Asterisk 1.4.16.2, Copyright (C) 1999 - 2007 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/extconfig.conf': Found == Binding cdr to mysql/asterisk/cdr == Binding queues to mysql/asterisk/queues == Binding sipusers to mysql/asterisk/sip == Binding sippeers to mysql/asterisk/sip == Binding extensions to mysql/asterisk/extensions == Binding voicemail to mysql/asterisk/voicemail_users == Binding queue_members to mysql/asterisk/queue_members Connected to Asterisk 1.4.16.2 currently running on KGSIVOIPP01 (pid = 28235) KGSIVOIPP01*CLI> Verbosity is at least 16 KGSIVOIPP01*CLI> Core debug is at least 9 KGSIVOIPP01*CLI> reload KGSIVOIPP01*CLI> [Dec 31 13:03:29] NOTICE[30011]: cdr.c:1371 do_reload: CDR simple logging enabled. == Parsing '/etc/asterisk/extconfig.conf': Found == Binding cdr to mysql/asterisk/cdr == Binding queues to mysql/asterisk/queues == Binding sipusers to mysql/asterisk/sip == Binding sippeers to mysql/asterisk/sip == Binding extensions to mysql/asterisk/extensions == Binding voicemail to mysql/asterisk/voicemail_users == Binding queue_members to mysql/asterisk/queue_members == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 10000 -> 10010 == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger restarted Asterisk Queue Logger restarted -- Reloading module 'res_smdi' (Simplified Message Desk Interface (SMDI) Resource) == Parsing '/etc/asterisk/smdi.conf': [Dec 31 13:03:29] DEBUG[30011]: config.c:889 config_text_file_load: Parsing /etc/asterisk/smdi.conf Found [Dec 31 13:03:29] WARNING[30011]: res_smdi.c:746 reload: No SMDI interfaces were specified to listen on, not starting SDMI listener. -- Reloading module 'res_odbc' (ODBC Resource) -- Reloading module 'res_musiconhold' (Music On Hold Resource) -- Reloading module 'res_indications' (Indications Resource) -- Reloading module 'res_features' (Call Features Resource) [Dec 31 13:03:29] WARNING[30011]: res_features.c:2247 load_config: Could not load features.conf -- Reloading module 'res_crypto' (Cryptographic Digital Signatures) -- Reloading module 'res_adsi' (ADSI Resource) -- Reloading module 'pbx_dundi' (Distributed Universal Number Discovery (DUNDi)) == Parsing '/etc/asterisk/dundi.conf': [Dec 31 13:03:29] DEBUG[30011]: config.c:889 config_text_file_load: Parsing /etc/asterisk/dundi.conf Found [Dec 31 13:03:29] DEBUG[30011]: pbx_dundi.c:418 reset_global_eid: Seeding global EID '00:50:da:bb:ad:e0' from 'eth0' -- Reloading module 'pbx_config' (Text Extension Configuration) == Parsing '/etc/asterisk/extensions.conf': [Dec 31 13:03:29] DEBUG[30011]: config.c:889 config_text_file_load: Parsing /etc/asterisk/extensions.conf Found [Dec 31 13:03:29] DEBUG[30011]: pbx.c:3891 __ast_context_create: Registered context 'default' -- Registered extension context 'default' [Dec 31 13:03:29] DEBUG[30011]: pbx.c:4867 ast_add_extension2: Added extension 's' priority 1 to default -- Added extension 's' priority 1 to default [Dec 31 13:03:29] DEBUG[30011]: pbx.c:4867 ast_add_extension2: Added extension 's' priority 2 to default -- Added extension 's' priority 2 to default [Dec 31 13:03:29] DEBUG[30011]: pbx.c:4867 ast_add_extension2: Added extension '1224' priority 1 to default -- Added extension '1224' priority 1 to default [Dec 31 13:03:29] DEBUG[30011]: pbx.c:4867 ast_add_extension2: Added extension '1224' priority 2 to default -- Added extension '1224' priority 2 to default -- Including context 'phones' in context 'default' [Dec 31 13:03:29] DEBUG[30011]: pbx.c:3891 __ast_context_create: Registered context 'incoming_calls' -- Registered extension context 'incoming_calls' [Dec 31 13:03:29] DEBUG[30011]: pbx.c:3891 __ast_context_create: Registered context 'internal' -- Registered extension context 'internal' [Dec 31 13:03:29] DEBUG[30011]: pbx.c:4867 ast_add_extension2: Added extension '_812XX' priority 1 to internal -- Added extension '_812XX' priority 1 to internal [Dec 31 13:03:29] DEBUG[30011]: pbx.c:4867 ast_add_extension2: Added extension '_816-XXX-XXXX' priority 1 to internal -- Added extension '_816-XXX-XXXX' priority 1 to internal [Dec 31 13:03:29] DEBUG[30011]: pbx.c:4867 ast_add_extension2: Added extension '_913-XXX-XXXX' priority 1 to internal -- Added extension '_913-XXX-XXXX' priority 1 to internal [Dec 31 13:03:29] DEBUG[30011]: pbx.c:4867 ast_add_extension2: Added extension '555' priority 1 to internal -- Added extension '555' priority 1 to internal [Dec 31 13:03:29] DEBUG[30011]: pbx.c:4867 ast_add_extension2: Added extension '555' priority 2 to internal -- Added extension '555' priority 2 to internal [Dec 31 13:03:29] DEBUG[30011]: pbx.c:3891 __ast_context_create: Registered context 'phones' -- Registered extension context 'phones' -- Including context 'internal' in context 'phones' [Dec 31 13:03:29] DEBUG[30011]: pbx.c:4867 ast_add_extension2: Added extension '91224' priority 1 to phones -- Added extension '91224' priority 1 to phones [Dec 31 13:03:29] DEBUG[30011]: pbx.c:4867 ast_add_extension2: Added extension '91200' priority 1 to phones -- Added extension '91200' priority 1 to phones [Dec 31 13:03:29] DEBUG[30011]: pbx.c:4867 ast_add_extension2: Added extension '888' priority 1 to phones -- Added extension '888' priority 1 to phones [Dec 31 13:03:29] DEBUG[30011]: pbx.c:4867 ast_add_extension2: Added extension '886' priority 1 to phones -- Added extension '886' priority 1 to phones [Dec 31 13:03:29] DEBUG[30011]: pbx.c:4867 ast_add_extension2: Added extension '886' priority 2 to phones -- Added extension '886' priority 2 to phones [Dec 31 13:03:29] DEBUG[30011]: pbx.c:4867 ast_add_extension2: Added extension '887' priority 1 to phones -- Added extension '887' priority 1 to phones [Dec 31 13:03:29] DEBUG[30011]: pbx.c:4867 ast_add_extension2: Added extension '887' priority 2 to phones -- Added extension '887' priority 2 to phones [Dec 31 13:03:29] DEBUG[30011]: pbx.c:4867 ast_add_extension2: Added extension '889' priority 1 to phones -- Added extension '889' priority 1 to phones [Dec 31 13:03:29] DEBUG[30011]: pbx.c:4867 ast_add_extension2: Added extension '889' priority 2 to phones -- Added extension '889' priority 2 to phones [Dec 31 13:03:29] DEBUG[30011]: pbx.c:4867 ast_add_extension2: Added extension '889' priority 3 to phones -- Added extension '889' priority 3 to phones == Parsing '/etc/asterisk/users.conf': [Dec 31 13:03:29] DEBUG[30011]: config.c:889 config_text_file_load: Parsing /etc/asterisk/users.conf Found [Dec 31 13:03:29] DEBUG[30011]: pbx.c:3966 ast_merge_contexts_and_delete: must remove any reg pbx_config [Dec 31 13:03:29] DEBUG[30011]: pbx.c:5297 __ast_context_destroy: check ctx phones pbx_config [Dec 31 13:03:29] DEBUG[30011]: pbx.c:5306 __ast_context_destroy: delete ctx phones pbx_config [Dec 31 13:03:29] DEBUG[30011]: pbx.c:5297 __ast_context_destroy: check ctx internal pbx_config [Dec 31 13:03:29] DEBUG[30011]: pbx.c:5306 __ast_context_destroy: delete ctx internal pbx_config [Dec 31 13:03:29] DEBUG[30011]: pbx.c:5297 __ast_context_destroy: check ctx incoming_calls pbx_config [Dec 31 13:03:29] DEBUG[30011]: pbx.c:5306 __ast_context_destroy: delete ctx incoming_calls pbx_config [Dec 31 13:03:29] DEBUG[30011]: pbx.c:5297 __ast_context_destroy: check ctx default pbx_config [Dec 31 13:03:29] DEBUG[30011]: pbx.c:5306 __ast_context_destroy: delete ctx default pbx_config -- Reloading module 'pbx_ael' (Asterisk Extension Language Compiler) [Dec 31 13:03:29] NOTICE[30011]: pbx_ael.c:4094 pbx_load_module: Starting AEL load process. [Dec 31 13:03:29] NOTICE[30011]: pbx_ael.c:4101 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. [Dec 31 13:03:29] ERROR[30011]: ael.y:756 ael_yyerror: ==== File: /etc/asterisk/extensions.ael, Line 1, Cols: 0-0: Error: syntax error, unexpected $end, expecting 'context' [Dec 31 13:03:29] NOTICE[30011]: pbx_ael.c:4109 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [Dec 31 13:03:29] ERROR[30011]: pbx_ael.c:4122 pbx_load_module: Sorry, but 1 syntax errors and 0 semantic errors were detected. It doesn't make sense to compile. -- Reloading module 'func_odbc' (ODBC lookups) [Dec 31 13:03:29] WARNING[30011]: func_odbc.c:611 reload: Unable to load config for func_odbc: func_odbc.conf -- Reloading module 'codec_zap' (Generic Zaptel Transcoder Codec Translator) == Parsing '/etc/asterisk/codecs.conf': [Dec 31 13:03:29] DEBUG[30011]: config.c:889 config_text_file_load: Parsing /etc/asterisk/codecs.conf Found -- codec_zap: using generic PLC -- Reloading module 'codec_ulaw' (mu-Law Coder/Decoder) == Parsing '/etc/asterisk/codecs.conf': [Dec 31 13:03:29] DEBUG[30011]: config.c:889 config_text_file_load: Parsing /etc/asterisk/codecs.conf Found -- codec_ulaw: using generic PLC -- Reloading module 'codec_lpc10' (LPC10 2.4kbps Coder/Decoder) == Parsing '/etc/asterisk/codecs.conf': [Dec 31 13:03:29] DEBUG[30011]: config.c:889 config_text_file_load: Parsing /etc/asterisk/codecs.conf Found -- codec_lpc10: using generic PLC -- Reloading module 'codec_gsm' (GSM Coder/Decoder) == Parsing '/etc/asterisk/codecs.conf': [Dec 31 13:03:29] DEBUG[30011]: config.c:889 config_text_file_load: Parsing /etc/asterisk/codecs.conf Found -- codec_gsm: using generic PLC -- Reloading module 'codec_g726' (ITU G.726-32kbps G726 Transcoder) == Parsing '/etc/asterisk/codecs.conf': [Dec 31 13:03:29] DEBUG[30011]: config.c:889 config_text_file_load: Parsing /etc/asterisk/codecs.conf Found -- codec_g726: using generic PLC -- Reloading module 'codec_alaw' (A-law Coder/Decoder) == Parsing '/etc/asterisk/codecs.conf': [Dec 31 13:03:29] DEBUG[30011]: config.c:889 config_text_file_load: Parsing /etc/asterisk/codecs.conf Found -- codec_alaw: using generic PLC -- Reloading module 'codec_adpcm' (Adaptive Differential PCM Coder/Decoder) == Parsing '/etc/asterisk/codecs.conf': [Dec 31 13:03:29] DEBUG[30011]: config.c:889 config_text_file_load: Parsing /etc/asterisk/codecs.conf Found -- codec_adpcm: using generic PLC -- Reloading module 'chan_zap' (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': [Dec 31 13:03:29] DEBUG[30011]: config.c:889 config_text_file_load: Parsing /etc/asterisk/zapata.conf Found [Dec 31 13:03:29] WARNING[30011]: chan_zap.c:11176 process_zap: Ignoring switchtype [Dec 31 13:03:29] WARNING[30011]: chan_zap.c:11176 process_zap: Ignoring signalling [Dec 31 13:03:29] WARNING[30011]: chan_zap.c:11176 process_zap: Ignoring rxwink == Parsing '/etc/asterisk/users.conf': [Dec 31 13:03:29] DEBUG[30011]: config.c:889 config_text_file_load: Parsing /etc/asterisk/users.conf Found -- Reloading module 'chan_sip' (Session Initiation Protocol (SIP)) Reloading SIP == Parsing '/etc/asterisk/sip.conf': [Dec 31 13:03:29] DEBUG[28244]: config.c:889 config_text_file_load: Parsing /etc/asterisk/sip.conf Found [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:16695 reload_config: --------------- SIP reload started [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:2658 sip_destroy_user: Destroying user object from memory: dave [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:2658 sip_destroy_user: Destroying user object from memory: robin [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:2658 sip_destroy_user: Destroying user object from memory: 1000 [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:16718 reload_config: --------------- Done destroying user list [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:16721 reload_config: --------------- Done destroying registry list == Parsing '/etc/asterisk/users.conf': [Dec 31 13:03:29] DEBUG[28244]: config.c:889 config_text_file_load: Parsing /etc/asterisk/users.conf Found [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:17737 sip_do_reload: --------------- Done destroying pruned peers [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:17746 sip_do_reload: --------------- SIP reload done -- Reloading module 'chan_mgcp' (Media Gateway Control Protocol (MGCP)) Reloading MGCP KGSIVOIPP01*CLI> [Dec 31 13:03:29] NOTICE[28245]: chan_mgcp.c:4095 reload_config: Unable to load config mgcp.conf, MGCP disabled -- Reloading module 'chan_iax2' (Inter Asterisk eXchange (Ver 2)) == Parsing '/etc/asterisk/iax.conf': [Dec 31 13:03:29] DEBUG[30011]: config.c:889 config_text_file_load: Parsing /etc/asterisk/iax.conf Found == Parsing '/etc/asterisk/users.conf': [Dec 31 13:03:29] DEBUG[30011]: config.c:889 config_text_file_load: Parsing /etc/asterisk/users.conf Found > doing dnsmgr_lookup for '216.207.245.47' == Loaded firmware 'iaxy.bin' == Parsing '/etc/asterisk/iaxprov.conf': [Dec 31 13:03:29] DEBUG[30011]: config.c:889 config_text_file_load: Parsing /etc/asterisk/iaxprov.conf Found -- Loaded provisioning template 'default' -- Reloading module 'chan_agent' (Agent Proxy Channel) [Dec 31 13:03:29] NOTICE[30011]: chan_agent.c:1077 read_agent_config: No agent configuration found -- agent support disabled -- Reloading module 'cdr_odbc' (ODBC CDR Backend) [Dec 31 13:03:29] WARNING[30011]: cdr_odbc.c:256 odbc_load_module: cdr_odbc: Unable to load config for ODBC CDR's: cdr_odbc.conf -- Reloading module 'cdr_manager' (Asterisk Manager Interface CDR Backend) -- Reloading module 'cdr_custom' (Customizable Comma Separated Values CDR Backend) [Dec 31 13:03:29] WARNING[30011]: cdr_custom.c:97 load_config: Failed to reload configuration file. -- Reloading module 'cdr_csv' (Comma Separated Values CDR Backend) [Dec 31 13:03:29] WARNING[30011]: cdr_csv.c:111 load_config: unable to load config: cdr.conf -- Reloading module 'app_voicemail' (Comedian Mail (Voicemail System) with ODBC Storage) [Dec 31 13:03:29] WARNING[30011]: app_voicemail.c:7814 load_config: Failed to load configuration file. -- Reloading module 'app_queue' (True Call Queueing) [Dec 31 13:03:29] NOTICE[30011]: app_queue.c:4010 reload_queues: No call queueing config file (queues.conf), so no call queues -- Reloading module 'app_playback' (Sound File Playback Application) -- Reloading module 'app_meetme' (MeetMe conference bridge) -- Reloading module 'app_followme' (Find-Me/Follow-Me Application) [Dec 31 13:03:29] WARNING[30011]: app_followme.c:299 reload_followme: No follow me config file (followme.conf), so no follow me -- Reloading module 'app_amd' (Answering Machine Detection Application) == Parsing '/etc/asterisk/amd.conf': [Dec 31 13:03:29] DEBUG[30011]: config.c:889 config_text_file_load: Parsing /etc/asterisk/amd.conf Found -- AMD defaults: initialSilence [2500] greeting [1500] afterGreetingSilence [800] totalAnalysisTime [5000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256] KGSIVOIPP01*CLI> [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4509 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 0: OPTIONS sip:dave@198.247.174.254:37802;rinstance=f3ca7aa3adaef7d9 SIP/2.0 (73) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 1: Via: SIP/2.0/UDP 206.55.114.5:5060;branch=z9hG4bK4b87e936 (57) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 2: From: "asterisk" ;tag=as4db09928 (59) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 3: To: (63) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 4: Contact: (36) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 5: Call-ID: 3a4fed6c7154a16700f485ad0a637501@206.55.114.5 (54) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 9: Date: Mon, 31 Dec 2007 19:03:29 GMT (35) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 11: Supported: replaces (19) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 12: Content-Length: 0 (17) Reliably Transmitting (NAT) to 198.247.174.254:37802: OPTIONS sip:dave@198.247.174.254:37802;rinstance=f3ca7aa3adaef7d9 SIP/2.0 Via: SIP/2.0/UDP 206.55.114.5:5060;branch=z9hG4bK4b87e936 From: "asterisk" ;tag=as4db09928 To: Contact: Call-ID: 3a4fed6c7154a16700f485ad0a637501@206.55.114.5 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 31 Dec 2007 19:03:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:2041 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #241 KGSIVOIPP01*CLI> <--- SIP read from 198.247.174.254:37802 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 206.55.114.5:5060;branch=z9hG4bK4b87e936 Contact: To: ;tag=d772b82b From: "asterisk";tag=as4db09928 Call-ID: 3a4fed6c7154a16700f485ad0a637501@206.55.114.5 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: eyeBeam release 1013t stamp 43070 Content-Length: 0 <-------------> [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 1: Via: SIP/2.0/UDP 206.55.114.5:5060;branch=z9hG4bK4b87e936 (57) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 2: Contact: (34) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 3: To: ;tag=d772b82b (76) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 4: From: "asterisk";tag=as4db09928 (58) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 5: Call-ID: 3a4fed6c7154a16700f485ad0a637501@206.55.114.5 (54) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 7: Accept: application/sdp (23) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 8: Accept-Language: en (19) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 10: User-Agent: eyeBeam release 1013t stamp 43070 (45) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 11: Content-Length: 0 (17) [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:4562 find_call: = Found Their Call ID: 3a4fed6c7154a16700f485ad0a637501@206.55.114.5 Their Tag Our tag: as4db09928 [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:2160 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #241 [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:2170 __sip_ack: Stopping retransmission on '3a4fed6c7154a16700f485ad0a637501@206.55.114.5' of Request 102: Match Not Found Really destroying SIP dialog '3a4fed6c7154a16700f485ad0a637501@206.55.114.5' Method: OPTIONS [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:11018 sip_dump_history: ---------- SIP HISTORY for '3a4fed6c7154a16700f485ad0a637501@206.55.114.5' [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:11022 sip_dump_history: * SIP Call [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:11027 sip_dump_history: Call '3a4fed6c7154a16700f485ad0a637501@206.55.114.5' has no history [Dec 31 13:03:29] DEBUG[28244]: chan_sip.c:11028 sip_dump_history: ---------- END SIP HISTORY for '3a4fed6c7154a16700f485ad0a637501@206.55.114.5' KGSIVOIPP01*CLI> reloadquitcore set debug 9rtp debug KGSIVOIPP01*CLI> RTP Debugging Enabled KGSIVOIPP01*CLI> rtp debugeloadquitcore set debug 9rtp debug[4@sip set KGSIVOIPP01*CLI> SIP Debugging re-enabled KGSIVOIPP01*CLI> <--- SIP read from 198.247.174.254:37802 ---> INVITE sip:555@206.55.114.5 SIP/2.0 Via: SIP/2.0/UDP 198.247.174.254:37802;branch=z9hG4bK-d8754z-90773e57cc6b8836-1---d8754z-;rport Max-Forwards: 70 Contact: To: "555" From: "Dave-dev";tag=3236496f Call-ID: YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1013t stamp 43070 Content-Length: 425 v=0 o=- 3 2 IN IP4 198.247.174.254 s=CounterPath eyeBeam 1.5 c=IN IP4 198.247.174.254 t=0 0 m=audio 15442 RTP/AVP 107 100 106 6 0 105 8 18 3 5 101 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:11A02DCBF57D42829508EF1F2FCABEC9 <-------------> [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 0: INVITE sip:555@206.55.114.5 SIP/2.0 (35) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 1: Via: SIP/2.0/UDP 198.247.174.254:37802;branch=z9hG4bK-d8754z-90773e57cc6b8836-1---d8754z-;rport (95) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 2: Max-Forwards: 70 (16) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 3: Contact: (41) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 4: To: "555" (31) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 5: From: "Dave-dev";tag=3236496f (52) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 6: Call-ID: YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. (53) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 7: CSeq: 1 INVITE (14) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 9: Content-Type: application/sdp (29) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 10: User-Agent: eyeBeam release 1013t stamp 43070 (45) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 11: Content-Length: 425 (19) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 12: (0) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: v=0 (3) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: o=- 3 2 IN IP4 198.247.174.254 (30) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: s=CounterPath eyeBeam 1.5 (25) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: c=IN IP4 198.247.174.254 (24) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: t=0 0 (5) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: m=audio 15442 RTP/AVP 107 100 106 6 0 105 8 18 3 5 101 (54) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=fmtp:18 annexb=yes (20) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=fmtp:101 0-15 (15) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=rtpmap:107 BV32/16000 (23) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=rtpmap:100 SPEEX/16000 (24) KGSIVOIPP01*CLI> [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=rtpmap:106 SPEEX-FEC/16000 (28) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=rtpmap:105 SPEEX-FEC/8000 (27) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=sendrecv (10) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=x-rtp-session-id:11A02DCBF57D42829508EF1F2FCABEC9 (51) --- (12 headers 16 lines) --- [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:2733 do_setnat: Setting NAT on RTP to On [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4509 sip_alloc: Allocating new SIP dialog for YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. - INVITE (With RTP) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:15153 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 198.247.174.254 : 37802 (NAT) Using INVITE request as basis request - YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:2733 do_setnat: Setting NAT on RTP to On <--- Reliably Transmitting (NAT) to 198.247.174.254:37802 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 198.247.174.254:37802;branch=z9hG4bK-d8754z-90773e57cc6b8836-1---d8754z-;rport;received=198.247.174.254 From: "Dave-dev";tag=3236496f To: "555";tag=as0aeb8dca Call-ID: YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="740e07b1" Content-Length: 0 <------------> [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:2041 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #244 Scheduling destruction of SIP dialog 'YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ.' in 32000 ms (Method: INVITE) Found user 'dave' <--- SIP read from 198.247.174.254:37802 ---> ACK sip:555@206.55.114.5 SIP/2.0 Via: SIP/2.0/UDP 198.247.174.254:37802;branch=z9hG4bK-d8754z-90773e57cc6b8836-1---d8754z-;rport To: "555";tag=as0aeb8dca From: "Dave-dev";tag=3236496f Call-ID: YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. CSeq: 1 ACK Content-Length: 0 <-------------> [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 0: ACK sip:555@206.55.114.5 SIP/2.0 (32) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 1: Via: SIP/2.0/UDP 198.247.174.254:37802;branch=z9hG4bK-d8754z-90773e57cc6b8836-1---d8754z-;rport (95) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 2: To: "555";tag=as0aeb8dca (46) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 3: From: "Dave-dev";tag=3236496f (52) KGSIVOIPP01*CLI> [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 4: Call-ID: YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. (53) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 5: CSeq: 1 ACK (11) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 6: Content-Length: 0 (17) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 7: (0) --- (7 headers 0 lines) --- [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4562 find_call: = Found Their Call ID: YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. Their Tag 3236496f Our tag: as0aeb8dca [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:15153 handle_request: **** Received ACK (6) - Command in SIP ACK [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:2160 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #244 [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:2170 __sip_ack: Stopping retransmission on 'YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ.' of Response 1: Match Not Found <--- SIP read from 198.247.174.254:37802 ---> INVITE sip:555@206.55.114.5 SIP/2.0 Via: SIP/2.0/UDP 198.247.174.254:37802;branch=z9hG4bK-d8754z-d50fad2b7e1b0224-1---d8754z-;rport Max-Forwards: 70 Contact: To: "555" From: "Dave-dev";tag=3236496f Call-ID: YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="dave",realm="asterisk",nonce="740e07b1",uri="sip:555@206.55.114.5",response="2f6889f29d926bab55dff89aa04b96e2",algorithm=MD5 User-Agent: eyeBeam release 1013t stamp 43070 Content-Length: 425 v=0 o=- 3 2 IN IP4 198.247.174.254 s=CounterPath eyeBeam 1.5 c=IN IP4 198.247.174.254 t=0 0 m=audio 15442 RTP/AVP 107 100 106 6 0 105 8 18 3 5 101 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:11A02DCBF57D42829508EF1F2FCABEC9 <-------------> [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 0: INVITE sip:555@206.55.114.5 SIP/2.0 (35) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 1: Via: SIP/2.0/UDP 198.247.174.254:37802;branch=z9hG4bK-d8754z-d50fad2b7e1b0224-1---d8754z-;rport (95) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 2: Max-Forwards: 70 (16) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 3: Contact: (41) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 4: To: "555" (31) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 5: From: "Dave-dev";tag=3236496f (52) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 6: Call-ID: YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. (53) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 7: CSeq: 2 INVITE (14) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 8: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 9: Content-Type: application/sdp (29) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 10: Proxy-Authorization: Digest username="dave",realm="asterisk",nonce="740e07b1",uri="sip:555@206.55.114.5",response="2f6889f29d926bab55dff89aa04b96e2",algorithm=MD5 (162) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 11: User-Agent: eyeBeam release 1013t stamp 43070 (45) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 12: Content-Length: 425 (19) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 13: (0) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: v=0 (3) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: o=- 3 2 IN IP4 198.247.174.254 (30) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: s=CounterPath eyeBeam 1.5 (25) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: c=IN IP4 198.247.174.254 (24) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: t=0 0 (5) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: m=audio 15442 RTP/AVP 107 100 106 6 0 105 8 18 3 5 101 (54) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=fmtp:18 annexb=yes (20) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=fmtp:101 0-15 (15) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=rtpmap:107 BV32/16000 (23) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=rtpmap:100 SPEEX/16000 (24) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=rtpmap:106 SPEEX-FEC/16000 (28) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=rtpmap:105 SPEEX-FEC/8000 (27) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=sendrecv (10) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4806 parse_request: Line: a=x-rtp-session-id:11A02DCBF57D42829508EF1F2FCABEC9 (51) --- (13 headers 16 lines) --- [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4562 find_call: = Found Their Call ID: YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. Their Tag 3236496f Our tag: as0aeb8dca [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:15153 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 198.247.174.254 : 37802 (NAT) Using INVITE request as basis request - YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:2733 do_setnat: Setting NAT on RTP to On Found user 'dave' Found RTP audio format 107 Found RTP audio format 100 Found RTP audio format 106 Found RTP audio format 6 Found RTP audio format 0 Found RTP audio format 105 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 5 Found RTP audio format 101 Peer audio RTP is at port 198.247.174.254:15442 Got unsupported a:fmtp in SDP offer Got unsupported a:fmtp in SDP offer Found unknown media description format BV32 for ID 107 Found audio description format SPEEX for ID 100 Found unknown media description format SPEEX-FEC for ID 106 Found unknown media description format SPEEX-FEC for ID 105 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:5388 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x32e (gsm|ulaw|alaw|adpcm|g729|speex)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 198.247.174.254:15442 [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:5468 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:13833 handle_request_invite: Checking SIP call limits for device dave [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:3173 update_call_counter: Updating call counter for incoming call Looking for 555 in phones (domain 206.55.114.5) [Dec 31 13:03:55] DEBUG[28244]: frame.c:1267 ast_codec_choose: Could not find preferred codec - Going for the best codec [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:3993 sip_new: *** Our native formats are 0x4 (ulaw) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:3994 sip_new: *** Joint capabilities are 0xe (gsm|ulaw|alaw) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:3995 sip_new: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263) [Dec 31 13:03:55] DEBUG[28244]: frame.c:1267 ast_codec_choose: Could not find preferred codec - Going for the best codec [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:3996 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4019 sip_new: This channel will not be able to handle video. [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:8245 build_route: build_route: Contact hop: list_route: hop: [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:13912 handle_request_invite: SIP/dave-08a4dec8: New call is still down.... Trying... <--- Transmitting (NAT) to 198.247.174.254:37802 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 198.247.174.254:37802;branch=z9hG4bK-d8754z-d50fad2b7e1b0224-1---d8754z-;rport;received=198.247.174.254 From: "Dave-dev";tag=3236496f To: "555" Call-ID: YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Dec 31 13:03:55] DEBUG[28244]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/dave-08a4dec8 [Dec 31 13:03:55] DEBUG[28239]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - dave [Dec 31 13:03:55] DEBUG[28239]: chan_sip.c:15780 sip_devicestate: Checking device state for peer dave [Dec 31 13:03:55] DEBUG[28239]: devicestate.c:287 do_state_change: Changing state for SIP/dave - state 1 (Not in use) [Dec 31 13:03:55] DEBUG[30017]: pbx.c:1827 pbx_extension_helper: Launching 'Answer' -- Executing [555@phones:1] Answer("SIP/dave-08a4dec8", "") in new stack [Dec 31 13:03:55] DEBUG[30017]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/dave-08a4dec8 [Dec 31 13:03:55] DEBUG[28239]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - dave [Dec 31 13:03:55] DEBUG[28239]: chan_sip.c:15780 sip_devicestate: Checking device state for peer dave [Dec 31 13:03:55] DEBUG[28239]: devicestate.c:287 do_state_change: Changing state for SIP/dave - state 1 (Not in use) [Dec 31 13:03:55] DEBUG[30017]: chan_sip.c:3647 sip_answer: SIP answering channel: SIP/dave-08a4dec8 [Dec 31 13:03:55] DEBUG[30017]: chan_sip.c:6661 transmit_response_with_sdp: Setting framing from config on incoming call [Dec 31 13:03:55] DEBUG[30017]: chan_sip.c:6425 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 31 13:03:55] DEBUG[30017]: chan_sip.c:6426 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 206.55.114.5 port 10008 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 31 13:03:55] DEBUG[30017]: chan_sip.c:6557 add_sdp: -- Done with adding codecs to SDP [Dec 31 13:03:55] DEBUG[30017]: channel.c:2576 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=20) [Dec 31 13:03:55] DEBUG[30017]: chan_sip.c:6602 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) <--- Reliably Transmitting (NAT) to 198.247.174.254:37802 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 198.247.174.254:37802;branch=z9hG4bK-d8754z-d50fad2b7e1b0224-1---d8754z-;rport;received=198.247.174.254 From: "Dave-dev";tag=3236496f To: "555";tag=as7697caee Call-ID: YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 28235 28235 IN IP4 206.55.114.5 s=session c=IN IP4 206.55.114.5 t=0 0 m=audio 10008 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Dec 31 13:03:55] DEBUG[30017]: chan_sip.c:2041 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #246 [Dec 31 13:03:55] DEBUG[30017]: pbx.c:1827 pbx_extension_helper: Launching 'DateTime' -- Executing [555@phones:2] DateTime("SIP/dave-08a4dec8", "") in new stack [Dec 31 13:03:55] DEBUG[30017]: say.c:3010 ast_say_date_with_format_en: Parsing A (offset 0) in ABdY 'digits/at' IMp [Dec 31 13:03:55] DEBUG[30017]: rtp.c:2759 ast_rtp_write: Ooh, format changed from unknown to ulaw [Dec 31 13:03:55] DEBUG[30017]: rtp.c:2776 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 Sent RTP packet to 198.247.174.254:15442 (type 00, seq 041983, ts 000160, len 000160) [Dec 31 13:03:55] DEBUG[30017]: channel.c:2095 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'digits/day-1' (language 'en') KGSIVOIPP01*CLI> [Dec 31 13:03:55] DEBUG[30017]: rtp.c:874 ast_rtcp_read: RTCP NAT: Got RTCP from other end. Now sending to address 198.247.174.254:41627 [Dec 31 13:03:55] DEBUG[30017]: rtp.c:879 ast_rtcp_read: Got RTCP report of 132 bytes KGSIVOIPP01*CLI> [Dec 31 13:03:55] DEBUG[30017]: rtp.c:1177 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 198.247.174.254:41628 Got RTP packet from 198.247.174.254:41628 (type 00, seq 007750, ts 1560100, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007751, ts 1560260, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007752, ts 1560420, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007753, ts 1560580, len 000160) KGSIVOIPP01*CLI> <--- SIP read from 198.247.174.254:37802 ---> ACK sip:555@206.55.114.5 SIP/2.0 Via: SIP/2.0/UDP 198.247.174.254:37802;branch=z9hG4bK-d8754z-c14cce129943105e-1---d8754z-;rport Max-Forwards: 70 Contact: To: "555";tag=as7697caee From: "Dave-dev";tag=3236496f Call-ID: YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. CSeq: 2 ACK Proxy-Authorization: Digest username="dave",realm="asterisk",nonce="740e07b1",uri="sip:555@206.55.114.5",response="2f6889f29d926bab55dff89aa04b96e2",algorithm=MD5 User-Agent: eyeBeam release 1013t stamp 43070 Content-Length: 0 <-------------> [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 0: ACK sip:555@206.55.114.5 SIP/2.0 (32) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 1: Via: SIP/2.0/UDP 198.247.174.254:37802;branch=z9hG4bK-d8754z-c14cce129943105e-1---d8754z-;rport (95) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 2: Max-Forwards: 70 (16) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 3: Contact: (41) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 4: To: "555";tag=as7697caee (46) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 5: From: "Dave-dev";tag=3236496f (52) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 6: Call-ID: YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. (53) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 7: CSeq: 2 ACK (11) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 8: Proxy-Authorization: Digest username="dave",realm="asterisk",nonce="740e07b1",uri="sip:555@206.55.114.5",response="2f6889f29d926bab55dff89aa04b96e2",algorithm=MD5 (162) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 9: User-Agent: eyeBeam release 1013t stamp 43070 (45) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 10: Content-Length: 0 (17) [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 11: (0) --- (11 headers 0 lines) --- [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:4562 find_call: = Found Their Call ID: YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. Their Tag 3236496f Our tag: as7697caee [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:15153 handle_request: **** Received ACK (6) - Command in SIP ACK [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:2160 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #246 [Dec 31 13:03:55] DEBUG[28244]: chan_sip.c:2170 __sip_ack: Stopping retransmission on 'YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ.' of Response 2: Match Not Found KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007754, ts 1560740, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007755, ts 1560900, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007756, ts 1561060, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007757, ts 1561220, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007758, ts 1561380, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007759, ts 1561540, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007760, ts 1561700, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007761, ts 1561860, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007762, ts 1562020, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007763, ts 1562180, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007764, ts 1562340, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007765, ts 1562500, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007766, ts 1562660, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007767, ts 1562820, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007768, ts 1562980, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007769, ts 1563140, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007770, ts 1563300, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007771, ts 1563460, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007772, ts 1563620, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007773, ts 1563780, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007774, ts 1563940, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007775, ts 1564100, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007776, ts 1564260, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007777, ts 1564420, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007778, ts 1564580, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007779, ts 1564740, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007780, ts 1564900, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007781, ts 1565060, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007782, ts 1565220, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007783, ts 1565380, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007784, ts 1565540, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007785, ts 1565700, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007786, ts 1565860, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007787, ts 1566020, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007788, ts 1566180, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007789, ts 1566340, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007790, ts 1566500, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007791, ts 1566660, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007792, ts 1566820, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007793, ts 1566980, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007794, ts 1567140, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007795, ts 1567300, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007796, ts 1567460, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007797, ts 1567620, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007798, ts 1567780, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007799, ts 1567940, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007800, ts 1568100, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007801, ts 1568260, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007802, ts 1568420, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007803, ts 1568580, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007804, ts 1568740, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007805, ts 1568900, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007806, ts 1569060, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007807, ts 1569220, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007808, ts 1569380, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007809, ts 1569540, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007810, ts 1569700, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007811, ts 1569860, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007812, ts 1570020, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007813, ts 1570180, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007814, ts 1570340, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007815, ts 1570500, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007816, ts 1570660, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007817, ts 1570820, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007818, ts 1570980, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007819, ts 1571140, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007820, ts 1571300, len 000160) KGSIVOIPP01*CLI> Got RTP packet from 198.247.174.254:41628 (type 00, seq 007821, ts 1571460, len 000160) KGSIVOIPP01*CLI> [Dec 31 13:03:56] DEBUG[30017]: rtp.c:874 ast_rtcp_read: RTCP NAT: Got RTCP from other end. Now sending to address 198.247.174.254:41627 [Dec 31 13:03:56] DEBUG[30017]: rtp.c:879 ast_rtcp_read: Got RTCP report of 160 bytes KGSIVOIPP01*CLI> <--- SIP read from 198.247.174.254:37802 ---> BYE sip:555@206.55.114.5 SIP/2.0 Via: SIP/2.0/UDP 198.247.174.254:37802;branch=z9hG4bK-d8754z-f254e25207071b53-1---d8754z-;rport Max-Forwards: 70 Contact: To: "555";tag=as7697caee From: "Dave-dev";tag=3236496f Call-ID: YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. CSeq: 3 BYE Proxy-Authorization: Digest username="dave",realm="asterisk",nonce="740e07b1",uri="sip:555@206.55.114.5",response="9c5f9563fa8d71595830efb5d0ca5f7e",algorithm=MD5 User-Agent: eyeBeam release 1013t stamp 43070 Reason: SIP;description="User Hung Up" Content-Length: 0 <-------------> [Dec 31 13:03:56] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 0: BYE sip:555@206.55.114.5 SIP/2.0 (32) [Dec 31 13:03:56] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 1: Via: SIP/2.0/UDP 198.247.174.254:37802;branch=z9hG4bK-d8754z-f254e25207071b53-1---d8754z-;rport (95) [Dec 31 13:03:56] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 2: Max-Forwards: 70 (16) [Dec 31 13:03:56] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 3: Contact: (41) [Dec 31 13:03:56] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 4: To: "555";tag=as7697caee (46) [Dec 31 13:03:56] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 5: From: "Dave-dev";tag=3236496f (52) [Dec 31 13:03:56] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 6: Call-ID: YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. (53) [Dec 31 13:03:56] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 7: CSeq: 3 BYE (11) [Dec 31 13:03:56] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 8: Proxy-Authorization: Digest username="dave",realm="asterisk",nonce="740e07b1",uri="sip:555@206.55.114.5",response="9c5f9563fa8d71595830efb5d0ca5f7e",algorithm=MD5 (162) [Dec 31 13:03:56] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 9: User-Agent: eyeBeam release 1013t stamp 43070 (45) [Dec 31 13:03:56] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 10: Reason: SIP;description="User Hung Up" (38) [Dec 31 13:03:56] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 11: Content-Length: 0 (17) [Dec 31 13:03:56] DEBUG[28244]: chan_sip.c:4774 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 31 13:03:56] DEBUG[28244]: chan_sip.c:4562 find_call: = Found Their Call ID: YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. Their Tag 3236496f Our tag: as7697caee [Dec 31 13:03:56] DEBUG[28244]: chan_sip.c:15153 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 198.247.174.254 : 37802 (NAT) [Dec 31 13:03:56] DEBUG[28244]: chan_sip.c:1652 sip_alreadygone: Setting SIP_ALREADYGONE on dialog YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. [Dec 31 13:03:56] DEBUG[28244]: chan_sip.c:14705 handle_request_bye: Received bye, issuing owner hangup <--- Transmitting (NAT) to 198.247.174.254:37802 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 198.247.174.254:37802;branch=z9hG4bK-d8754z-f254e25207071b53-1---d8754z-;rport;received=198.247.174.254 From: "Dave-dev";tag=3236496f To: "555";tag=as7697caee Call-ID: YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> KGSIVOIPP01*CLI> [Dec 31 13:03:56] DEBUG[30017]: pbx.c:2425 __ast_pbx_run: Spawn extension (phones,555,2) exited non-zero on 'SIP/dave-08a4dec8' == Spawn extension (phones, 555, 2) exited non-zero on 'SIP/dave-08a4dec8' [Dec 31 13:03:56] DEBUG[30017]: channel.c:1567 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/dave-08a4dec8' [Dec 31 13:03:56] DEBUG[30017]: channel.c:2095 ast_settimeout: Scheduling timer at 0 sample intervals [Dec 31 13:03:56] DEBUG[30017]: channel.c:1786 ast_hangup: Hanging up channel 'SIP/dave-08a4dec8' [Dec 31 13:03:56] DEBUG[30017]: chan_sip.c:3485 sip_hangup: Hangup call SIP/dave-08a4dec8, SIP callid YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ.) [Dec 31 13:03:56] DEBUG[30017]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/dave-08a4dec8 KGSIVOIPP01*CLI> [Dec 31 13:03:56] DEBUG[28239]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - dave [Dec 31 13:03:56] DEBUG[28239]: chan_sip.c:15780 sip_devicestate: Checking device state for peer dave [Dec 31 13:03:56] DEBUG[28239]: devicestate.c:287 do_state_change: Changing state for SIP/dave - state 1 (Not in use) KGSIVOIPP01*CLI> Really destroying SIP dialog 'YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ.' Method: BYE [Dec 31 13:03:57] DEBUG[28244]: chan_sip.c:11018 sip_dump_history: ---------- SIP HISTORY for 'YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ.' [Dec 31 13:03:57] DEBUG[28244]: chan_sip.c:11022 sip_dump_history: * SIP Call [Dec 31 13:03:57] DEBUG[28244]: chan_sip.c:11025 sip_dump_history: 001. AuthChal Auth challenge sent for - nc 0 [Dec 31 13:03:57] DEBUG[28244]: chan_sip.c:11025 sip_dump_history: 002. CancelDestroy [Dec 31 13:03:57] DEBUG[28244]: chan_sip.c:11025 sip_dump_history: 003. Invite New call: YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ. [Dec 31 13:03:57] DEBUG[28244]: chan_sip.c:11025 sip_dump_history: 004. AuthOK Auth challenge succesful for dave [Dec 31 13:03:57] DEBUG[28244]: chan_sip.c:11025 sip_dump_history: 005. Hangup Cause Unknown [Dec 31 13:03:57] DEBUG[28244]: chan_sip.c:11028 sip_dump_history: ---------- END SIP HISTORY for 'YzU1MjQ0ZTUzNmM3ODg2NTMxMTkwNDJhYzUwNDFjMTQ.' KGSIVOIPP01*CLI> quit Executing last minute cleanups