25160] 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=z9hG4bKR443AGMIQ17CJcUA;received=172.16.10.124 From: "3050" ;tag=dMcJkgjfMV4RpGyS To: "3097" ;tag=as5068fabb Call-ID: QKXGJdVPDLiqjXYU@172.16.10.124 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="1dd7c137" Content-Length: 0 --- Jun 21 18:00:07 DEBUG[25160] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #38 Jun 21 18:00:07 VERBOSE[25160] logger.c: Scheduling destruction of call 'QKXGJdVPDLiqjXYU@172.16.10.124' in 15000 ms Jun 21 18:00:07 VERBOSE[25160] logger.c: Found user '3050' Jun 21 18:00:07 VERBOSE[25160] 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=z9hG4bKR443AGMIQ17CJcUA Max-Forwards: 70 User-Agent: SOYO.SIP V1.49.002 CFG0 From: "3050" ;tag=dMcJkgjfMV4RpGyS To: "3097" ;tag=as5068fabb Call-ID: QKXGJdVPDLiqjXYU@172.16.10.124 Contact: CSeq: 1 ACK Content-Length: 0 Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 0: ACK sip:3097@pfdesenv.planetarium.com.br SIP/2.0 (48) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKR443AGMIQ17CJcUA (66) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 2: Max-Forwards: 70 (16) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 3: User-Agent: SOYO.SIP V1.49.002 CFG0 (36) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 4: From: "3050" ;tag=dMcJkgjfMV4RpGyS (72) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 5: To: "3097" ;tag=as5068fabb (64) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 6: Call-ID: QKXGJdVPDLiqjXYU@172.16.10.124 (39) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 7: Contact: (38) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 8: CSeq: 1 ACK (11) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 9: Content-Length: 0 (17) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 10: (0) Jun 21 18:00:07 VERBOSE[25160] logger.c: --- (10 headers 0 lines)Jun 21 18:00:07 VERBOSE[25160] logger.c: --- (10 headers 0 lines)--- Jun 21 18:00:07 DEBUG[25160] chan_sip.c: = Found Their Call ID: QKXGJdVPDLiqjXYU@172.16.10.124 Their Tag dMcJkgjfMV4RpGyS Our tag: as5068fabb Jun 21 18:00:07 DEBUG[25160] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Jun 21 18:00:07 DEBUG[25160] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #38 Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Stopping retransmission on 'QKXGJdVPDLiqjXYU@172.16.10.124' of Response 1: Match Found Jun 21 18:00:07 VERBOSE[25160] 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=z9hG4bKVCaHHu0YNX19f3Yy Max-Forwards: 70 User-Agent: SOYO.SIP V1.49.002 CFG0 From: "3050" ;tag=dMcJkgjfMV4RpGyS To: "3097" Call-ID: QKXGJdVPDLiqjXYU@172.16.10.124 Contact: Proxy-Authorization: Digest username="3050", realm="asterisk", nonce="1dd7c137", uri="sip:3097@pfdesenv.planetarium.com.br", response="57cdb5d7473e2e9f63ef0e7fb49f7928", algorithm=MD5 CSeq: 2 INVITE Supported: replaces Content-Type: application/sdp Content-Length: 235 v=0 o=- 71555858 36231518 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 Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 0: INVITE sip:3097@pfdesenv.planetarium.com.br SIP/2.0 (51) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKVCaHHu0YNX19f3Yy (66) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 2: Max-Forwards: 70 (16) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 3: User-Agent: SOYO.SIP V1.49.002 CFG0 (36) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 4: From: "3050" ;tag=dMcJkgjfMV4RpGyS (72) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 5: To: "3097" (49) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 6: Call-ID: QKXGJdVPDLiqjXYU@172.16.10.124 (39) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 7: Contact: (38) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 8: Proxy-Authorization: Digest username="3050", realm="asterisk", nonce="1dd7c137", uri="sip:3097@pfdesenv.planetarium.com.br", response="57cdb5d7473e2e9f63ef0e7fb49f7928", algorithm=MD5 (183) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 9: CSeq: 2 INVITE (14) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 10: Supported: replaces (19) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 11: Content-Type: application/sdp (29) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 12: Content-Length: 235 (19) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Header 13: (0) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: v=0 (3) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: o=- 71555858 36231518 IN IP4 172.16.10.124 (42) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: s=SIP CALL (10) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: c=IN IP4 172.16.10.124 (22) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: t=0 0 (5) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: m=audio 1722 RTP/AVP 18 4 0 8 3 (31) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: a=rtpmap:4 G723/8000 (20) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) Jun 21 18:00:07 VERBOSE[25160] logger.c: --- (13 headers 11 lines)Jun 21 18:00:07 VERBOSE[25160] logger.c: --- (13 headers 11 lines)--- Jun 21 18:00:07 DEBUG[25160] chan_sip.c: = Found Their Call ID: QKXGJdVPDLiqjXYU@172.16.10.124 Their Tag dMcJkgjfMV4RpGyS Our tag: as5068fabb Jun 21 18:00:07 DEBUG[25160] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Jun 21 18:00:07 VERBOSE[25160] logger.c: Using INVITE request as basis request - QKXGJdVPDLiqjXYU@172.16.10.124 Jun 21 18:00:07 VERBOSE[25160] logger.c: Sending to 172.16.10.124 : 5060 (NAT) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Setting NAT on RTP to 524288 Jun 21 18:00:07 VERBOSE[25160] logger.c: Found user '3050' Jun 21 18:00:07 VERBOSE[25160] logger.c: Found RTP audio format 18 Jun 21 18:00:07 VERBOSE[25160] logger.c: Found RTP audio format 4 Jun 21 18:00:07 VERBOSE[25160] logger.c: Found RTP audio format 0 Jun 21 18:00:07 VERBOSE[25160] logger.c: Found RTP audio format 8 Jun 21 18:00:07 VERBOSE[25160] logger.c: Found RTP audio format 3 Jun 21 18:00:07 VERBOSE[25160] logger.c: Peer audio RTP is at port 172.16.10.124:1722 Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Peer audio RTP is at port 172.16.10.124:1722 Jun 21 18:00:07 VERBOSE[25160] logger.c: Found description format G729 Jun 21 18:00:07 VERBOSE[25160] logger.c: Found description format G723 Jun 21 18:00:07 VERBOSE[25160] logger.c: Found description format PCMU Jun 21 18:00:07 VERBOSE[25160] logger.c: Found description format PCMA Jun 21 18:00:07 VERBOSE[25160] logger.c: Found description format GSM Jun 21 18:00:07 VERBOSE[25160] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x8 (alaw) Jun 21 18:00:07 VERBOSE[25160] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Checking SIP call limits for device 3050 Jun 21 18:00:07 DEBUG[25160] chan_sip.c: Updating call counter for incoming call Jun 21 18:00:07 VERBOSE[25160] logger.c: Looking for 3097 in 3050_aux (domain pfdesenv.planetarium.com.br) Jun 21 18:00:07 DEBUG[25160] chan_sip.c: build_route: Contact hop: Jun 21 18:00:07 VERBOSE[25160] logger.c: list_route: hop: Jun 21 18:00:07 VERBOSE[25160] 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=z9hG4bKVCaHHu0YNX19f3Yy;received=172.16.10.124 From: "3050" ;tag=dMcJkgjfMV4RpGyS To: "3097" Call-ID: QKXGJdVPDLiqjXYU@172.16.10.124 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jun 21 18:00:07 DEBUG[25145] chan_sip.c: Checking device state for peer 3050 Jun 21 18:00:07 DEBUG[25145] devicestate.c: Changing state for SIP/3050 - state 2 (In use) Jun 21 18:00:07 DEBUG[25246] pbx.c: Expression result is '0' Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'GotoIf' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing GotoIf("SIP/3050-c564", "0 ? 1000") in new stack Jun 21 18:00:07 DEBUG[25246] pbx.c: Not taking any branch Jun 21 18:00:07 DEBUG[25246] pbx.c: Expression result is '1' Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'GotoIf' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing GotoIf("SIP/3050-c564", "1 ? 200:400") in new stack Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Goto (3050_aux,3097,200) Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'DBget' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing DBget("SIP/3050-c564", "ramalbloqueado=BLOQUEIORAMAL/3050") in new stack Jun 21 18:00:07 VERBOSE[25246] logger.c: -- DBget: varname=ramalbloqueado, family=BLOQUEIORAMAL, key=3050 Jun 21 18:00:07 DEBUG[25246] db.c: Unable to find key '3050' in family 'BLOQUEIORAMAL' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- DBget: Value not found in database. Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'Goto' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing Goto("SIP/3050-c564", "400") in new stack Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Goto (3050_aux,3097,400) Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'SetVar' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing SetVar("SIP/3050-c564", "NUMERODISCADO=3097") in new stack Jun 21 18:00:07 DEBUG[25246] pbx.c: Expression result is '0' Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'GotoIf' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing GotoIf("SIP/3050-c564", "0 ? 404") in new stack Jun 21 18:00:07 DEBUG[25246] pbx.c: Not taking any branch Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'SetVar' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing SetVar("SIP/3050-c564", "TRANSFER_CONTEXT=3050_aux") in new stack Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'Set' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing Set("SIP/3050-c564", "GROUP=3050") in new stack Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'SetVar' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing SetVar("SIP/3050-c564", "LIMIT_WARNING_FILE=beep") in new stack Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'SetVar' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing SetVar("SIP/3050-c564", "LIMIT_TIMEOUT_FILE=beep5") in new stack Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'Goto' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing Goto("SIP/3050-c564", "500") in new stack Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Goto (3050_aux,3097,500) Jun 21 18:00:07 DEBUG[25246] pbx.c: Expression result is '0' Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'GotoIf' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing GotoIf("SIP/3050-c564", "0 ? 700") in new stack Jun 21 18:00:07 DEBUG[25246] pbx.c: Not taking any branch Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'DBget' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing DBget("SIP/3050-c564", "GRAVACAOCHAMADASAIDA=GRAVACAOCHAMADASAIDA/3050") in new stack Jun 21 18:00:07 VERBOSE[25246] logger.c: -- DBget: varname=GRAVACAOCHAMADASAIDA, family=GRAVACAOCHAMADASAIDA, key=3050 Jun 21 18:00:07 DEBUG[25246] db.c: Unable to find key '3050' in family 'GRAVACAOCHAMADASAIDA' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- DBget: Value not found in database. Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'Goto' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing Goto("SIP/3050-c564", "700") in new stack Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Goto (3050_aux,3097,700) Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'Goto' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing Goto("SIP/3050-c564", "3050_out|3097|1") in new stack Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Goto (3050_out,3097,1) Jun 21 18:00:07 DEBUG[25246] pbx.c: Expression result is '1' Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'GotoIf' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing GotoIf("SIP/3050-c564", "1 ? 2:5") in new stack Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Goto (3050_out,3097,2) Jun 21 18:00:07 DEBUG[25246] pbx.c: Expression result is '1' Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'GotoIf' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing GotoIf("SIP/3050-c564", "1 ? 3:5") in new stack Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Goto (3050_out,3097,3) Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'SetVar' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing SetVar("SIP/3050-c564", "CHAMADAINTERNA=T") in new stack Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'SetCIDNum' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing SetCIDNum("SIP/3050-c564", "3050") in new stack Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'Goto' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing Goto("SIP/3050-c564", "disca|3097|1") in new stack Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Goto (disca,3097,1) Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'Goto' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing Goto("SIP/3050-c564", "fila_desenvolvimento|3097|1") in new stack Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Goto (fila_desenvolvimento,3097,1) Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'SetLanguage' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing SetLanguage("SIP/3050-c564", "br") in new stack Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'Goto' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing Goto("SIP/3050-c564", "3") in new stack Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Goto (fila_desenvolvimento,3097,3) Jun 21 18:00:07 DEBUG[25246] pbx.c: Launching 'Wait' Jun 21 18:00:07 VERBOSE[25246] logger.c: -- Executing Wait("SIP/3050-c564", "1") in new stack Jun 21 18:00:07 DEBUG[25247] app_queue.c: Device 'SIP/3050' changed to state '2' (In use) Jun 21 18:00:08 DEBUG[25246] pbx.c: Launching 'SetCIDName' Jun 21 18:00:08 VERBOSE[25246] logger.c: -- Executing SetCIDName("SIP/3050-c564", "Fila:desenvolvimento:Ramal Teste") in new stack Jun 21 18:00:08 DEBUG[25246] pbx.c: Launching 'SetVar' Jun 21 18:00:08 VERBOSE[25246] logger.c: -- Executing SetVar("SIP/3050-c564", "NOMEFILA=desenvolvimento") in new stack Jun 21 18:00:08 DEBUG[25246] pbx.c: Launching 'SetVar' Jun 21 18:00:08 VERBOSE[25246] logger.c: -- Executing SetVar("SIP/3050-c564", "MONITOR_FILENAME=1150923607.7-filadesenvolvimento-3097-in-20060621-180008-3050") in new stack Jun 21 18:00:08 DEBUG[25246] pbx.c: Launching 'Goto' Jun 21 18:00:08 VERBOSE[25246] logger.c: -- Executing Goto("SIP/3050-c564", "8") in new stack Jun 21 18:00:08 VERBOSE[25246] logger.c: -- Goto (fila_desenvolvimento,3097,8) Jun 21 18:00:08 DEBUG[25246] pbx.c: Launching 'SetVar' Jun 21 18:00:08 VERBOSE[25246] logger.c: -- Executing SetVar("SIP/3050-c564", "DATAENTRADAFILA=1150923608") in new stack Jun 21 18:00:08 DEBUG[25246] pbx.c: Launching 'SetVar' Jun 21 18:00:08 VERBOSE[25246] logger.c: -- Executing SetVar("SIP/3050-c564", "NUMEROCLIENTE=666") in new stack Jun 21 18:00:08 DEBUG[25246] pbx.c: Launching 'SetVar' Jun 21 18:00:08 VERBOSE[25246] logger.c: -- Executing SetVar("SIP/3050-c564", "IDTSOLICITANTE=C") in new stack Jun 21 18:00:08 DEBUG[25246] pbx.c: Launching 'SetVar' Jun 21 18:00:08 VERBOSE[25246] logger.c: -- Executing SetVar("SIP/3050-c564", "NUMEROTELEFCHAMADO=3097") in new stack Jun 21 18:00:08 DEBUG[25246] pbx.c: Launching 'Queue' Jun 21 18:00:08 VERBOSE[25246] logger.c: -- Executing Queue("SIP/3050-c564", "desenvolvimento|tr|||600") in new stack Jun 21 18:00:08 DEBUG[25246] app_queue.c: NO QUEUE_PRIO variable found. Using default. Jun 21 18:00:08 DEBUG[25246] app_queue.c: queue: desenvolvimento, options: tr, url: , announce: , expires: 1150924208, priority: 0 Jun 21 18:00:08 DEBUG[25246] app_queue.c: Queue 'desenvolvimento' Join, Channel 'SIP/3050-c564', Position '1' Jun 21 18:00:08 VERBOSE[25246] 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=z9hG4bKVCaHHu0YNX19f3Yy;received=172.16.10.124 From: "3050" ;tag=dMcJkgjfMV4RpGyS To: "3097" ;tag=as285a4727 Call-ID: QKXGJdVPDLiqjXYU@172.16.10.124 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jun 21 18:00:08 DEBUG[25246] app_queue.c: It's our turn (SIP/3050-c564). Jun 21 18:00:08 DEBUG[25246] app_queue.c: SIP/3050-c564 is trying to call a queue member. Jun 21 18:00:08 DEBUG[25246] app_queue.c: Queue with URL=_ Jun 21 18:00:08 DEBUG[25246] app_queue.c: Queue with URL=_ Jun 21 18:00:08 DEBUG[25246] app_queue.c: Queue with URL=_ Jun 21 18:00:08 DEBUG[25246] app_queue.c: Queue with URL=_ Jun 21 18:00:08 DEBUG[25246] app_queue.c: Queue with URL=_ Jun 21 18:00:08 DEBUG[25246] app_queue.c: Queue with URL=_ Jun 21 18:00:08 DEBUG[25246] app_queue.c: Trying 'Agent/8003' with metric 0 Jun 21 18:00:08 DEBUG[25246] app_queue.c: Trying 'Agent/3051' with metric 0 Jun 21 18:00:08 DEBUG[25246] app_queue.c: Trying 'Agent/3050' with metric 0 Jun 21 18:00:08 DEBUG[25246] app_queue.c: Trying 'Agent/3052' with metric 0 Jun 21 18:00:08 DEBUG[25246] app_queue.c: Trying 'Agent/3054' with metric 0 Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-12. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable NUMEROTELEFCHAMADO. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-11. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable IDTSOLICITANTE. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-10. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable NUMEROCLIENTE. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-9. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable DATAENTRADAFILA. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-8. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-7. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable MONITOR_FILENAME. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-6. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable NOMEFILA. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-5. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-4. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-3. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-2. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-fila_desenvolvimento-3097-1. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-disca-3097-1. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_out-3097-5. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_out-3097-4. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable CHAMADAINTERNA. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_out-3097-3. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_out-3097-2. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_out-3097-1. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_aux-3097-700. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_aux-3097-602. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable DBGETSTATUS. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_aux-3097-501. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_aux-3097-500. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_aux-3097-406. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable LIMIT_TIMEOUT_FILE. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_aux-3097-405. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable LIMIT_WARNING_FILE. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_aux-3097-404. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable GROUP. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_aux-3097-403. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable TRANSFER_CONTEXT. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_aux-3097-402. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_aux-3097-401. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable NUMERODISCADO. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_aux-3097-400. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_aux-3097-301. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_aux-3097-200. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_aux-3097-2. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable STACK-3050_aux-3097-1. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable SIPCALLID. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable SIPUSERAGENT. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable SIPDOMAIN. Jun 21 18:00:08 DEBUG[25246] channel.c: Not copying variable SIPURI. Jun 21 18:00:08 VERBOSE[25246] logger.c: -- outgoing agentcall, to agent '3054', on 'Local/3054@dac_suporte-e480,1' Jun 21 18:00:08 VERBOSE[25246] logger.c: -- Called Agent/3054 Jun 21 18:00:08 DEBUG[25145] devicestate.c: Changing state for Local/3054@dac_suporte - state 2 (In use) Jun 21 18:00:08 DEBUG[25248] pbx.c: Launching 'Set' Jun 21 18:00:08 VERBOSE[25248] logger.c: -- Executing Set("Local/3054@dac_suporte-e480,2", "GROUP=3054") in new stack Jun 21 18:00:08 DEBUG[25248] pbx.c: Function result is '1' Jun 21 18:00:08 DEBUG[25248] pbx.c: Launching 'NoOp' Jun 21 18:00:08 VERBOSE[25248] logger.c: -- Executing NoOp("Local/3054@dac_suporte-e480,2", "Group: 3054 Count: 1") in new stack Jun 21 18:00:08 DEBUG[25248] pbx.c: Function result is '1' Jun 21 18:00:08 DEBUG[25248] pbx.c: Expression result is '0' Jun 21 18:00:08 DEBUG[25248] pbx.c: Launching 'GotoIf' Jun 21 18:00:08 VERBOSE[25248] logger.c: -- Executing GotoIf("Local/3054@dac_suporte-e480,2", "0 ? 1000") in new stack Jun 21 18:00:08 DEBUG[25248] pbx.c: Not taking any branch Jun 21 18:00:08 DEBUG[25248] pbx.c: Launching 'Goto' Jun 21 18:00:08 VERBOSE[25248] logger.c: -- Executing Goto("Local/3054@dac_suporte-e480,2", "200") in new stack Jun 21 18:00:08 VERBOSE[25248] logger.c: -- Goto (dac_suporte,3054,200) Jun 21 18:00:08 DEBUG[25248] pbx.c: Launching 'SetVar' Jun 21 18:00:08 VERBOSE[25248] logger.c: -- Executing SetVar("Local/3054@dac_suporte-e480,2", "CHAMADAINTERNA=T") in new stack Jun 21 18:00:08 DEBUG[25248] pbx.c: Launching 'Dial' Jun 21 18:00:08 VERBOSE[25248] logger.c: -- Executing Dial("Local/3054@dac_suporte-e480,2", "SIP/3054|15|T") in new stack Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Setting NAT on RTP to 524288 Jun 21 18:00:08 DEBUG[25248] channel.c: Not copying variable STACK-dac_suporte-3054-201. Jun 21 18:00:08 DEBUG[25248] channel.c: Not copying variable CHAMADAINTERNA. Jun 21 18:00:08 DEBUG[25248] channel.c: Not copying variable STACK-dac_suporte-3054-200. Jun 21 18:00:08 DEBUG[25248] channel.c: Not copying variable STACK-dac_suporte-3054-4. Jun 21 18:00:08 DEBUG[25248] channel.c: Not copying variable STACK-dac_suporte-3054-3. Jun 21 18:00:08 DEBUG[25248] channel.c: Not copying variable STACK-dac_suporte-3054-2. Jun 21 18:00:08 DEBUG[25248] channel.c: Not copying variable GROUP. Jun 21 18:00:08 DEBUG[25248] channel.c: Not copying variable STACK-dac_suporte-3054-1. Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Outgoing Call for 3054 Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Updating call counter for outgoing call Jun 21 18:00:08 VERBOSE[25248] logger.c: We're at 200.196.44.45 port 15528 Jun 21 18:00:08 VERBOSE[25248] logger.c: Adding codec 0x8 (alaw) to SDP Jun 21 18:00:08 VERBOSE[25248] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Header 0: INVITE sip:3054@172.16.10.130;user=phone SIP/2.0 (48) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK31f1d14e;rport (64) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Header 2: From: "Fila:desenvolvimento:Ramal Teste" ;tag=as2f5e57fe (80) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Header 3: To: (39) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Header 4: Contact: (33) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Header 5: Call-ID: 7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45 (55) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Header 6: CSeq: 102 INVITE (16) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Header 8: Max-Forwards: 70 (16) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Header 9: Date: Wed, 21 Jun 2006 21:00:08 GMT (35) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Header 11: Content-Type: application/sdp (29) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Header 12: Content-Length: 218 (19) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Header 13: (0) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Line: v=0 (3) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Line: o=root 25128 25128 IN IP4 200.196.44.45 (39) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Line: s=session (9) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Line: c=IN IP4 200.196.44.45 (22) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Line: t=0 0 (5) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Line: m=audio 15528 RTP/AVP 8 101 (27) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Line: a=fmtp:101 0-16 (15) Jun 21 18:00:08 DEBUG[25248] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Jun 21 18:00:08 VERBOSE[25248] logger.c: 13 headers, 10 lines Jun 21 18:00:08 VERBOSE[25248] logger.c: Reliably Transmitting (NAT) to 172.16.10.130:5060: INVITE sip:3054@172.16.10.130;user=phone SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK31f1d14e;rport From: "Fila:desenvolvimento:Ramal Teste" ;tag=as2f5e57fe To: Contact: Call-ID: 7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Jun 2006 21:00:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 218 v=0 o=root 25128 25128 IN IP4 200.196.44.45 s=session c=IN IP4 200.196.44.45 t=0 0 m=audio 15528 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jun 21 18:00:08 DEBUG[25248] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #40 Jun 21 18:00:08 VERBOSE[25248] logger.c: -- Called 3054 Jun 21 18:00:08 DEBUG[25248] channel.c: Set channel SIP/3054-ccb2 to read format alaw Jun 21 18:00:08 DEBUG[25248] channel.c: Set channel Local/3054@dac_suporte-e480,2 to write format alaw Jun 21 18:00:08 DEBUG[25248] channel.c: Set channel Local/3054@dac_suporte-e480,2 to read format alaw Jun 21 18:00:08 DEBUG[25248] channel.c: Set channel SIP/3054-ccb2 to write format alaw Jun 21 18:00:08 DEBUG[25249] app_queue.c: Device 'Local/3054@dac_suporte' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jun 21 18:00:08 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.130:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK31f1d14e;rport From: "Fila:desenvolvimento:Ramal Teste" ;tag=as2f5e57fe To: Call-ID: 7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45 CSeq: 102 INVITE User-Agent: Grandstream BT100 1.0.6.7 Content-Length: 0 Jun 21 18:00:08 DEBUG[25160] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) Jun 21 18:00:08 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK31f1d14e;rport (64) Jun 21 18:00:08 DEBUG[25160] chan_sip.c: Header 2: From: "Fila:desenvolvimento:Ramal Teste" ;tag=as2f5e57fe (80) Jun 21 18:00:08 DEBUG[25160] chan_sip.c: Header 3: To: (39) Jun 21 18:00:08 DEBUG[25160] chan_sip.c: Header 4: Call-ID: 7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45 (55) Jun 21 18:00:08 DEBUG[25160] chan_sip.c: Header 5: CSeq: 102 INVITE (16) Jun 21 18:00:08 DEBUG[25160] chan_sip.c: Header 6: User-Agent: Grandstream BT100 1.0.6.7 (37) Jun 21 18:00:08 DEBUG[25160] chan_sip.c: Header 7: Content-Length: 0 (17) Jun 21 18:00:08 DEBUG[25160] chan_sip.c: Header 8: (0) Jun 21 18:00:08 VERBOSE[25160] logger.c: --- (8 headers 0 lines)Jun 21 18:00:08 VERBOSE[25160] logger.c: --- (8 headers 0 lines)--- Jun 21 18:00:08 DEBUG[25160] chan_sip.c: = Found Their Call ID: 7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45 Their Tag Our tag: as2f5e57fe Jun 21 18:00:08 DEBUG[25160] chan_sip.c: *** SIP TIMER: Cancelling retransmission #40 - INVITE (got response) Jun 21 18:00:08 DEBUG[25160] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45' Request 102: Found Jun 21 18:00:08 DEBUG[25160] chan_sip.c: SIP response 100 to standard invite Jun 21 18:00:08 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.130:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK31f1d14e;rport From: "Fila:desenvolvimento:Ramal Teste" ;tag=as2f5e57fe To: ;tag=b6fafcf71211ee0e Call-ID: 7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45 CSeq: 102 INVITE User-Agent: Grandstream BT100 1.0.6.7 Content-Length: 0 Jun 21 18:00:08 DEBUG[25160] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) Jun 21 18:00:08 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK31f1d14e;rport (64) Jun 21 18:00:08 DEBUG[25160] chan_sip.c: Header 2: From: "Fila:desenvolvimento:Ramal Teste" ;tag=as2f5e57fe (80) Jun 21 18:00:08 DEBUG[25160] chan_sip.c: Header 3: To: ;tag=b6fafcf71211ee0e (60) Jun 21 18:00:08 DEBUG[25160] chan_sip.c: Header 4: Call-ID: 7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45 (55) Jun 21 18:00:08 DEBUG[25160] chan_sip.c: Header 5: CSeq: 102 INVITE (16) Jun 21 18:00:08 DEBUG[25160] chan_sip.c: Header 6: User-Agent: Grandstream BT100 1.0.6.7 (37) Jun 21 18:00:08 DEBUG[25160] chan_sip.c: Header 7: Content-Length: 0 (17) Jun 21 18:00:08 DEBUG[25160] chan_sip.c: Header 8: (0) Jun 21 18:00:08 VERBOSE[25160] logger.c: --- (8 headers 0 lines)Jun 21 18:00:08 VERBOSE[25160] logger.c: --- (8 headers 0 lines)--- Jun 21 18:00:08 DEBUG[25160] chan_sip.c: = Found Their Call ID: 7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45 Their Tag Our tag: as2f5e57fe Jun 21 18:00:08 DEBUG[25160] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45' Request 102: Found Jun 21 18:00:08 DEBUG[25160] chan_sip.c: SIP response 180 to standard invite Jun 21 18:00:08 VERBOSE[25248] logger.c: -- SIP/3054-ccb2 is ringing Jun 21 18:00:08 DEBUG[25145] chan_sip.c: Checking device state for peer 3054 Jun 21 18:00:08 DEBUG[25145] devicestate.c: Changing state for SIP/3054 - state 6 (Ringing) Jun 21 18:00:08 VERBOSE[25246] logger.c: -- Agent/3054 is ringing Jun 21 18:00:08 DEBUG[25250] app_queue.c: Device 'SIP/3054' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. Jun 21 18:00:10 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.130:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK31f1d14e;rport From: "Fila:desenvolvimento:Ramal Teste" ;tag=as2f5e57fe To: ;tag=b6fafcf71211ee0e Call-ID: 7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45 CSeq: 102 INVITE User-Agent: Grandstream BT100 1.0.6.7 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Supported: replaces Content-Length: 158 v=0 o=3054 8000 8000 IN IP4 172.16.10.130 s=SIP Call c=IN IP4 172.16.10.130 t=0 0 m=audio 10000 RTP/AVP 8 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK31f1d14e;rport (64) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 2: From: "Fila:desenvolvimento:Ramal Teste" ;tag=as2f5e57fe (80) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 3: To: ;tag=b6fafcf71211ee0e (60) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 4: Call-ID: 7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45 (55) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 5: CSeq: 102 INVITE (16) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 6: User-Agent: Grandstream BT100 1.0.6.7 (37) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 7: Contact: (44) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 9: Content-Type: application/sdp (29) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 10: Supported: replaces (19) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 11: Content-Length: 158 (19) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 12: (0) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Line: v=0 (3) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Line: o=3054 8000 8000 IN IP4 172.16.10.130 (37) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Line: s=SIP Call (10) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Line: c=IN IP4 172.16.10.130 (22) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Line: t=0 0 (5) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Line: m=audio 10000 RTP/AVP 8 (23) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Line: a=sendrecv (10) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Line: a=ptime:20 (10) Jun 21 18:00:10 VERBOSE[25160] logger.c: --- (12 headers 9 lines)Jun 21 18:00:10 VERBOSE[25160] logger.c: --- (12 headers 9 lines)--- Jun 21 18:00:10 DEBUG[25160] chan_sip.c: = Found Their Call ID: 7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45 Their Tag b6fafcf71211ee0e Our tag: as2f5e57fe Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Acked pending invite 102 Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Stopping retransmission on '7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45' of Request 102: Match Found Jun 21 18:00:10 DEBUG[25160] chan_sip.c: SIP response 200 to standard invite Jun 21 18:00:10 VERBOSE[25160] logger.c: Found RTP audio format 8 Jun 21 18:00:10 VERBOSE[25160] logger.c: Peer audio RTP is at port 172.16.10.130:10000 Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Peer audio RTP is at port 172.16.10.130:10000 Jun 21 18:00:10 VERBOSE[25160] logger.c: Found description format PCMA Jun 21 18:00:10 VERBOSE[25160] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Jun 21 18:00:10 VERBOSE[25160] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: build_route: Contact hop: Jun 21 18:00:10 VERBOSE[25160] logger.c: list_route: hop: Jun 21 18:00:10 VERBOSE[25160] logger.c: set_destination: Parsing for address/port to send to Jun 21 18:00:10 VERBOSE[25160] logger.c: set_destination: set destination to 172.16.10.130, port 5060 Jun 21 18:00:10 VERBOSE[25160] logger.c: Transmitting (NAT) to 172.16.10.130:5060: ACK sip:3054@172.16.10.130;user=phone SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK04dc5adf;rport From: "Fila:desenvolvimento:Ramal Teste" ;tag=as2f5e57fe To: ;tag=b6fafcf71211ee0e Contact: Call-ID: 7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jun 21 18:00:10 VERBOSE[25248] logger.c: -- SIP/3054-ccb2 answered Local/3054@dac_suporte-e480,2 Jun 21 18:00:10 DEBUG[25248] channel.c: Set channel Local/3054@dac_suporte-e480,2 to read format alaw Jun 21 18:00:10 DEBUG[25248] channel.c: Set channel SIP/3054-ccb2 to write format alaw Jun 21 18:00:10 DEBUG[25248] channel.c: Set channel SIP/3054-ccb2 to read format alaw Jun 21 18:00:10 DEBUG[25248] channel.c: Set channel Local/3054@dac_suporte-e480,2 to write format alaw Jun 21 18:00:10 DEBUG[25145] chan_sip.c: Checking device state for peer 3054 Jun 21 18:00:10 DEBUG[25145] devicestate.c: Changing state for SIP/3054 - state 2 (In use) Jun 21 18:00:10 DEBUG[25145] devicestate.c: Changing state for Local/3054@dac_suporte - state 2 (In use) Jun 21 18:00:10 DEBUG[25246] app_queue.c: Dunno what to do with control type -1 Jun 21 18:00:10 VERBOSE[25246] logger.c: -- Agent/3054 answered SIP/3050-c564 Jun 21 18:00:10 DEBUG[25246] channel.c: Set channel SIP/3050-c564 to read format alaw Jun 21 18:00:10 DEBUG[25246] channel.c: Set channel Agent/3054 to write format alaw Jun 21 18:00:10 DEBUG[25246] channel.c: Set channel Agent/3054 to read format alaw Jun 21 18:00:10 DEBUG[25246] channel.c: Set channel SIP/3050-c564 to write format alaw Jun 21 18:00:10 DEBUG[25246] chan_sip.c: sip_answer(SIP/3050-c564) Jun 21 18:00:10 VERBOSE[25246] logger.c: We're at 200.196.44.45 port 11670 Jun 21 18:00:10 VERBOSE[25246] logger.c: Adding codec 0x8 (alaw) to SDP Jun 21 18:00:10 VERBOSE[25246] 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=z9hG4bKVCaHHu0YNX19f3Yy;received=172.16.10.124 From: "3050" ;tag=dMcJkgjfMV4RpGyS To: "3097" ;tag=as285a4727 Call-ID: QKXGJdVPDLiqjXYU@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 25128 25128 IN IP4 200.196.44.45 s=session c=IN IP4 200.196.44.45 t=0 0 m=audio 11670 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- Jun 21 18:00:10 DEBUG[25246] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #42 Jun 21 18:00:10 DEBUG[25251] app_queue.c: Device 'SIP/3054' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jun 21 18:00:10 DEBUG[25252] app_queue.c: Device 'Local/3054@dac_suporte' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jun 21 18:00:10 DEBUG[25145] devicestate.c: Changing state for Local/3054@dac_suporte - state 2 (In use) Jun 21 18:00:10 DEBUG[25145] devicestate.c: Changing state for Agent/3054 - state 3 (Busy) Jun 21 18:00:10 DEBUG[25145] chan_sip.c: Checking device state for peer 3050 Jun 21 18:00:10 DEBUG[25145] devicestate.c: Changing state for SIP/3050 - state 2 (In use) Jun 21 18:00:10 DEBUG[25253] app_queue.c: Device 'Local/3054@dac_suporte' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jun 21 18:00:10 DEBUG[25254] app_queue.c: Device 'Agent/3054' changed to state '3' (Busy) Jun 21 18:00:10 DEBUG[25255] app_queue.c: Device 'SIP/3050' changed to state '2' (In use) Jun 21 18:00:10 DEBUG[25248] channel.c: Planning to masquerade channel SIP/3054-ccb2 into the structure of Local/3054@dac_suporte-e480,1 Jun 21 18:00:10 DEBUG[25248] channel.c: Done planning to masquerade channel SIP/3054-ccb2 into the structure of Local/3054@dac_suporte-e480,1 Jun 21 18:00:10 DEBUG[25248] chan_local.c: Not posting to queue since already masked on 'Local/3054@dac_suporte-e480,2' Jun 21 18:00:10 DEBUG[25246] channel.c: Actually Masquerading SIP/3054-ccb2(6) into the structure of Local/3054@dac_suporte-e480,1(6) Jun 21 18:00:10 DEBUG[25246] channel.c: Got clone lock for masquerade on 'SIP/3054-ccb2' at 0x8ea1c54 Jun 21 18:00:10 DEBUG[25246] channel.c: Set channel SIP/3054-ccb2 to write format alaw Jun 21 18:00:10 DEBUG[25246] channel.c: Set channel SIP/3054-ccb2 to read format alaw Jun 21 18:00:10 DEBUG[25246] channel.c: Putting channel SIP/3054-ccb2 in 8/8 formats Jun 21 18:00:10 DEBUG[25246] channel.c: Released clone lock on 'Local/3054@dac_suporte-e480,1' Jun 21 18:00:10 DEBUG[25246] channel.c: Done Masquerading SIP/3054-ccb2 (6) Jun 21 18:00:10 DEBUG[25246] chan_agent.c: Bridge on 'SIP/3054-ccb2' being set to 'Agent/3054' (3) Jun 21 18:00:10 DEBUG[25248] channel.c: Didn't get a frame from channel: Local/3054@dac_suporte-e480,1 Jun 21 18:00:10 DEBUG[25248] channel.c: Bridge stops bridging channels Local/3054@dac_suporte-e480,2 and Local/3054@dac_suporte-e480,1 Jun 21 18:00:10 DEBUG[25248] channel.c: Hanging up zombie 'Local/3054@dac_suporte-e480,1' Jun 21 18:00:10 DEBUG[25248] app_dial.c: Exiting with DIALSTATUS=ANSWER. Jun 21 18:00:10 DEBUG[25248] pbx.c: Spawn extension (dac_suporte,3054,201) exited non-zero on 'Local/3054@dac_suporte-e480,2' Jun 21 18:00:10 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3054@dac_suporte-e480,2' Jun 21 18:00:10 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3054@dac_suporte-e480,2' Jun 21 18:00:10 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3054@dac_suporte-e480,2' Jun 21 18:00:10 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3054@dac_suporte-e480,2' Jun 21 18:00:10 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3054@dac_suporte-e480,2' Jun 21 18:00:10 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3054@dac_suporte-e480,2' Jun 21 18:00:10 DEBUG[25246] rtp.c: Ooh, format changed from unknown to alaw Jun 21 18:00:10 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3054@dac_suporte-e480,2' Jun 21 18:00:10 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3054@dac_suporte-e480,2' Jun 21 18:00:10 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3054@dac_suporte-e480,2' Jun 21 18:00:10 DEBUG[25145] channel.c: Avoiding initial deadlock for 'Local/3054@dac_suporte-e480,2' Jun 21 18:00:10 WARNING[25145] channel.c: Avoided initial deadlock for '0x8e96148', 10 retries! Jun 21 18:00:10 DEBUG[25145] devicestate.c: Changing state for Local/3054@dac_suporte - state 0 (Unknown) Jun 21 18:00:10 DEBUG[25256] app_queue.c: Device 'Local/3054@dac_suporte' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. Jun 21 18:00:10 DEBUG[25248] channel.c: Hanging up channel 'Local/3054@dac_suporte-e480,2' Jun 21 18:00:10 DEBUG[25145] devicestate.c: Changing state for Local/3054@dac_suporte - state 0 (Unknown) Jun 21 18:00:10 DEBUG[25257] app_queue.c: Device 'Local/3054@dac_suporte' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. Jun 21 18:00:10 VERBOSE[25160] 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=z9hG4bKNQOq9mrOYqUGp14o Max-Forwards: 70 User-Agent: SOYO.SIP V1.49.002 CFG0 From: "3050" ;tag=dMcJkgjfMV4RpGyS To: "3097" ;tag=as285a4727 Call-ID: QKXGJdVPDLiqjXYU@172.16.10.124 Contact: Proxy-Authorization: Digest username="3050", realm="asterisk", nonce="1dd7c137", uri="sip:3097@200.196.44.45", response="fbfc9a1996fc1014e5cca97951540a7d", algorithm=MD5 CSeq: 2 ACK Content-Length: 0 Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 0: ACK sip:3097@200.196.44.45 SIP/2.0 (34) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKNQOq9mrOYqUGp14o (66) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 2: Max-Forwards: 70 (16) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 3: User-Agent: SOYO.SIP V1.49.002 CFG0 (36) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 4: From: "3050" ;tag=dMcJkgjfMV4RpGyS (72) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 5: To: "3097" ;tag=as285a4727 (64) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 6: Call-ID: QKXGJdVPDLiqjXYU@172.16.10.124 (39) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 7: Contact: (38) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 8: Proxy-Authorization: Digest username="3050", realm="asterisk", nonce="1dd7c137", uri="sip:3097@200.196.44.45", response="fbfc9a1996fc1014e5cca97951540a7d", algorithm=MD5 (169) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 9: CSeq: 2 ACK (11) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 10: Content-Length: 0 (17) Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Header 11: (0) Jun 21 18:00:10 VERBOSE[25160] logger.c: --- (11 headers 0 lines)Jun 21 18:00:10 VERBOSE[25160] logger.c: --- (11 headers 0 lines)--- Jun 21 18:00:10 DEBUG[25160] chan_sip.c: = No match Their Call ID: 7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45 Their Tag b6fafcf71211ee0e Our tag: as2f5e57fe Jun 21 18:00:10 DEBUG[25160] chan_sip.c: = Found Their Call ID: QKXGJdVPDLiqjXYU@172.16.10.124 Their Tag dMcJkgjfMV4RpGyS Our tag: as285a4727 Jun 21 18:00:10 DEBUG[25160] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Jun 21 18:00:10 DEBUG[25160] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #42 Jun 21 18:00:10 DEBUG[25160] chan_sip.c: Stopping retransmission on 'QKXGJdVPDLiqjXYU@172.16.10.124' of Response 2: Match Found Jun 21 18:00:11 DEBUG[25246] rtp.c: Ooh, format changed from unknown to alaw Jun 21 18:00:12 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.124:5060: BYE sip:3097@200.196.44.45 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKMvG9IVjyy8NlRlcy Max-Forwards: 70 User-Agent: SOYO.SIP V1.49.002 CFG0 From: "3050" ;tag=dMcJkgjfMV4RpGyS To: "3097" ;tag=as285a4727 Call-ID: QKXGJdVPDLiqjXYU@172.16.10.124 Contact: Proxy-Authorization: Digest username="3050", realm="asterisk", nonce="1dd7c137", uri="sip:3097@200.196.44.45", response="7645361bb8326c9102587a1d30d81f96", algorithm=MD5 CSeq: 3 BYE Content-Length: 0 Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 0: BYE sip:3097@200.196.44.45 SIP/2.0 (34) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKMvG9IVjyy8NlRlcy (66) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 2: Max-Forwards: 70 (16) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 3: User-Agent: SOYO.SIP V1.49.002 CFG0 (36) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 4: From: "3050" ;tag=dMcJkgjfMV4RpGyS (72) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 5: To: "3097" ;tag=as285a4727 (64) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 6: Call-ID: QKXGJdVPDLiqjXYU@172.16.10.124 (39) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 7: Contact: (38) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 8: Proxy-Authorization: Digest username="3050", realm="asterisk", nonce="1dd7c137", uri="sip:3097@200.196.44.45", response="7645361bb8326c9102587a1d30d81f96", algorithm=MD5 (169) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 9: CSeq: 3 BYE (11) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 10: Content-Length: 0 (17) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 11: (0) Jun 21 18:00:12 VERBOSE[25160] logger.c: --- (11 headers 0 lines)Jun 21 18:00:12 VERBOSE[25160] logger.c: --- (11 headers 0 lines)--- Jun 21 18:00:12 DEBUG[25160] chan_sip.c: = No match Their Call ID: 7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45 Their Tag b6fafcf71211ee0e Our tag: as2f5e57fe Jun 21 18:00:12 DEBUG[25160] chan_sip.c: = Found Their Call ID: QKXGJdVPDLiqjXYU@172.16.10.124 Their Tag dMcJkgjfMV4RpGyS Our tag: as285a4727 Jun 21 18:00:12 DEBUG[25160] chan_sip.c: **** Received BYE (8) - Command in SIP BYE Jun 21 18:00:12 VERBOSE[25160] logger.c: Sending to 172.16.10.124 : 5060 (NAT) Jun 21 18:00:12 VERBOSE[25160] logger.c: Transmitting (NAT) to 172.16.10.124:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKMvG9IVjyy8NlRlcy;received=172.16.10.124 From: "3050" ;tag=dMcJkgjfMV4RpGyS To: "3097" ;tag=as285a4727 Call-ID: QKXGJdVPDLiqjXYU@172.16.10.124 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jun 21 18:00:12 DEBUG[25246] channel.c: Didn't get a frame from channel: SIP/3050-c564 Jun 21 18:00:12 DEBUG[25246] channel.c: Bridge stops bridging channels SIP/3050-c564 and Agent/3054 Jun 21 18:00:12 DEBUG[25246] channel.c: Hanging up channel 'Agent/3054' Jun 21 18:00:12 DEBUG[25246] chan_agent.c: Hangup called for state Up Jun 21 18:00:12 DEBUG[25246] channel.c: Hanging up channel 'SIP/3054-ccb2' Jun 21 18:00:12 DEBUG[25246] chan_sip.c: Hangup call SIP/3054-ccb2, SIP callid 7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45) Jun 21 18:00:12 DEBUG[25246] chan_sip.c: update_call_counter(3054) - decrement call limit counter Jun 21 18:00:12 DEBUG[25246] chan_sip.c: Updating call counter for outgoing call Jun 21 18:00:12 VERBOSE[25246] logger.c: set_destination: Parsing for address/port to send to Jun 21 18:00:12 VERBOSE[25246] logger.c: set_destination: set destination to 172.16.10.130, port 5060 Jun 21 18:00:12 VERBOSE[25246] logger.c: Reliably Transmitting (NAT) to 172.16.10.130:5060: BYE sip:3054@172.16.10.130;user=phone SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK42dce1f0;rport From: "Fila:desenvolvimento:Ramal Teste" ;tag=as2f5e57fe To: ;tag=b6fafcf71211ee0e Contact: Call-ID: 7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jun 21 18:00:12 DEBUG[25246] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #43 Jun 21 18:00:12 DEBUG[25246] chan_agent.c: Hungup, howlong is 0, autologoff is 0 Jun 21 18:00:12 DEBUG[25246] pbx.c: Spawn extension (fila_desenvolvimento,3097,12) exited non-zero on 'SIP/3050-c564' Jun 21 18:00:12 DEBUG[25145] chan_sip.c: Checking device state for peer 3054 Jun 21 18:00:12 DEBUG[25145] devicestate.c: Changing state for SIP/3054 - state 1 (Not in use) Jun 21 18:00:12 DEBUG[25145] devicestate.c: Changing state for Agent/3054 - state 1 (Not in use) Jun 21 18:00:12 DEBUG[25145] devicestate.c: Changing state for Agent/3054 - state 1 (Not in use) Jun 21 18:00:12 DEBUG[25258] app_queue.c: Device 'SIP/3054' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Jun 21 18:00:12 DEBUG[25259] app_queue.c: Device 'Agent/3054' changed to state '1' (Not in use) Jun 21 18:00:12 DEBUG[25260] app_queue.c: Device 'Agent/3054' changed to state '1' (Not in use) Jun 21 18:00:12 DEBUG[25246] channel.c: Hanging up channel 'SIP/3050-c564' Jun 21 18:00:12 DEBUG[25246] chan_sip.c: Hangup call SIP/3050-c564, SIP callid QKXGJdVPDLiqjXYU@172.16.10.124) Jun 21 18:00:12 DEBUG[25246] chan_sip.c: update_call_counter(3050) - decrement call limit counter Jun 21 18:00:12 DEBUG[25246] chan_sip.c: Updating call counter for incoming call Jun 21 18:00:12 DEBUG[25246] res_monitor.c: monitor executing "/bin/nice -n 40 /usr/local/sbin/gc.registragravacao" "/var/spool/asterisk/monitor/1150923607.7-filadesenvolvimento-3097-in-20060621-180008-3050-in.WAV" "/var/spool/asterisk/monitor/1150923607.7-filadesenvolvimento-3097-in-20060621-180008-3050-out.WAV" "/var/spool/asterisk/monitor/1150923607.7-filadesenvolvimento-3097-in-20060621-180008-3050.WAV" & Jun 21 18:00:12 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.130:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK42dce1f0;rport From: "Fila:desenvolvimento:Ramal Teste" ;tag=as2f5e57fe To: ;tag=b6fafcf71211ee0e Call-ID: 7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45 CSeq: 103 BYE User-Agent: Grandstream BT100 1.0.6.7 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Supported: replaces Content-Length: 0 Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK42dce1f0;rport (64) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 2: From: "Fila:desenvolvimento:Ramal Teste" ;tag=as2f5e57fe (80) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 3: To: ;tag=b6fafcf71211ee0e (60) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 4: Call-ID: 7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45 (55) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 5: CSeq: 103 BYE (13) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 6: User-Agent: Grandstream BT100 1.0.6.7 (37) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 7: Contact: (44) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 9: Supported: replaces (19) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 10: Content-Length: 0 (17) Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Header 11: (0) Jun 21 18:00:12 VERBOSE[25160] logger.c: --- (11 headers 0 lines)Jun 21 18:00:12 VERBOSE[25160] logger.c: --- (11 headers 0 lines)--- Jun 21 18:00:12 DEBUG[25160] chan_sip.c: = Found Their Call ID: 7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45 Their Tag b6fafcf71211ee0e Our tag: as2f5e57fe Jun 21 18:00:12 DEBUG[25160] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #43 Jun 21 18:00:12 DEBUG[25160] chan_sip.c: Stopping retransmission on '7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45' of Request 103: Match Found Jun 21 18:00:12 VERBOSE[25160] logger.c: Destroying call '7910d70a7714aa8b7c44b9f71e91e28a@200.196.44.45' Jun 21 18:00:12 VERBOSE[25160] logger.c: Destroying call 'QKXGJdVPDLiqjXYU@172.16.10.124' Jun 21 18:00:12 DEBUG[25145] chan_sip.c: Checking device state for peer 3050 Jun 21 18:00:12 DEBUG[25145] devicestate.c: Changing state for SIP/3050 - state 1 (Not in use) Jun 21 18:00:12 DEBUG[25263] app_queue.c: Device 'SIP/3050' changed to state '1' (Not in use) Jun 21 18:00:16 VERBOSE[25160] 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=z9hG4bKHnT1d1twep0KjLSW Max-Forwards: 70 User-Agent: SOYO.SIP V1.49.002 CFG0 From: "3050" ;tag=qmw8HscnenFa5oIA To: "3097" Call-ID: biBHNMXiCtFQRNgk@172.16.10.124 Contact: CSeq: 1 INVITE Supported: replaces Content-Type: application/sdp Content-Length: 235 v=0 o=- 30258302 00482038 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 Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 0: INVITE sip:3097@pfdesenv.planetarium.com.br SIP/2.0 (51) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKHnT1d1twep0KjLSW (66) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 2: Max-Forwards: 70 (16) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 3: User-Agent: SOYO.SIP V1.49.002 CFG0 (36) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 4: From: "3050" ;tag=qmw8HscnenFa5oIA (72) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 5: To: "3097" (49) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 6: Call-ID: biBHNMXiCtFQRNgk@172.16.10.124 (39) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 7: Contact: (38) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 8: CSeq: 1 INVITE (14) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 9: Supported: replaces (19) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 10: Content-Type: application/sdp (29) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 11: Content-Length: 235 (19) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 12: (0) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Line: v=0 (3) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Line: o=- 30258302 00482038 IN IP4 172.16.10.124 (42) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Line: s=SIP CALL (10) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Line: c=IN IP4 172.16.10.124 (22) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Line: t=0 0 (5) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Line: m=audio 1722 RTP/AVP 18 4 0 8 3 (31) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Line: a=rtpmap:4 G723/8000 (20) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) Jun 21 18:00:16 VERBOSE[25160] logger.c: --- (12 headers 11 lines)Jun 21 18:00:16 VERBOSE[25160] logger.c: --- (12 headers 11 lines)--- Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Allocating new SIP dialog for biBHNMXiCtFQRNgk@172.16.10.124 - INVITE (With RTP) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Begin: parsing SIP "Supported: replaces" Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Found SIP option: -replaces- Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Matched SIP option: replaces Jun 21 18:00:16 DEBUG[25160] chan_sip.c: * SIP extension value: 1 for call biBHNMXiCtFQRNgk@172.16.10.124 Jun 21 18:00:16 VERBOSE[25160] logger.c: Using INVITE request as basis request - biBHNMXiCtFQRNgk@172.16.10.124 Jun 21 18:00:16 VERBOSE[25160] logger.c: Sending to 172.16.10.124 : 5060 (non-NAT) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Setting NAT on RTP to 524288 Jun 21 18:00:16 VERBOSE[25160] 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=z9hG4bKHnT1d1twep0KjLSW;received=172.16.10.124 From: "3050" ;tag=qmw8HscnenFa5oIA To: "3097" ;tag=as5c26153c Call-ID: biBHNMXiCtFQRNgk@172.16.10.124 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="6995c753" Content-Length: 0 --- Jun 21 18:00:16 DEBUG[25160] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #44 Jun 21 18:00:16 VERBOSE[25160] logger.c: Scheduling destruction of call 'biBHNMXiCtFQRNgk@172.16.10.124' in 15000 ms Jun 21 18:00:16 VERBOSE[25160] logger.c: Found user '3050' Jun 21 18:00:16 VERBOSE[25160] 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=z9hG4bKHnT1d1twep0KjLSW Max-Forwards: 70 User-Agent: SOYO.SIP V1.49.002 CFG0 From: "3050" ;tag=qmw8HscnenFa5oIA To: "3097" ;tag=as5c26153c Call-ID: biBHNMXiCtFQRNgk@172.16.10.124 Contact: CSeq: 1 ACK Content-Length: 0 Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 0: ACK sip:3097@pfdesenv.planetarium.com.br SIP/2.0 (48) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bKHnT1d1twep0KjLSW (66) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 2: Max-Forwards: 70 (16) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 3: User-Agent: SOYO.SIP V1.49.002 CFG0 (36) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 4: From: "3050" ;tag=qmw8HscnenFa5oIA (72) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 5: To: "3097" ;tag=as5c26153c (64) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 6: Call-ID: biBHNMXiCtFQRNgk@172.16.10.124 (39) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 7: Contact: (38) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 8: CSeq: 1 ACK (11) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 9: Content-Length: 0 (17) Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Header 10: (0) Jun 21 18:00:16 VERBOSE[25160] logger.c: --- (10 headers 0 lines)Jun 21 18:00:16 VERBOSE[25160] logger.c: --- (10 headers 0 lines)--- Jun 21 18:00:16 DEBUG[25160] chan_sip.c: = Found Their Call ID: biBHNMXiCtFQRNgk@172.16.10.124 Their Tag qmw8HscnenFa5oIA Our tag: as5c26153c Jun 21 18:00:16 DEBUG[25160] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Jun 21 18:00:16 DEBUG[25160] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #44 Jun 21 18:00:16 DEBUG[25160] chan_sip.c: Stopping retransmission on 'biBHNMXiCtFQRNgk@172.16.10.124' of Response 1: Match Found Jun 21 18:00:16 DEBUG[25157] chan_iax2.c: Peer lastms 3, historicms 3, maxms 2000 Jun 21 18:00:16 DEBUG[25157] chan_iax2.c: Peer lastms 3, historicms 3, maxms 2000 Jun 21 18:00:16 DEBUG[25157] chan_iax2.c: Peer lastms 1, historicms 1, maxms 2000 Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 0: OPTIONS sip:3054@172.16.10.130;user=phone SIP/2.0 (49) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK0ff6a5c4;rport (64) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 2: From: "asterisk" ;tag=as488aa1dd (60) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 3: To: (39) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 4: Contact: (37) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 5: Call-ID: 4157d5ea2136479470b65a9770b62de8@200.196.44.45 (55) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 8: Max-Forwards: 70 (16) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 9: Date: Wed, 21 Jun 2006 21:00:17 GMT (35) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 11: Content-Length: 0 (17) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 12: (0) Jun 21 18:00:17 VERBOSE[25160] logger.c: 12 headers, 0 lines Jun 21 18:00:17 VERBOSE[25160] logger.c: Reliably Transmitting (NAT) to 172.16.10.130:5060: OPTIONS sip:3054@172.16.10.130;user=phone SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK0ff6a5c4;rport From: "asterisk" ;tag=as488aa1dd To: Contact: Call-ID: 4157d5ea2136479470b65a9770b62de8@200.196.44.45 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Jun 2006 21:00:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Jun 21 18:00:17 DEBUG[25160] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #46 Jun 21 18:00:17 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.130:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK0ff6a5c4;rport From: "asterisk" ;tag=as488aa1dd To: ;tag=890d78559b96d3a4 Call-ID: 4157d5ea2136479470b65a9770b62de8@200.196.44.45 CSeq: 102 OPTIONS User-Agent: Grandstream BT100 1.0.6.7 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Supported: replaces Content-Length: 0 Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK0ff6a5c4;rport (64) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 2: From: "asterisk" ;tag=as488aa1dd (60) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 3: To: ;tag=890d78559b96d3a4 (60) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 4: Call-ID: 4157d5ea2136479470b65a9770b62de8@200.196.44.45 (55) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 6: User-Agent: Grandstream BT100 1.0.6.7 (37) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 7: Contact: (44) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 9: Supported: replaces (19) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 10: Content-Length: 0 (17) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 11: (0) Jun 21 18:00:17 VERBOSE[25160] logger.c: --- (11 headers 0 lines)Jun 21 18:00:17 VERBOSE[25160] logger.c: --- (11 headers 0 lines)--- Jun 21 18:00:17 DEBUG[25160] chan_sip.c: = Found Their Call ID: 4157d5ea2136479470b65a9770b62de8@200.196.44.45 Their Tag Our tag: as488aa1dd Jun 21 18:00:17 DEBUG[25160] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #46 Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Stopping retransmission on '4157d5ea2136479470b65a9770b62de8@200.196.44.45' of Request 102: Match Found Jun 21 18:00:17 VERBOSE[25160] logger.c: Destroying call '4157d5ea2136479470b65a9770b62de8@200.196.44.45' Jun 21 18:00:17 VERBOSE[25160] 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=z9hG4bK2odi6W0TXuyeUHoU Max-Forwards: 70 User-Agent: SOYO.SIP V1.49.002 CFG0 From: "3050" ;tag=qmw8HscnenFa5oIA To: "3097" Call-ID: biBHNMXiCtFQRNgk@172.16.10.124 Contact: Proxy-Authorization: Digest username="3050", realm="asterisk", nonce="6995c753", uri="sip:3097@pfdesenv.planetarium.com.br", response="f211447369cbc71117fc2ddd053a6073", algorithm=MD5 CSeq: 2 INVITE Supported: replaces Content-Type: application/sdp Content-Length: 235 v=0 o=- 93402724 07116147 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 Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 0: INVITE sip:3097@pfdesenv.planetarium.com.br SIP/2.0 (51) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.10.124:5060;branch=z9hG4bK2odi6W0TXuyeUHoU (66) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 2: Max-Forwards: 70 (16) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 3: User-Agent: SOYO.SIP V1.49.002 CFG0 (36) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 4: From: "3050" ;tag=qmw8HscnenFa5oIA (72) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 5: To: "3097" (49) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 6: Call-ID: biBHNMXiCtFQRNgk@172.16.10.124 (39) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 7: Contact: (38) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 8: Proxy-Authorization: Digest username="3050", realm="asterisk", nonce="6995c753", uri="sip:3097@pfdesenv.planetarium.com.br", response="f211447369cbc71117fc2ddd053a6073", algorithm=MD5 (183) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 9: CSeq: 2 INVITE (14) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 10: Supported: replaces (19) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 11: Content-Type: application/sdp (29) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 12: Content-Length: 235 (19) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 13: (0) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Line: v=0 (3) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Line: o=- 93402724 07116147 IN IP4 172.16.10.124 (42) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Line: s=SIP CALL (10) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Line: c=IN IP4 172.16.10.124 (22) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Line: t=0 0 (5) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Line: m=audio 1722 RTP/AVP 18 4 0 8 3 (31) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Line: a=rtpmap:4 G723/8000 (20) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) Jun 21 18:00:17 VERBOSE[25160] logger.c: --- (13 headers 11 lines)Jun 21 18:00:17 VERBOSE[25160] logger.c: --- (13 headers 11 lines)--- Jun 21 18:00:17 DEBUG[25160] chan_sip.c: = Found Their Call ID: biBHNMXiCtFQRNgk@172.16.10.124 Their Tag qmw8HscnenFa5oIA Our tag: as5c26153c Jun 21 18:00:17 DEBUG[25160] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Jun 21 18:00:17 VERBOSE[25160] logger.c: Using INVITE request as basis request - biBHNMXiCtFQRNgk@172.16.10.124 Jun 21 18:00:17 VERBOSE[25160] logger.c: Sending to 172.16.10.124 : 5060 (NAT) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Setting NAT on RTP to 524288 Jun 21 18:00:17 VERBOSE[25160] logger.c: Found user '3050' Jun 21 18:00:17 VERBOSE[25160] logger.c: Found RTP audio format 18 Jun 21 18:00:17 VERBOSE[25160] logger.c: Found RTP audio format 4 Jun 21 18:00:17 VERBOSE[25160] logger.c: Found RTP audio format 0 Jun 21 18:00:17 VERBOSE[25160] logger.c: Found RTP audio format 8 Jun 21 18:00:17 VERBOSE[25160] logger.c: Found RTP audio format 3 Jun 21 18:00:17 VERBOSE[25160] logger.c: Peer audio RTP is at port 172.16.10.124:1722 Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Peer audio RTP is at port 172.16.10.124:1722 Jun 21 18:00:17 VERBOSE[25160] logger.c: Found description format G729 Jun 21 18:00:17 VERBOSE[25160] logger.c: Found description format G723 Jun 21 18:00:17 VERBOSE[25160] logger.c: Found description format PCMU Jun 21 18:00:17 VERBOSE[25160] logger.c: Found description format PCMA Jun 21 18:00:17 VERBOSE[25160] logger.c: Found description format GSM Jun 21 18:00:17 VERBOSE[25160] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x8 (alaw) Jun 21 18:00:17 VERBOSE[25160] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Checking SIP call limits for device 3050 Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Updating call counter for incoming call Jun 21 18:00:17 VERBOSE[25160] logger.c: Looking for 3097 in 3050_aux (domain pfdesenv.planetarium.com.br) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: build_route: Contact hop: Jun 21 18:00:17 VERBOSE[25160] logger.c: list_route: hop: Jun 21 18:00:17 VERBOSE[25160] 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=z9hG4bK2odi6W0TXuyeUHoU;received=172.16.10.124 From: "3050" ;tag=qmw8HscnenFa5oIA To: "3097" Call-ID: biBHNMXiCtFQRNgk@172.16.10.124 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 0: OPTIONS sip:3050@172.16.10.124:5060 SIP/2.0 (43) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK2ae8cb3f;rport (64) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 2: From: "asterisk" ;tag=as707c168c (60) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 3: To: (33) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 4: Contact: (37) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 5: Call-ID: 4d545951081ed02172bd526f68890284@200.196.44.45 (55) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 8: Max-Forwards: 70 (16) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 9: Date: Wed, 21 Jun 2006 21:00:17 GMT (35) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 11: Content-Length: 0 (17) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 12: (0) Jun 21 18:00:17 VERBOSE[25160] logger.c: 12 headers, 0 lines Jun 21 18:00:17 VERBOSE[25160] logger.c: Reliably Transmitting (NAT) to 172.16.10.124:5060: OPTIONS sip:3050@172.16.10.124:5060 SIP/2.0 Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK2ae8cb3f;rport From: "asterisk" ;tag=as707c168c To: Contact: Call-ID: 4d545951081ed02172bd526f68890284@200.196.44.45 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Jun 2006 21:00:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Jun 21 18:00:17 DEBUG[25160] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #49 Jun 21 18:00:17 DEBUG[25145] chan_sip.c: Checking device state for peer 3050 Jun 21 18:00:17 DEBUG[25145] devicestate.c: Changing state for SIP/3050 - state 2 (In use) Jun 21 18:00:17 DEBUG[25265] pbx.c: Expression result is '0' Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'GotoIf' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing GotoIf("SIP/3050-0fb0", "0 ? 1000") in new stack Jun 21 18:00:17 DEBUG[25265] pbx.c: Not taking any branch Jun 21 18:00:17 DEBUG[25265] pbx.c: Expression result is '1' Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'GotoIf' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing GotoIf("SIP/3050-0fb0", "1 ? 200:400") in new stack Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Goto (3050_aux,3097,200) Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'DBget' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing DBget("SIP/3050-0fb0", "ramalbloqueado=BLOQUEIORAMAL/3050") in new stack Jun 21 18:00:17 VERBOSE[25265] logger.c: -- DBget: varname=ramalbloqueado, family=BLOQUEIORAMAL, key=3050 Jun 21 18:00:17 DEBUG[25265] db.c: Unable to find key '3050' in family 'BLOQUEIORAMAL' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- DBget: Value not found in database. Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'Goto' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing Goto("SIP/3050-0fb0", "400") in new stack Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Goto (3050_aux,3097,400) Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'SetVar' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing SetVar("SIP/3050-0fb0", "NUMERODISCADO=3097") in new stack Jun 21 18:00:17 DEBUG[25265] pbx.c: Expression result is '0' Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'GotoIf' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing GotoIf("SIP/3050-0fb0", "0 ? 404") in new stack Jun 21 18:00:17 DEBUG[25265] pbx.c: Not taking any branch Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'SetVar' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing SetVar("SIP/3050-0fb0", "TRANSFER_CONTEXT=3050_aux") in new stack Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'Set' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing Set("SIP/3050-0fb0", "GROUP=3050") in new stack Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'SetVar' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing SetVar("SIP/3050-0fb0", "LIMIT_WARNING_FILE=beep") in new stack Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'SetVar' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing SetVar("SIP/3050-0fb0", "LIMIT_TIMEOUT_FILE=beep5") in new stack Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'Goto' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing Goto("SIP/3050-0fb0", "500") in new stack Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Goto (3050_aux,3097,500) Jun 21 18:00:17 DEBUG[25265] pbx.c: Expression result is '0' Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'GotoIf' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing GotoIf("SIP/3050-0fb0", "0 ? 700") in new stack Jun 21 18:00:17 DEBUG[25265] pbx.c: Not taking any branch Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'DBget' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing DBget("SIP/3050-0fb0", "GRAVACAOCHAMADASAIDA=GRAVACAOCHAMADASAIDA/3050") in new stack Jun 21 18:00:17 VERBOSE[25265] logger.c: -- DBget: varname=GRAVACAOCHAMADASAIDA, family=GRAVACAOCHAMADASAIDA, key=3050 Jun 21 18:00:17 DEBUG[25265] db.c: Unable to find key '3050' in family 'GRAVACAOCHAMADASAIDA' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- DBget: Value not found in database. Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'Goto' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing Goto("SIP/3050-0fb0", "700") in new stack Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Goto (3050_aux,3097,700) Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'Goto' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing Goto("SIP/3050-0fb0", "3050_out|3097|1") in new stack Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Goto (3050_out,3097,1) Jun 21 18:00:17 DEBUG[25265] pbx.c: Expression result is '1' Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'GotoIf' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing GotoIf("SIP/3050-0fb0", "1 ? 2:5") in new stack Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Goto (3050_out,3097,2) Jun 21 18:00:17 DEBUG[25265] pbx.c: Expression result is '1' Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'GotoIf' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing GotoIf("SIP/3050-0fb0", "1 ? 3:5") in new stack Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Goto (3050_out,3097,3) Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'SetVar' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing SetVar("SIP/3050-0fb0", "CHAMADAINTERNA=T") in new stack Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'SetCIDNum' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing SetCIDNum("SIP/3050-0fb0", "3050") in new stack Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'Goto' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing Goto("SIP/3050-0fb0", "disca|3097|1") in new stack Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Goto (disca,3097,1) Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'Goto' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing Goto("SIP/3050-0fb0", "fila_desenvolvimento|3097|1") in new stack Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Goto (fila_desenvolvimento,3097,1) Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'SetLanguage' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing SetLanguage("SIP/3050-0fb0", "br") in new stack Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'Goto' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing Goto("SIP/3050-0fb0", "3") in new stack Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Goto (fila_desenvolvimento,3097,3) Jun 21 18:00:17 DEBUG[25265] pbx.c: Launching 'Wait' Jun 21 18:00:17 VERBOSE[25265] logger.c: -- Executing Wait("SIP/3050-0fb0", "1") in new stack Jun 21 18:00:17 DEBUG[25266] app_queue.c: Device 'SIP/3050' changed to state '2' (In use) Jun 21 18:00:17 VERBOSE[25160] logger.c: <-- SIP read from 172.16.10.124:5060: SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK2ae8cb3f;rport Call-ID: 4d545951081ed02172bd526f68890284@200.196.44.45 CSeq: 102 OPTIONS From: "asterisk" ;tag=as707c168c To: ;tag=heS63EzgegqN0DER Contact: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Accept: application/sdp, message/sipfrag, application/dtmf-relay Supported: replaces Content-Length: 0 Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 0: SIP/2.0 486 Busy Here (21) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 1: Via: SIP/2.0/UDP 200.196.44.45:5060;branch=z9hG4bK2ae8cb3f;rport (64) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 2: Call-ID: 4d545951081ed02172bd526f68890284@200.196.44.45 (55) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 3: CSeq: 102 OPTIONS (17) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 4: From: "asterisk" ;tag=as707c168c (60) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 5: To: ;tag=heS63EzgegqN0DER (54) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 6: Contact: (38) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 7: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE (85) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 8: Accept: application/sdp, message/sipfrag, application/dtmf-relay (64) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 9: Supported: replaces (19) Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Header 10: Content-Length: 0 (17) Jun 21 18:00:17 VERBOSE[25160] logger.c: --- (11 headers 0 lines)Jun 21 18:00:17 VERBOSE[25160] logger.c: --- (11 headers 0 lines)--- Jun 21 18:00:17 DEBUG[25160] chan_sip.c: = Found Their Call ID: 4d545951081ed02172bd526f68890284@200.196.44.45 Their Tag Our tag: as707c168c Jun 21 18:00:17 DEBUG[25160] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #49 Jun 21 18:00:17 DEBUG[25160] chan_sip.c: Stopping retransmission on '4d545951081ed02172bd526f68890284@200.196.44.45' of Request 102: Match Found Jun 21 18:00:17 VERBOSE[25160] logger.c: Destroying call '4d545951081ed02172bd526f68890284@200.196.44.45' Jun 21 18:00:18 DEBUG[25265] pbx.c: Launching 'SetCIDName' Jun 21 18:00:18 VERBOSE[25265] logger.c: -- Executing SetCIDName("SIP/3050-0fb0", "Fila:desenvolvimento:Ramal Teste") in new stack Jun 21 18:00:18 DEBUG[25265] pbx.c: Launching 'SetVar' Jun 21 18:00:18 VERBOSE[25265] logger.c: -- Executing SetVar("SIP/3050-0fb0", "NOMEFILA=desenvolvimento") in new stack Jun 21 18:00:18 DEBUG[25265] pbx.c: Launching 'SetVar' Jun 21 18:00:18 VERBOSE[25265] logger.c: -- Executing SetVar("SIP/3050-0fb0", "MONITOR_FILENAME=1150923617.12-filadesenvolvimento-3097-in-20060621-180018-3050") in new stack Jun 21 18:00:18 DEBUG[25265] pbx.c: Launching 'Goto' Jun 21 18:00:18 VERBOSE[25265] logger.c: -- Executing Goto("SIP/3050-0fb0", "8") in new stack Jun 21 18:00:18 VERBOSE[25265] logger.c: -- Goto (fila_desenvolvimento,3097,8) Jun 21 18:00:18 DEBUG[25265] pbx.c: Launching 'SetVar' Jun 21 18:00:18 VERBOSE[25265] logger.c: -- Executing SetVar("SIP/3050-0fb0", "DATAENTRADAFILA=1150923618") in new stack Jun 21 18:00:18 DEBUG[25265] pbx.c: Launching 'SetVar' Jun 21 18:00:18 VERBOSE[25265] logger.c: -- Executing SetVar("SIP/3050-0fb0", "NUMEROCLIENTE=666") in new stack Jun 21 18:00:18 DEBUG[25265] pbx.c: Launching 'SetVar' Jun 21 18:00:18 VERBOSE[25265] logger.c: -- Executing SetVar("SIP/3050-0fb0", "IDTSOLICITANTE=C") in new stack Jun 21 18:00:18 DEBUG[25265] pbx.c: Launching 'SetVar' Jun 21 18:00:18 VERBOSE[25265] logger.c: -- Executing SetVar("SIP/3050-0fb0", "NUMEROTELEFCHAMADO=3097") in new stack Jun 21 18:00:18 DEBUG[25265] pbx.c: Launching 'Queue' Jun 21 18:00:18 VERBOSE[25265] logger.c: -- Executing Queue("SIP/3050-0fb0", "desenvolvimento|tr|||600") in new stack Jun 21 18:00:18 DEBUG[25265] app_queue.c: NO QUEUE_PRIO variable found. Using default. Jun 21 18:00:18 DEBUG[25265] app_queue.c: queue: desenvolvimento, options: tr, url: , announce: , expires: 1150924218, priority: 0 Jun 21 18:00:18 DEBUG[25265] app_queue.c: Queue 'desenvolvimento' Join, Channel 'SIP/3050-0fb0', Position '1' Jun 21 18:00:18 VERBOSE[25265] 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=z9hG4bK2odi6W0TXuyeUHoU;received=172.16.10.124 From: "3050" ;tag=qmw8HscnenFa5oIA To: "3097" ;tag=as23c0b103 Call-ID: biBHNMXiCtFQRNgk@172.16.10.124 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jun 21 18:00:18 DEBUG[25265] app_queue.c: It's our turn (SIP/3050-0fb0). Jun 21 18:00:18 DEBUG[25265] app_queue.c: SIP/3050-0fb0 is trying to call a queue member. Jun 21 18:00:18 DEBUG[25265] app_queue.c: Queue with URL=_ Jun 21 18:00:18 DEBUG[25265] app_queue.c: Queue with URL=_ Jun 21 18:00:18 DEBUG[25265] app_queue.c: Queue with URL=_ Jun 21 18:00:18 DEBUG[25265] app_queue.c: Queue with URL=_ Jun 21 18:00:18 DEBUG[25265] app_queue.c: Queue with URL=_ Jun 21 18:00:18 DEBUG[25265] app_queue.c: Queue with URL=_ Jun 21 18:00:18 DEBUG[25265] app_queue.c: Trying 'Agent/8003' with metric 0 Jun 21 18:00:18 DEBUG[25265] app_queue.c: Trying 'Agent/3051' with metric 0 Jun 21 18:00:18 DEBUG[25265] app_queue.c: Trying 'Agent/3050' with metric 0 Jun 21 18:00:18 DEBUG[25265] app_queue.c: Trying 'Agent/3052' with metric 0 Jun 21 18:00:18 DEBUG[25265] app_queue.c: Trying 'Agent/3053' with metric 999982 Jun 21 18:00:18 DEBUG[25145] devicestate.c: Changing state for Local/3053@dac_suporte - state 2 (In use) Jun 21 18:00:18 DEBUG[25267] app_queue.c: Device 'Local/3053@dac_suporte' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jun 21 18:00:24 DEBUG[25145] chan_iax2.c: Checking device state for device 3051 Jun 21 18:00:24 DEBUG[25145] chan_iax2.c: iax2_devicestate(3051): Found peer. What's device state of 3051? addr=1896485036, defaddr=0 maxms=2000, lastms=1 Jun 21 18:00:24 DEBUG[25145] devicestate.c: Changing state for IAX2/3051 - state 1 (Not in use) Jun 21 18:00:24 DEBUG[25269] app_queue.c: Device 'IAX2/3051' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Jun 21 18:00:24 DEBUG[25145] chan_iax2.c: Checking device state for device 3051 Jun 21 18:00:24 DEBUG[25145] chan_iax2.c: iax2_devicestate(3051): Found peer. What's device state of 3051? addr=1896485036, defaddr=0 maxms=2000, lastms=1 Jun 21 18:00:24 DEBUG[25145] devicestate.c: Changing state for IAX2/3051 - state 1 (Not in use) Jun 21 18:00:24 DEBUG[25270] app_queue.c: Device 'IAX2/3051' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.