[root@foxtrot asterisk]# asterisk -vvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk SVN-trunk-r33869, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type '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 'show license' for details. ========================================================================= == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log == Parsing '/etc/asterisk/dnsmgr.conf': Found Asterisk Dynamic Loader loading preload modules: == Parsing '/etc/asterisk/modules.conf': Found [res_odbc.so] => (ODBC Resource) == Parsing '/etc/asterisk/res_odbc.conf': Found Jun 20 18:21:53 NOTICE[21502]: res_odbc.c:220 load_odbc_config: Adding ENV var: INFORMIXSERVER=my_special_database Jun 20 18:21:53 NOTICE[21502]: res_odbc.c:220 load_odbc_config: Adding ENV var: INFORMIXDIR=/opt/informix Jun 20 18:21:53 NOTICE[21502]: res_odbc.c:220 load_odbc_config: Adding ENV var: ODBCINI=/etc/odbc.ini Jun 20 18:21:53 NOTICE[21502]: res_odbc.c:493 odbc_obj_connect: Connecting debtnet Jun 20 18:21:53 NOTICE[21502]: res_odbc.c:507 odbc_obj_connect: res_odbc: Connected to debtnet [debtnet] Jun 20 18:21:53 NOTICE[21502]: res_odbc.c:293 load_odbc_config: Registered ODBC class 'debtnet' dsn->[debtnet] Jun 20 18:21:53 NOTICE[21502]: res_odbc.c:671 load_module: res_odbc loaded. [res_config_odbc.so] => (ODBC Configuration) Jun 20 18:21:53 NOTICE[21502]: config.c:854 ast_config_engine_register: Registered Config Engine odbc res_config_odbc loaded. Jun 20 18:21:53 DEBUG[21502]: loader.c:389 fixup: ---- fixup (load_modules): 2 modules, 0 new --- Jun 20 18:21:53 DEBUG[21502]: loader.c:394 fixup: ---- fixup: cycle 0 --- Jun 20 18:21:53 DEBUG[21502]: loader.c:425 fixup: ---- fixup complete --- == Parsing '/etc/asterisk/http.conf': Found == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxStatus == Manager registered action MailboxCount == Manager registered action ListCommands == Manager registered action UserEvent == Manager registered action WaitEvent == Parsing '/etc/asterisk/manager.conf': Found Asterisk Management interface listening on port 5038 == Parsing '/etc/asterisk/cdr.conf': Found Jun 20 18:21:53 NOTICE[21502]: cdr.c:1093 do_reload: CDR simple logging enabled. == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 10000 -> 20000 == Parsing '/etc/asterisk/udptl.conf': Found == UDPTL allocating from port range 4000 -> 4999 Asterisk PBX Core Initializing Registering builtin applications: [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [ExecIfTime] == Registered application 'ExecIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Progress] == Registered application 'Progress' [ResetCDR] == Registered application 'ResetCDR' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SayAlpha] == Registered application 'SayAlpha' [SayPhonetic] == Registered application 'SayPhonetic' [SetAMAFlags] == Registered application 'SetAMAFlags' [SetGlobalVar] == Registered application 'SetGlobalVar' [Set] == Registered application 'Set' [ImportVar] == Registered application 'ImportVar' [Wait] == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [res_musiconhold.so] => (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold' == Parsing '/etc/asterisk/musiconhold.conf': Found [res_convert.so] => (File format conversion CLI command) [res_jabber.so] => (AJI - Asterisk JABBER Interface) == Parsing '/etc/asterisk/jabber.conf': Found == Registered application 'JABBERSend' == Registered application 'JABBERStatus' Jun 20 18:21:53 NOTICE[21502]: res_jabber.c:2276 load_module: res_jabber.so loaded. [res_config_pgsql.so]JABBER: reconnecting. => (Postgresql RealTime Configuration Driver) Jun 20 18:21:53 DEBUG[21502]: res_config_pgsql.c:687 parse_config: Postgresql RealTime Host: Jun 20 18:21:53 DEBUG[21502]: res_config_pgsql.c:688 parse_config: Postgresql RealTime Port: 5432 Jun 20 18:21:53 DEBUG[21502]: res_config_pgsql.c:692 parse_config: Postgresql RealTime User: Jun 20 18:21:53 DEBUG[21502]: res_config_pgsql.c:693 parse_config: Postgresql RealTime Password: Jun 20 18:21:53 DEBUG[21502]: res_config_pgsql.c:694 parse_config: Postgresql RealTime DBName: Jun 20 18:21:53 DEBUG[21502]: res_config_pgsql.c:749 pgsql_reconnect: 100 connInfo=host= port=5432 dbname= user= password= Jun 20 18:21:53 DEBUG[21502]: res_config_pgsql.c:751 pgsql_reconnect: 100 connInfo=host= port=5432 dbname= user= password= Jun 20 18:21:53 DEBUG[21502]: res_config_pgsql.c:754 pgsql_reconnect: pgsqlConn=0x97843c8 Jun 20 18:21:53 DEBUG[21502]: res_config_pgsql.c:756 pgsql_reconnect: Postgresql RealTime: Successfully connected to database. Jun 20 18:21:53 NOTICE[21502]: config.c:854 ast_config_engine_register: Registered Config Engine pgsql Postgresql RealTime driver loaded. [res_agi.so] => (Asterisk Gateway Interface (AGI)) == Registered application 'DeadAGI' == Registered application 'EAGI' == Registered application 'AGI' [res_adsi.so] => (ADSI Resource) == Parsing '/etc/asterisk/adsi.conf': Found [res_indications.so] => (Indications Configuration) == Parsing '/etc/asterisk/indications.conf': Found -- Registered indication country 'at' -- Registered indication country 'au' -- Registered indication country 'br' -- Registered indication country 'be' -- Registered indication country 'ch' -- Registered indication country 'cl' -- Registered indication country 'cn' -- Registered indication country 'cz' -- Registered indication country 'de' -- Registered indication country 'dk' -- Registered indication country 'ee' -- Registered indication country 'es' -- Registered indication country 'fi' -- Registered indication country 'fr' -- Registered indication country 'gr' -- Registered indication country 'hu' -- Registered indication country 'it' -- Registered indication country 'lt' -- Registered indication country 'mx' -- Registered indication country 'my' -- Registered indication country 'nl' -- Registered indication country 'no' -- Registered indication country 'nz' -- Registered indication country 'pl' -- Registered indication country 'pt' -- Registered indication country 'ru' -- Registered indication country 'se' -- Registered indication country 'sg' -- Registered indication country 'th' -- Registered indication country 'uk' -- Registered indication country 'us' -- Registered indication country 'us-o' -- Registered indication country 'tw' -- Registered indication country 'za' -- Setting default indication country to 'us' == Registered application 'PlayTones' == Registered application 'StopPlayTones' [res_speech.so] => (Generic Speech Recognition API) [res_clioriginate.so] => (Call origination from the CLI) [res_features.so] => (Call Features Resource) == Parsing '/etc/asterisk/features.conf': Found -- Registered extension context 'parkedcalls' -- Added extension '700' priority 1 to parkedcalls == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls == Manager registered action Park [res_smdi.so] => (Asterisk Simplified Message Desk Interface (SMDI) Module) == Parsing '/etc/asterisk/smdi.conf': Found Jun 20 18:21:53 WARNING[21502]: res_smdi.c:728 load_module: No SMDI interfaces are available to listen on, not starting SDMI listener. [res_crypto.so] => (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'iaxtel' -- Loaded PUBLIC key 'freeworlddialup' [res_monitor.so] => (Call Monitoring Resource) == Registered application 'Monitor' == Registered application 'StopMonitor' == Registered application 'ChangeMonitor' == Registered application 'PauseMonitor' == Registered application 'UnpauseMonitor' == Manager registered action Monitor == Manager registered action StopMonitor == Manager registered action ChangeMonitor == Manager registered action PauseMonitor == Manager registered action UnpauseMonitor [pbx_config.so] => (Text Extension Configuration) == Parsing '/etc/asterisk/extensions.conf': Found == Parsing '/etc/asterisk/qdialplan.conf': Found == Setting global variable 'CONSOLE' to 'Console/dsp' -- Registered extension context 'isdn10' -- Added extension '611734' priority 1 (CID match '08005875290')to isdn10 -- Added extension '611734' priority 2 (CID match '08005875290')to isdn10 -- Added extension '611734' priority 3 (CID match '08005875290')to isdn10 -- Added extension '611734' priority 4 (CID match '08005875290')to isdn10 -- Added extension '444611' priority 1 (CID match '08005875290')to isdn10 -- Added extension '444611' priority 2 (CID match '08005875290')to isdn10 -- Added extension '444611' priority 3 (CID match '08005875290')to isdn10 -- Added extension '444611' priority 4 (CID match '08005875290')to isdn10 -- Added extension '_4446XX' priority 1 to isdn10 -- Added extension 'fax' priority 1 to isdn10 -- Registered extension context 'from-sip' -- Including context 'common' in context 'from-sip' -- Added extension '_**6XX' priority 1 to from-sip -- Added extension '_**6XX' priority 2 to from-sip -- Added extension '_**6XX' priority 3 to from-sip -- Added extension '_0.' priority 1 to from-sip -- Added extension '_0.' priority 2 to from-sip -- Added extension '_0.' priority 3 to from-sip -- Added extension '_XXXXXX' priority 1 to from-sip -- Added extension '_XXXXXX' priority 2 to from-sip -- Added extension '_XXXXXX' priority 3 to from-sip -- Added extension 'h' priority 1 to from-sip -- Added extension 'h' priority 2 to from-sip -- Registered extension context 'common' -- Added extension '779' priority 1 to common -- Added extension '701' priority 1 to common -- Added extension '702' priority 1 to common -- Added extension '705' priority 1 to common -- Added extension '705' priority 2 to common -- Added extension '705' priority 3 to common -- Added extension '711' priority 1 to common -- Added extension '711' priority 2 to common -- Added extension '714' priority 1 to common -- Added extension '714' priority 2 to common -- Added extension '715' priority 1 to common -- Added extension '715' priority 2 to common -- Added extension '_7XX' priority 1 to common -- Added extension '_7XX' priority 2 to common -- Added extension '_7XX' priority 3 to common -- Added extension '807' priority 1 to common -- Added extension '807' priority 2 to common -- Added extension '807' priority 3 to common -- Added extension '807' priority 4 to common -- Added extension '807' priority 5 to common -- Registered extension context 'fax' -- Registered extension context 'AgentQ' -- Added extension '_X.' priority 1 to AgentQ -- Added extension '_X.' priority 2 to AgentQ -- Added extension '_X.' priority 3 to AgentQ -- Added extension '_X.' priority 4 to AgentQ -- Added extension '_X.' priority 5 to AgentQ -- Added extension '_X.' priority 6 to AgentQ -- Added extension '_X.' priority 7 to AgentQ -- Added extension '_X.' priority 8 to AgentQ -- Added extension '_X.' priority 9 to AgentQ -- Added extension '_X.' priority 10 to AgentQ -- Added extension 'h' priority 1 to AgentQ -- Registered extension context 'from-debtnet' -- Added extension '_X.' priority 1 to from-debtnet -- Added extension '_X.' priority 2 to from-debtnet -- Added extension '_X.' priority 3 to from-debtnet -- Added extension '_X.' priority 4 to from-debtnet -- Added extension '_X.' priority 5 to from-debtnet -- Added extension '_X.' priority 6 to from-debtnet -- Added extension '_X.' priority 7 to from-debtnet -- Added extension 'h' priority 8 to from-debtnet -- Registered extension context 'DialResult' -- Added extension 'r1' priority 1 to DialResult -- Added extension 'r1' priority 2 to DialResult -- Added extension 'r16' priority 1 to DialResult -- Added extension 'r17' priority 1 to DialResult -- Added extension 'r18' priority 1 to DialResult -- Added extension 'r18' priority 2 to DialResult -- Added extension 'r18' priority 3 to DialResult -- Added extension 'r19' priority 1 to DialResult -- Added extension 'r19' priority 2 to DialResult -- Added extension 'r27' priority 1 to DialResult -- Added extension 'r27' priority 2 to DialResult -- Added extension 'r28' priority 1 to DialResult -- Added extension 'r28' priority 2 to DialResult -- Added extension 'r31' priority 1 to DialResult -- Added extension 'r31' priority 2 to DialResult -- Added extension 'r34' priority 1 to DialResult -- Added extension 'r34' priority 2 to DialResult -- Added extension 'r41' priority 1 to DialResult -- Added extension 'r41' priority 2 to DialResult -- Added extension 'r41' priority 3 to DialResult -- Added extension 'r42' priority 1 to DialResult -- Added extension 'r42' priority 2 to DialResult -- Added extension 'r63' priority 1 to DialResult -- Added extension 'r63' priority 2 to DialResult -- Added extension 'i' priority 1 to DialResult -- Added extension 'i' priority 2 to DialResult -- Added extension 'i' priority 3 to DialResult -- Added extension 'i' priority 4 to DialResult -- Added extension 'i' priority 5 to DialResult -- Registered extension context 'macro-DialExternal' -- Added extension 's' priority 1 to macro-DialExternal -- Added extension 's' priority 2 to macro-DialExternal -- Added extension 's' priority 3 to macro-DialExternal -- Added extension 's' priority 4 to macro-DialExternal -- Added extension 's' priority 5 to macro-DialExternal -- Added extension 's' priority 6 to macro-DialExternal -- Added extension 's' priority 7 to macro-DialExternal -- Added extension 'h' priority 1 to macro-DialExternal -- Registered extension context 'macro-connected' -- Added extension 's' priority 1 to macro-connected -- Added extension 's' priority 2 to macro-connected -- Registered extension context 'macro-AgentConnected' -- Added extension 's' priority 1 to macro-AgentConnected -- Added extension 's' priority 2 to macro-AgentConnected -- Added extension 's' priority 3 to macro-AgentConnected -- Registered extension context 'macro-callq' -- Added extension 's' priority 1 to macro-callq -- Added extension 's' priority 2 to macro-callq -- Added extension 's' priority 3 to macro-callq -- Added extension 's' priority 4 to macro-callq -- Added extension 's' priority 5 to macro-callq -- Added extension 's' priority 6 to macro-callq -- Added extension 's' priority 7 to macro-callq -- Added extension 's' priority 8 to macro-callq -- Added extension 's' priority 9 to macro-callq -- Added extension 's' priority 10 to macro-callq -- Added extension 's' priority 11 to macro-callq -- Added extension 'h' priority 1 to macro-callq -- Added extension 'h' priority 2 to macro-callq -- Added extension 'h' priority 3 to macro-callq -- Added extension 'h' priority 4 to macro-callq -- Added extension 'h' priority 5 to macro-callq -- Added extension 'h' priority 6 to macro-callq -- Added extension 'h' priority 7 to macro-callq Jun 20 18:21:54 DEBUG[21502]: pbx.c:3510 ast_merge_contexts_and_delete: must remove any reg pbx_config Jun 20 18:21:54 DEBUG[21502]: pbx.c:4845 __ast_context_destroy: check ctx parkedcalls res_features [pbx_spool.so] => (Outgoing Spool Support) [pbx_loopback.so] => (Loopback Switch) [pbx_ael.so] => (Asterisk Extension Language Compiler v2) Jun 20 18:21:54 NOTICE[21502]: pbx_ael.c:3440 pbx_load_module: Starting AEL load process. Jun 20 18:21:54 NOTICE[21502]: pbx_ael.c:3447 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. Jun 20 18:21:54 ERROR[21502]: ael.flex:561 ael2_parse: File /etc/asterisk/extensions.ael could not be opened Jun 20 18:21:54 NOTICE[21502]: pbx_ael.c:3450 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. Jun 20 18:21:54 ERROR[21502]: pbx_ael.c:3463 pbx_load_module: Sorry, but 1 syntax errors and 0 semantic errors were detected. It doesn't make sense to compile. [pbx_realtime.so] => (Realtime Switch) [pbx_dundi.so] => (Distributed Universal Number Discovery (DUNDi)) == Parsing '/etc/asterisk/dundi.conf': Found Jun 20 18:21:55 DEBUG[21502]: pbx_dundi.c:408 reset_global_eid: Seeding global EID '00:12:79:3c:c9:10' from 'eth0' == Using TOS bits 0 == DUNDi Ready and Listening on 0.0.0.0 port 4520 == Registered custom function DUNDILOOKUP [chan_features.so] => (Feature Proxy Channel) == Registered channel type 'Feature' (Feature Proxy Channel Driver) [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver) [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) == Parsing '/etc/asterisk/skinny.conf': Found == Skinny listening on 0.0.0.0:2000 == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [chan_agent.so] => (Agent Proxy Channel) == Registered channel type 'Agent' (Call Agent Proxy Channel) == Registered application 'AgentLogin' == Registered application 'AgentCallbackLogin' == Registered application 'AgentMonitorOutgoing' == Manager registered action Agents == Manager registered action AgentLogoff == Manager registered action AgentCallbackLogin == Registered custom function AGENT == Parsing '/etc/asterisk/agents.conf': Found [chan_local.so] => (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_zap.so] => (Zapata Telephony) == Registered application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/zapata.conf': Found Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1502 set_actual_txgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1520 set_actual_rxgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1357 update_conf: Updated conferencing on 1, with 0 conference users -- Registered channel 1, PRI Signalling signalling Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1502 set_actual_txgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1520 set_actual_rxgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1357 update_conf: Updated conferencing on 2, with 0 conference users -- Registered channel 2, PRI Signalling signalling Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1502 set_actual_txgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1520 set_actual_rxgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1357 update_conf: Updated conferencing on 3, with 0 conference users -- Registered channel 3, PRI Signalling signalling Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1502 set_actual_txgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1520 set_actual_rxgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1357 update_conf: Updated conferencing on 4, with 0 conference users -- Registered channel 4, PRI Signalling signalling Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1502 set_actual_txgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1520 set_actual_rxgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1357 update_conf: Updated conferencing on 5, with 0 conference users -- Registered channel 5, PRI Signalling signalling Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1502 set_actual_txgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1520 set_actual_rxgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1357 update_conf: Updated conferencing on 6, with 0 conference users -- Registered channel 6, PRI Signalling signalling Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1502 set_actual_txgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1520 set_actual_rxgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1357 update_conf: Updated conferencing on 7, with 0 conference users -- Registered channel 7, PRI Signalling signalling Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1502 set_actual_txgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1520 set_actual_rxgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1357 update_conf: Updated conferencing on 8, with 0 conference users -- Registered channel 8, PRI Signalling signalling Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1502 set_actual_txgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1520 set_actual_rxgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1357 update_conf: Updated conferencing on 9, with 0 conference users -- Registered channel 9, PRI Signalling signalling Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1502 set_actual_txgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1520 set_actual_rxgain: Failed to read gains: Invalid argument Jun 20 18:21:55 DEBUG[21502]: chan_zap.c:1357 update_conf: Updated conferencing on 10, with 0 conference users -- Registered channel 10, PRI Signalling signalling -- Automatically generated pseudo channel == Starting D-Channel on span 1 == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI) == Manager registered action ZapTransfer == Manager registered action ZapHangup == Manager registered action ZapDialOffhook == Manager registered action ZapDNDon == Manager registered action ZapDNDoff == Manager registered action ZapShowChannels == Manager registered action ZapRestart [chan_mgcp.so] => (Media Gateway Control Protocol (MGCP)) == Parsing '/etc/asterisk/mgcp.conf': Found == MGCP Listening on 0.0.0.0:2727 == Using TOS bits 0 == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found -- SIP Seeding peer from astdb: '747' at 747@192.168.6.215:5060 for 3600 -- SIP Seeding peer from astdb: '706' at 706@192.168.0.200:5060 for 3600 -- SIP Seeding peer from astdb: '711' at 711@192.168.6.252:5060 for 3600 -- SIP Seeding peer from astdb: '731' at 731@192.168.6.156:5060 for 3600 == SIP Listening on 0.0.0.0:5060 == Using SIP TOS: none == Parsing '/etc/asterisk/sip_notify.conf': Found == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' == Registered application 'SIPAddHeader' == Registered custom function SIP_HEADER == Registered custom function SIPPEER == Registered custom function SIPCHANINFO == Registered custom function CHECKSIPDOMAIN == Manager registered action SIPpeers == Manager registered action SIPshowpeer [chan_jingle.so] => (Jingle Channel) == Parsing '/etc/asterisk/jingle.conf': Found == Registered channel type 'Jingle' (Jingle Channel Driver) [chan_oss.so] => (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found == Registered channel type 'Console' (OSS Console Channel Driver) [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) == Registered custom function IAXPEER == Registered application 'IAX2Provision' == Manager registered action IAXpeers == Manager registered action IAXnetstats == Parsing '/etc/asterisk/iax.conf': Found == Using TOS bits 0 == Binding IAX2 to default address 0.0.0.0:4569 > doing dnsmgr_lookup for '216.207.245.47' == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == 10 helper threaads started == IAX Ready and Listening == Loaded firmware 'iaxy.bin' == Parsing '/etc/asterisk/iaxprov.conf': Found -- Loaded provisioning template 'default' [skipping chan_alsa.so] [app_record.so] => (Trivial Record Application) == Registered application 'Record' [app_getcpeid.so] => (Get ADSI CPE ID) == Registered application 'GetCPEID' [app_realtime.so] => (Realtime Data Lookup/Rewrite) == Registered application 'RealTimeUpdate' == Registered application 'RealTime' [app_sendtext.so] => (Send Text Applications) == Registered application 'SendText' [format_g726.so] => (Raw G.726 (16/24/32/40kbps) data) == Registered file format g726-40, extension(s) g726-40 == Registered file format g726-32, extension(s) g726-32 == Registered file format g726-24, extension(s) g726-24 == Registered file format g726-16, extension(s) g726-16 [func_math.so] => (Mathematical dialplan function) == Registered custom function MATH [app_directory.so] => (Extension Directory) == Registered application 'Directory' [app_privacy.so] => (Require phone number to be entered, if no CallerID sent) == Registered application 'PrivacyManager' [format_g729.so] => (Raw G729 data) == Registered file format g729, extension(s) g729 [app_externalivr.so] => (External IVR Interface Application) == Registered application 'ExternalIVR' [format_h264.so] => (Raw h264 data) == Registered file format h264, extension(s) h264 [app_mp3.so] => (Silly MP3 Application) == Registered application 'MP3Player' [app_system.so] => (Generic System() application) == Registered application 'TrySystem' == Registered application 'System' [func_base64.so] => (base64 encode/decode dialplan functions) == Registered custom function BASE64_ENCODE == Registered custom function BASE64_DECODE [app_setcallerid.so] => (Set CallerID Application) == Registered application 'SetCallerPres' == Registered application 'SetCallerID' [func_cdr.so] => (CDR dialplan function) == Registered custom function CDR [codec_g726.so] => (ITU G.726-32kbps G726 Transcoder) == Parsing '/etc/asterisk/codecs.conf': Found -- codec_g726: using generic PLC == Registered translator 'g726tolin' from format g726 to slin, cost 1 == Registered translator 'lintog726' from format slin to g726, cost 1 [app_while.so] => (While Loops and Conditional Execution) == Registered application 'While' == Registered application 'EndWhile' == Registered application 'ExitWhile' == Registered application 'ContinueWhile' [app_morsecode.so] => (Morse code) == Registered application 'Morsecode' [app_controlplayback.so] => (Control Playback Application) == Registered application 'ControlPlayback' [app_festival.so] => (Simple Festival Interface) == Registered application 'Festival' [func_language.so] => (Channel language dialplan function) == Registered custom function LANGUAGE [app_image.so] => (Image Transmission Application) == Registered application 'SendImage' [app_waitforsilence.so] => (Wait For Silence) == Registered application 'WaitForSilence' [func_odbc.so] => (ODBC lookups) == Parsing '/etc/asterisk/func_odbc.conf': Found == Registered custom function ODBC_SQL == Registered custom function ODBC_GETUSER == Registered custom function SQL_ESC [format_gsm.so] => (Raw GSM data) == Registered file format gsm, extension(s) gsm [app_authenticate.so] => (Authentication Application) == Registered application 'Authenticate' [app_nbscat.so] => (Silly NBS Stream Application) == Registered application 'NBScat' [app_talkdetect.so] => (Playback with Talk Detection) == Registered application 'BackgroundDetect' [app_sayunixtime.so] => (Say time) == Registered application 'SayUnixTime' == Registered application 'DateTime' [format_ogg_vorbis.so] => (OGG/Vorbis audio) == Registered file format ogg_vorbis, extension(s) ogg [app_mixmonitor.so] => (Mixed Audio Monitoring Application) == Registered application 'MixMonitor' == Registered application 'StopMixMonitor' [app_read.so] => (Read Variable Application) == Registered application 'Read' [codec_adpcm.so] => (Adaptive Differential PCM Coder/Decoder) == Parsing '/etc/asterisk/codecs.conf': Found -- codec_adpcm: using generic PLC == Registered translator 'adpcmtolin' from format adpcm to slin, cost 1 == Registered translator 'lintoadpcm' from format slin to adpcm, cost 1 [app_url.so] => (Send URL Applications) == Registered application 'SendURL' [app_dial.so] => (Dialing Application) == Registered application 'Dial' == Registered application 'RetryDial' [format_vox.so] => (Dialogic VOX (ADPCM) File Format) == Registered file format vox, extension(s) vox [func_moh.so] => (Music-on-hold dialplan function) == Registered custom function MUSICCLASS [app_voicemail.so] => (Comedian Mail (Voicemail System)) == Registered application 'VoiceMail' == Registered application 'VoiceMailMain' == Registered application 'MailboxExists' == Registered application 'VMAuthenticate' == Parsing '/etc/asterisk/voicemail.conf': Found Jun 20 18:21:56 DEBUG[21502]: app_voicemail.c:6341 load_config: VM Review Option disabled globally Jun 20 18:21:56 DEBUG[21502]: app_voicemail.c:6348 load_config: VM Temperary Greeting Reminder Option disabled globally Jun 20 18:21:56 DEBUG[21502]: app_voicemail.c:6356 load_config: VM Operator break disabled globally Jun 20 18:21:56 DEBUG[21502]: app_voicemail.c:6362 load_config: VM CID Info before msg disabled globally Jun 20 18:21:56 DEBUG[21502]: app_voicemail.c:6368 load_config: Send Voicemail msg disabled globally Jun 20 18:21:56 DEBUG[21502]: app_voicemail.c:6374 load_config: ENVELOPE before msg enabled globally Jun 20 18:21:56 DEBUG[21502]: app_voicemail.c:6380 load_config: Duration info before msg enabled globally Jun 20 18:21:56 DEBUG[21502]: app_voicemail.c:6395 load_config: We are not going to skip to the next msg after save/delete Jun 20 18:21:56 WARNING[21502]: app_voicemail.c:6450 load_config: Invalid timezone definition at line 178 [app_dictate.so] => (Virtual Dictation Machine) == Registered application 'Dictate' [app_flash.so] => (Flash zap trunk application) == Registered application 'Flash' [app_milliwatt.so] => (Digital Milliwatt (mu-law) Test Application) == Registered application 'Milliwatt' [app_db.so] => (Database Access Functions) == Registered application 'DBdel' == Registered application 'DBdeltree' [func_rand.so] => (Random number dialplan function) == Registered custom function RAND [cdr_manager.so] => (Asterisk Call Manager CDR Backend) == Parsing '/etc/asterisk/cdr_manager.conf': Found [func_db.so] => (Database (astdb) related dialplan functions) == Registered custom function DB == Registered custom function DB_EXISTS == Registered custom function DB_DELETE [app_readfile.so] => (Stores output of file into a variable) == Registered application 'ReadFile' [func_enum.so] => (ENUM related dialplan functions) == Registered custom function ENUMLOOKUP == Registered custom function TXTCIDNAME [app_verbose.so] => (Send verbose output) == Registered application 'Log' == Registered application 'Verbose' [app_channelredirect.so] => (Channel Redirect) == Registered application 'ChannelRedirect' [format_g723.so] => (G.723.1 Simple Timestamp File Format) == Registered file format g723sf, extension(s) g723|g723sf [app_lookupcidname.so] => (Look up CallerID Name from local database) == Registered application 'LookupCIDName' [app_setcdruserfield.so] => (CDR user field apps) == Registered application 'SetCDRUserField' == Registered application 'AppendCDRUserField' == Manager registered action SetCDRUserField [format_wav.so] => (Microsoft WAV format (8000hz Signed Linear)) == Registered file format wav, extension(s) wav [func_realtime.so] => (Read/Write values from a RealTime repository) == Registered custom function REALTIME [app_amd.so] => (Answering Machine Detection Application) == Parsing '/etc/asterisk/amd.conf': Found -- AMD defaults: initialSilence [2500] greeting [1500] afterGreetingSilence [800] totalAnalysisTime [5000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256] == Registered application 'AMD' [app_cdr.so] => (Tell Asterisk to not maintain a CDR for the current call) == Registered application 'NoCDR' [func_timeout.so] => (Channel timeout dialplan functions) == Registered custom function TIMEOUT [app_hasnewvoicemail.so] => (Indicator for whether a voice mailbox has messages in a given folder. == Registered custom function VMCOUNT == Registered application 'HasVoicemail' == Registered application 'HasNewVoicemail' [format_sln.so] => (Raw Signed Linear Audio support (SLN)) == Registered file format sln, extension(s) sln|raw [app_lookupblacklist.so] => (Look up Caller*ID name/number from blacklist database) == Registered custom function BLACKLIST == Registered application 'LookupBlacklist' [func_strings.so] => (String handling dialplan functions) == Registered custom function FIELDQTY == Registered custom function FILTER == Registered custom function REGEX == Registered custom function ARRAY == Registered custom function QUOTE == Registered custom function LEN == Registered custom function STRFTIME == Registered custom function STRPTIME == Registered custom function EVAL == Registered custom function KEYPADHASH == Registered custom function SPRINTF [app_ices.so] => (Encode and Stream via icecast and ices) == Registered application 'ICES' [app_disa.so] => (DISA (Direct Inward System Access) Application) == Registered application 'DISA' [app_queue.so] => (True Call Queueing) == Registered application 'Queue' == Manager registered action Queues == Manager registered action QueueStatus == Manager registered action QueueAdd == Manager registered action QueueRemove == Manager registered action QueuePause == Registered application 'AddQueueMember' == Registered application 'RemoveQueueMember' == Registered application 'PauseQueueMember' == Registered application 'UnpauseQueueMember' == Registered custom function QUEUEAGENTCOUNT == Registered custom function QUEUE_MEMBER_COUNT == Registered custom function QUEUE_MEMBER_LIST == Registered custom function QUEUE_WAITING_COUNT == Parsing '/etc/asterisk/queues.conf': Found Jun 20 18:21:56 NOTICE[21502]: app_queue.c:2903 reload_queue_members: Queue members successfully reloaded from database. [format_h263.so] => (Raw h263 data) == Registered file format h263, extension(s) h263 [func_md5.so] => (MD5 digest dialplan functions) == Registered custom function MD5 == Registered custom function CHECK_MD5 [func_sha1.so] => (SHA-1 computation dialplan function) == Registered custom function SHA1 [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator) == Parsing '/etc/asterisk/codecs.conf': Found -- codec_gsm: using generic PLC == Registered translator 'gsmtolin' from format gsm to slin, cost 1 == Registered translator 'lintogsm' from format slin to gsm, cost 1 [func_cut.so] => (Cut out information from a string) == Registered custom function CUT == Registered custom function SORT [cdr_odbc.so] => (ODBC CDR Backend) == Parsing '/etc/asterisk/cdr_odbc.conf': Found Jun 20 18:21:56 DEBUG[21502]: cdr_odbc.c:314 odbc_load_module: cdr_odbc: Logging uniqueid Jun 20 18:21:56 DEBUG[21502]: cdr_odbc.c:327 odbc_load_module: cdr_odbc: Logging in GMT -- cdr_odbc: dsn is debtnet -- cdr_odbc: username is sysprogress -- cdr_odbc: password is [secret] -- cdr_odbc: table is PUB.cdr [func_channel.so] => (Channel information dialplan function) == Registered custom function CHANNEL [func_global.so] => (Global variable dialplan functions) == Registered custom function GLOBAL [app_forkcdr.so] => (Fork The CDR into 2 separate entities.) == Registered application 'ForkCDR' [format_pcm.so] => (Raw/Sun uLaw/ALaw 8khz Audio support (PCM,PCMA,AU)) == Registered file format pcm, extension(s) pcm|ulaw|ul|mu == Registered file format alaw, extension(s) alaw|al == Registered file format au, extension(s) au [func_env.so] => (Environment/filesystem dialplan functions) == Registered custom function ENV == Registered custom function STAT [func_uri.so] => (URI encode/decode dialplan functions) == Registered custom function URIDECODE == Registered custom function URIENCODE [codec_ulaw.so] => (Mu-law Coder/Decoder) == Parsing '/etc/asterisk/codecs.conf': Found -- codec_ulaw: using generic PLC == Registered translator 'ulawtolin' from format ulaw to slin, cost 1 == Registered translator 'lintoulaw' from format slin to ulaw, cost 1 [app_settransfercapability.so] => (Set ISDN Transfer Capability) == Registered application 'SetTransferCapability' [cdr_csv.so] => (Comma Separated Values CDR Backend) == Parsing '/etc/asterisk/cdr.conf': Found Jun 20 18:21:56 DEBUG[21502]: cdr_csv.c:126 load_config: logging time in GMT Jun 20 18:21:56 DEBUG[21502]: cdr_csv.c:134 load_config: logging CDR field UNIQUEID Jun 20 18:21:56 DEBUG[21502]: cdr_csv.c:142 load_config: logging CDR user-defined field [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) == Primary D-Channel on span 1 up == Registered translator 'ilbctolin' from format ilbc to slin, cost 2 == Registered translator 'lintoilbc' from format slin to ilbc, cost 13 [app_parkandannounce.so] => (Call Parking and Announce Application) == Registered application 'ParkAndAnnounce' [codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder) == Parsing '/etc/asterisk/codecs.conf': Found -- codec_lpc10: using generic PLC == Registered translator 'lpc10tolin' from format lpc10 to slin, cost 2 == Registered translator 'lintolpc10' from format slin to lpc10, cost 3 [codec_a_mu.so] => (A-law and Mulaw direct Coder/Decoder) == Registered translator 'alawtoulaw' from format alaw to ulaw, cost 1 == Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 1 [app_dumpchan.so] => (Dump Info About The Calling Channel) == Registered application 'DumpChan' [cdr_pgsql.so] => (PostgreSQL CDR Backend) == Parsing '/etc/asterisk/cdr_pgsql.conf': Found [app_random.so] => (Random goto) == Registered application 'Random' [app_transfer.so] => (Transfer) == Registered application 'Transfer' [format_ilbc.so] => (Raw iLBC data) == Registered file format iLBC, extension(s) ilbc [app_chanisavail.so] => (Check channel availability) == Registered application 'ChanIsAvail' [app_zapras.so] => (Zap RAS Application) == Registered application 'ZapRAS' [app_speech_utils.so] => (Dialplan Speech Applications) == Registered application 'SpeechCreate' == Registered application 'SpeechLoadGrammar' == Registered application 'SpeechUnloadGrammar' == Registered application 'SpeechActivateGrammar' == Registered application 'SpeechDeactivateGrammar' == Registered application 'SpeechStart' == Registered application 'SpeechBackground' == Registered application 'SpeechDestroy' == Registered application 'SpeechProcessingSound' == Registered custom function SPEECH == Registered custom function SPEECH_SCORE == Registered custom function SPEECH_TEXT == Registered custom function SPEECH_GRAMMAR [func_groupcount.so] => (Channel group dialplan functions) == Registered custom function GROUP_COUNT == Registered custom function GROUP_MATCH_COUNT == Registered custom function GROUP_LIST == Registered custom function GROUP [format_jpeg.so] => (JPEG (Joint Picture Experts Group) Image Format) == Registered format 'jpg' (JPEG (Joint Picture Experts Group)) [app_page.so] => (Page Multiple Phones) == Registered application 'Page' [app_alarmreceiver.so] => (Alarm Receiver for Asterisk) == Parsing '/etc/asterisk/alarmreceiver.conf': Found == Registered application 'AlarmReceiver' [app_adsiprog.so] => (Asterisk ADSI Programming Application) == Registered application 'ADSIProg' [app_exec.so] => (Executes dialplan applications) == Registered application 'Exec' == Registered application 'TryExec' == Registered application 'ExecIf' [codec_alaw.so] => (A-law Coder/Decoder) == Parsing '/etc/asterisk/codecs.conf': Found -- codec_alaw: using generic PLC == Registered translator 'alawtolin' from format alaw to slin, cost 1 == Registered translator 'lintoalaw' from format slin to alaw, cost 1 [app_followme.so] => (Find-Me/Follow-Me Application) Jun 20 18:21:56 WARNING[21502]: app_followme.c:298 reload_followme: No follow me config file (followme.conf), so no follow me == Registered application 'FollowMe' [format_wav_gsm.so] => (Microsoft WAV format (Proprietary GSM)) == Registered file format wav49, extension(s) WAV|wav49 [app_userevent.so] => (Custom User Event Application) == Registered application 'UserEvent' [cdr_custom.so] => (Customizable Comma Separated Values CDR Backend) == Parsing '/etc/asterisk/cdr_custom.conf': Found [app_meetme.so] => (MeetMe conference bridge) == Parsing '/etc/asterisk/meetme.conf': Found == Manager registered action MeetmeMute == Manager registered action MeetmeUnmute == Registered application 'MeetMeAdmin' == Registered application 'MeetMeCount' == Registered application 'MeetMe' [func_callerid.so] => (Caller ID related dialplan function) == Registered custom function CALLERID [app_senddtmf.so] => (Send DTMF digits Application) == Manager registered action PlayDTMF == Registered application 'SendDTMF' [app_directed_pickup.so] => (Directed Call Pickup Application) == Registered application 'Pickup' [app_zapbarge.so] => (Barge in on Zap channel application) == Registered application 'ZapBarge' [app_chanspy.so] => (Listen to the audio of an active channel) == Registered application 'ChanSpy' [app_zapscan.so] => (Scan Zap channels application) == Registered application 'ZapScan' [app_playback.so] => (Sound File Playback Application) == Parsing '/etc/asterisk/say.conf': Found == Registered application 'Playback' [app_macro.so] => (Extension Macros) == Registered application 'MacroExit' == Registered application 'MacroIf' == Registered application 'Macro' [func_logic.so] => (Logical dialplan functions) == Registered custom function ISNULL == Registered custom function SET == Registered custom function EXISTS == Registered custom function IF == Registered custom function IFTIME [app_zapateller.so] => (Block Telemarketers with Special Information Tone) == Registered application 'Zapateller' [app_test.so] => (Interface Test Application) == Registered application 'TestClient' == Registered application 'TestServer' [app_echo.so] => (Simple Echo Application) == Registered application 'Echo' [skipping chan_alsa.so] [func_curl.so] => (Load external URL) == Registered custom function CURL [app_stack.so] => (Stack Routines) == Registered application 'StackPop' == Registered application 'Return' == Registered application 'GosubIf' == Registered application 'Gosub' [app_sms.so] => (SMS/PSTN handler) == Registered application 'SMS' [app_waitforring.so] => (Waits until first ring after time) == Registered application 'WaitForRing' [app_softhangup.so] => (Hangs up the requested channel) == Registered application 'SoftHangup' Jun 20 18:21:56 DEBUG[21502]: loader.c:389 fixup: ---- fixup (load_modules): 153 modules, 0 new --- Jun 20 18:21:56 DEBUG[21502]: loader.c:394 fixup: ---- fixup: cycle 0 --- Jun 20 18:21:56 DEBUG[21502]: loader.c:425 fixup: ---- fixup complete --- == Manager registered action DBGet == Manager registered action DBPut == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI> sip debug SIP Debugging enabled *CLI> set <-- SIP read from 192.168.0.200:52426: REGISTER sip:90.0.0.62 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK1a06037f From: sip:706@90.0.0.62 To: sip:706@90.0.0.62 Call-ID: 00078599-3d3d0003-7cc53c17-529aa999@192.168.0.200 Date: Tue, 20 Jun 2006 17:22:07 GMT CSeq: 71572 REGISTER User-Agent: CSCO/7 Contact: Content-Length: 0 Expires: 3600 --- (11 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.0.200 : 5060 (no NAT) Transmitting (no NAT) to 192.168.0.200:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK1a06037f;received=192.168.0.200 From: sip:706@90.0.0.62 To: sip:706@90.0.0.62 Call-ID: 00078599-3d3d0003-7cc53c17-529aa999@192.168.0.200 CSeq: 71572 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.0.200:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK1a06037f;received=192.168.0.200 From: sip:706@90.0.0.62 To: sip:706@90.0.0.62;tag=as2ec68228 Call-ID: 00078599-3d3d0003-7cc53c17-529aa999@192.168.0.200 CSeq: 71572 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: WWW-Authenticate: Digest realm="asterisk", nonce="2646a6e5" Content-Length: 0 --- Scheduling destruction of SIP dialog '00078599-3d3d0003-7cc53c17-529aa999@192.168.0.200' in 32000 ms (Method: REGISTER) de <-- SIP read from 192.168.0.200:52426: REGISTER sip:90.0.0.62 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK4689d42f From: sip:706@90.0.0.62 To: sip:706@90.0.0.62 Call-ID: 00078599-3d3d0003-7cc53c17-529aa999@192.168.0.200 Date: Tue, 20 Jun 2006 17:22:07 GMT CSeq: 71573 REGISTER User-Agent: CSCO/7 Contact: Authorization: Digest username="706",realm="asterisk",uri="sip:90.0.0.62",response="368c8aed89a14be42c8907d57c0a7385",nonce="2646a6e5",algorithm=md5 Content-Length: 0 Expires: 3600 --- (12 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.0.200 : 5060 (no NAT) Transmitting (no NAT) to 192.168.0.200:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK4689d42f;received=192.168.0.200 From: sip:706@90.0.0.62 To: sip:706@90.0.0.62 Call-ID: 00078599-3d3d0003-7cc53c17-529aa999@192.168.0.200 CSeq: 71573 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- -- Saved useragent "CSCO/7" for peer 706 Transmitting (no NAT) to 192.168.0.200:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK4689d42f;received=192.168.0.200 From: sip:706@90.0.0.62 To: sip:706@90.0.0.62;tag=as2ec68228 Call-ID: 00078599-3d3d0003-7cc53c17-529aa999@192.168.0.200 CSeq: 71573 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 3600 Contact: ;expires=3600 Date: Tue, 20 Jun 2006 17:22:08 GMT Content-Length: 0 --- Scheduling destruction of SIP dialog '00078599-3d3d0003-7cc53c17-529aa999@192.168.0.200' in 15000 ms (Method: REGISTER) bug 4 Core debug was 0 and is now 4 *CLI> <-- SIP read from 192.168.0.200:52426: INVITE sip:705@90.0.0.62 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK2fdcb952 From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 To: Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 Date: Tue, 20 Jun 2006 17:22:13 GMT CSeq: 101 INVITE User-Agent: CSCO/7 Contact: Expires: 180 Content-Type: application/sdp Content-Length: 249 Accept: application/sdp Remote-Party-ID: "jmls" ;party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 13817 23081 IN IP4 192.168.0.200 s=SIP Call c=IN IP4 192.168.0.200 t=0 0 m=audio 27686 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 0: INVITE sip:705@90.0.0.62 SIP/2.0 (32) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK2fdcb952 (58) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 2: From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 (70) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 3: To: (23) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 4: Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 (58) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 5: Date: Tue, 20 Jun 2006 17:22:13 GMT (35) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 6: CSeq: 101 INVITE (16) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 7: User-Agent: CSCO/7 (18) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 8: Contact: (37) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 9: Expires: 180 (12) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 10: Content-Type: application/sdp (29) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 11: Content-Length: 249 (19) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 12: Accept: application/sdp (23) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 13: Remote-Party-ID: "jmls" ;party=calling;id-type=subscriber;privacy=off;screen=no (102) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 14: (0) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: v=0 (3) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: o=Cisco-SIPUA 13817 23081 IN IP4 192.168.0.200 (46) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: s=SIP Call (10) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: c=IN IP4 192.168.0.200 (22) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: t=0 0 (5) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: m=audio 27686 RTP/AVP 0 8 18 101 (32) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=rtpmap:18 G729/8000 (21) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=fmtp:101 0-15 (15) --- (14 headers 11 lines)--- Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:3910 sip_alloc: Allocating new SIP dialog for 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 - INVITE (With RTP) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:13253 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 192.168.0.200 : 5060 (no NAT) Using INVITE request as basis request - 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:8072 check_user_full: Setting NAT on RTP to Off Reliably Transmitting (no NAT) to 192.168.0.200:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK2fdcb952;received=192.168.0.200 From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 To: ;tag=as638daa86 Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="2bec2a08" Content-Length: 0 --- Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:1816 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #16 Scheduling destruction of SIP dialog '00078599-3d3d03d2-51e05994-3286d044@192.168.0.200' in 32000 ms (Method: INVITE) Found user '706' <-- SIP read from 192.168.0.200:52768: ACK sip:705@90.0.0.62 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK2fdcb952 From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 To: ;tag=as638daa86 Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 Date: Tue, 20 Jun 2006 17:22:13 GMT CSeq: 101 ACK Content-Length: 0 Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 0: ACK sip:705@90.0.0.62 SIP/2.0 (29) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK2fdcb952 (58) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 2: From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 (70) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 3: To: ;tag=as638daa86 (38) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 4: Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 (58) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 5: Date: Tue, 20 Jun 2006 17:22:13 GMT (35) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 6: CSeq: 101 ACK (13) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 7: Content-Length: 0 (17) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:13253 handle_request: **** Received ACK (6) - Command in SIP ACK Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:1909 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16 Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:1919 __sip_ack: Stopping retransmission on '00078599-3d3d03d2-51e05994-3286d044@192.168.0.200' of Response 101: Match Not Found <-- SIP read from 192.168.0.200:52426: INVITE sip:705@90.0.0.62 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK783a0a70 From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 To: Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 Date: Tue, 20 Jun 2006 17:22:13 GMT CSeq: 102 INVITE User-Agent: CSCO/7 Contact: Proxy-Authorization: Digest username="706",realm="asterisk",uri="sip:90.0.0.62",response="188c68047438dccce1fd1b0b4f20b030",nonce="2bec2a08",algorithm=md5 Expires: 180 Content-Type: application/sdp Content-Length: 249 Remote-Party-ID: "jmls" ;party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 13817 23081 IN IP4 192.168.0.200 s=SIP Call c=IN IP4 192.168.0.200 t=0 0 m=audio 27686 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 0: INVITE sip:705@90.0.0.62 SIP/2.0 (32) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK783a0a70 (58) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 2: From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 (70) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 3: To: (23) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 4: Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 (58) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 5: Date: Tue, 20 Jun 2006 17:22:13 GMT (35) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 6: CSeq: 102 INVITE (16) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 7: User-Agent: CSCO/7 (18) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 8: Contact: (37) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 9: Proxy-Authorization: Digest username="706",realm="asterisk",uri="sip:90.0.0.62",response="188c68047438dccce1fd1b0b4f20b030",nonce="2bec2a08",algorithm=md5 (154) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 10: Expires: 180 (12) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 11: Content-Type: application/sdp (29) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 12: Content-Length: 249 (19) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 13: Remote-Party-ID: "jmls" ;party=calling;id-type=subscriber;privacy=off;screen=no (102) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 14: (0) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: v=0 (3) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: o=Cisco-SIPUA 13817 23081 IN IP4 192.168.0.200 (46) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: s=SIP Call (10) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: c=IN IP4 192.168.0.200 (22) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: t=0 0 (5) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: m=audio 27686 RTP/AVP 0 8 18 101 (32) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=rtpmap:18 G729/8000 (21) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=fmtp:101 0-15 (15) --- (14 headers 11 lines)--- Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:13253 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 192.168.0.200 : 5060 (no NAT) Using INVITE request as basis request - 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:8072 check_user_full: Setting NAT on RTP to Off Found user '706' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Found description format rtpmap:0 PCMU/8000 for ID 101 Found description format rtpmap:8 PCMA/8000 for ID 101 Found description format rtpmap:18 G729/8000 for ID 101 Found description format rtpmap:101 telephone-event/8000 for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264)/video=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264), 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 192.168.0.200:27686 Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:4545 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:12145 handle_request_invite: Checking SIP call limits for device 706 Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:2743 update_call_counter: Updating call counter for incoming call Looking for 705 in from-sip (domain 90.0.0.62) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:3456 sip_new: *** Our native formats are 0x4 (ulaw) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:3457 sip_new: *** Joint capabilities are 0xe (gsm|ulaw|alaw) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:3458 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:3459 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:3482 sip_new: This channel will not be able to handle video. Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:7037 build_route: build_route: Contact hop: list_route: hop: Jun 20 18:22:14 DEBUG[21518]: chan_sip.c:12215 handle_request_invite: SIP/706-0e7b: New call is still down.... Trying... Transmitting (no NAT) to 192.168.0.200:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK783a0a70;received=192.168.0.200 From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 To: Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- Jun 20 18:22:14 DEBUG[21506]: chan_sip.c:13820 sip_devicestate: Checking device state for peer 706 Jun 20 18:22:14 DEBUG[21506]: devicestate.c:189 do_state_change: Changing state for SIP/706 - state 1 (Not in use) Jun 20 18:22:14 DEBUG[21534]: pbx.c:1675 pbx_extension_helper: Launching 'Answer' -- Executing [705@from-sip:1] Answer("SIP/706-0e7b", "") in new stack Jun 20 18:22:14 DEBUG[21506]: chan_sip.c:13820 sip_devicestate: Checking device state for peer 706 Jun 20 18:22:14 DEBUG[21506]: devicestate.c:189 do_state_change: Changing state for SIP/706 - state 1 (Not in use) Jun 20 18:22:14 DEBUG[21534]: chan_sip.c:3184 sip_answer: SIP answering channel: SIP/706-0e7b Jun 20 18:22:14 DEBUG[21534]: chan_sip.c:5306 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True Jun 20 18:22:14 DEBUG[21534]: chan_sip.c:5307 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 90.0.0.62 port 17250 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Jun 20 18:22:14 DEBUG[21534]: chan_sip.c:5452 add_sdp: -- Done with adding codecs to SDP Jun 20 18:22:14 DEBUG[21534]: chan_sip.c:5491 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) Reliably Transmitting (no NAT) to 192.168.0.200:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK783a0a70;received=192.168.0.200 From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 To: ;tag=as0388523b Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 269 v=0 o=root 21502 21502 IN IP4 90.0.0.62 s=session c=IN IP4 90.0.0.62 t=0 0 m=audio 17250 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- Jun 20 18:22:14 DEBUG[21535]: app_queue.c:557 changethread: Device 'SIP/706' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Jun 20 18:22:14 DEBUG[21536]: app_queue.c:557 changethread: Device 'SIP/706' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Jun 20 18:22:14 DEBUG[21534]: chan_sip.c:1816 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #18 Jun 20 18:22:14 DEBUG[21534]: pbx.c:1675 pbx_extension_helper: Launching 'MusicOnHold' -- Executing [705@from-sip:2] MusicOnHold("SIP/706-0e7b", "") in new stack -- Started music on hold, class 'default', on SIP/706-0e7b Jun 20 18:22:14 DEBUG[21534]: channel.c:1792 ast_settimeout: Scheduling timer at 160 sample intervals Jun 20 18:22:14 DEBUG[21534]: channel.c:2487 set_format: Set channel SIP/706-0e7b to write format ulaw Jun 20 18:22:14 DEBUG[21534]: res_musiconhold.c:251 ast_moh_files_next: SIP/706-0e7b Opened file 11 '/var/lib/asterisk/sounds/custom/moh/track20' Jun 20 18:22:14 DEBUG[21534]: rtp.c:2337 ast_rtp_write: Ooh, format changed from unknown to ulaw Jun 20 18:22:15 DEBUG[21534]: channel.c:2072 __ast_read: Generator got voice, switching to phase locked mode Jun 20 18:22:15 DEBUG[21534]: channel.c:1792 ast_settimeout: Scheduling timer at 0 sample intervals <-- SIP read from 192.168.0.200:52426: ACK sip:705@90.0.0.62:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK30a0cef5 From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 To: ;tag=as0388523b Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 Date: Tue, 20 Jun 2006 17:22:14 GMT CSeq: 102 ACK User-Agent: CSCO/7 Proxy-Authorization: Digest username="706",realm="asterisk",uri="sip:90.0.0.62",response="188c68047438dccce1fd1b0b4f20b030",nonce="2bec2a08",algorithm=md5 Content-Length: 0 Jun 20 18:22:15 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 0: ACK sip:705@90.0.0.62:5060 SIP/2.0 (34) Jun 20 18:22:15 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK30a0cef5 (58) Jun 20 18:22:15 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 2: From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 (70) Jun 20 18:22:15 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 3: To: ;tag=as0388523b (38) Jun 20 18:22:15 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 4: Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 (58) Jun 20 18:22:15 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 5: Date: Tue, 20 Jun 2006 17:22:14 GMT (35) Jun 20 18:22:15 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 6: CSeq: 102 ACK (13) Jun 20 18:22:15 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 7: User-Agent: CSCO/7 (18) Jun 20 18:22:15 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 8: Proxy-Authorization: Digest username="706",realm="asterisk",uri="sip:90.0.0.62",response="188c68047438dccce1fd1b0b4f20b030",nonce="2bec2a08",algorithm=md5 (154) Jun 20 18:22:15 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 9: Content-Length: 0 (17) Jun 20 18:22:15 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 10: (0) --- (10 headers 0 lines)--- Jun 20 18:22:15 DEBUG[21518]: chan_sip.c:13253 handle_request: **** Received ACK (6) - Command in SIP ACK Jun 20 18:22:15 DEBUG[21518]: chan_sip.c:1909 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #18 Jun 20 18:22:15 DEBUG[21518]: chan_sip.c:1919 __sip_ack: Stopping retransmission on '00078599-3d3d03d2-51e05994-3286d044@192.168.0.200' of Response 102: Match Not Found Jun 20 18:22:15 DEBUG[21534]: rtp.c:2236 ast_rtp_raw_write: Difference is 896, ms is 132 Jun 20 18:22:16 DEBUG[21534]: rtp.c:2236 ast_rtp_raw_write: Difference is 2096, ms is 282 Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:3910 sip_alloc: Allocating new SIP dialog for (No Call-ID) - NOTIFY (No RTP) Scheduling destruction of SIP dialog '64b21dc717ea0f653111aec66f034f1f@90.0.0.62' in 32000 ms (Method: NOTIFY) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 0: NOTIFY sip:706@192.168.0.200:5060 SIP/2.0 (41) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 1: Via: SIP/2.0/UDP 90.0.0.62:5060;branch=z9hG4bK7755f1db;rport (60) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 2: From: "asterisk" ;tag=as656ad799 (56) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 3: To: (32) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 4: Contact: (33) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 5: Call-ID: 64b21dc717ea0f653111aec66f034f1f@90.0.0.62 (51) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 6: CSeq: 102 NOTIFY (16) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 7: User-Agent: Asterisk PBX (24) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 8: Max-Forwards: 70 (16) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 9: Event: message-summary (22) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 10: Content-Type: application/simple-message-summary (48) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 11: Content-Length: 89 (18) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 12: (0) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: Messages-Waiting: no (20) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: Message-Account: sip:asterisk@90.0.0.62 (39) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: Voice-Message: 0/0 (0/0) (24) Reliably Transmitting (no NAT) to 192.168.0.200:5060: NOTIFY sip:706@192.168.0.200:5060 SIP/2.0 Via: SIP/2.0/UDP 90.0.0.62:5060;branch=z9hG4bK7755f1db;rport From: "asterisk" ;tag=as656ad799 To: Contact: Call-ID: 64b21dc717ea0f653111aec66f034f1f@90.0.0.62 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 89 Messages-Waiting: no Message-Account: sip:asterisk@90.0.0.62 Voice-Message: 0/0 (0/0) --- Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:1816 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #21 <-- SIP read from 192.168.0.200:52769: SIP/2.0 200 OK Via: SIP/2.0/UDP 90.0.0.62:5060;branch=z9hG4bK7755f1db;rport From: "asterisk" ;tag=as656ad799 To: Call-ID: 64b21dc717ea0f653111aec66f034f1f@90.0.0.62 Date: Tue, 20 Jun 2006 17:22:17 GMT CSeq: 102 NOTIFY Content-Length: 0 Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 0: SIP/2.0 200 OK (14) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 1: Via: SIP/2.0/UDP 90.0.0.62:5060;branch=z9hG4bK7755f1db;rport (60) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 2: From: "asterisk" ;tag=as656ad799 (56) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 3: To: (32) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 4: Call-ID: 64b21dc717ea0f653111aec66f034f1f@90.0.0.62 (51) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 5: Date: Tue, 20 Jun 2006 17:22:17 GMT (35) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 6: CSeq: 102 NOTIFY (16) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 7: Content-Length: 0 (17) Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:1909 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #21 Jun 20 18:22:18 DEBUG[21518]: chan_sip.c:1919 __sip_ack: Stopping retransmission on '64b21dc717ea0f653111aec66f034f1f@90.0.0.62' of Request 102: Match Not Found Really destroying SIP dialog '64b21dc717ea0f653111aec66f034f1f@90.0.0.62' Method: NOTIFY Jun 20 18:22:23 DEBUG[21518]: chan_sip.c:1848 __sip_autodestruct: Auto destroying call '00078599-3d3d0003-7cc53c17-529aa999@192.168.0.200' Jun 20 18:22:23 DEBUG[21518]: chan_sip.c:2849 sip_destroy: Destroying SIP dialog 00078599-3d3d0003-7cc53c17-529aa999@192.168.0.200 Really destroying SIP dialog '00078599-3d3d0003-7cc53c17-529aa999@192.168.0.200' Method: REGISTER <-- SIP read from 192.168.0.200:52426: INVITE sip:705@90.0.0.62:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK76e45681 From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 To: ;tag=as0388523b Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 Date: Tue, 20 Jun 2006 17:22:28 GMT CSeq: 103 INVITE User-Agent: CSCO/7 Contact: Content-Type: application/sdp Content-Length: 241 Remote-Party-ID: "jmls" ;party=calling;id-type=subscriber;privacy=off;screen=no Proxy-Authorization: Digest username="706",realm="asterisk",uri="sip:90.0.0.62",response="188c68047438dccce1fd1b0b4f20b030",nonce="2bec2a08",algorithm=md5 v=0 o=Cisco-SIPUA 23753 897 IN IP4 192.168.0.200 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 27686 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 0: INVITE sip:705@90.0.0.62:5060 SIP/2.0 (37) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK76e45681 (58) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 2: From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 (70) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 3: To: ;tag=as0388523b (38) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 4: Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 (58) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 5: Date: Tue, 20 Jun 2006 17:22:28 GMT (35) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 6: CSeq: 103 INVITE (16) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 7: User-Agent: CSCO/7 (18) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 8: Contact: (37) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 9: Content-Type: application/sdp (29) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 10: Content-Length: 241 (19) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 11: Remote-Party-ID: "jmls" ;party=calling;id-type=subscriber;privacy=off;screen=no (102) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 12: Proxy-Authorization: Digest username="706",realm="asterisk",uri="sip:90.0.0.62",response="188c68047438dccce1fd1b0b4f20b030",nonce="2bec2a08",algorithm=md5 (154) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 13: (0) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: v=0 (3) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: o=Cisco-SIPUA 23753 897 IN IP4 192.168.0.200 (44) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: s=SIP Call (10) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: c=IN IP4 0.0.0.0 (16) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: t=0 0 (5) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: m=audio 27686 RTP/AVP 0 8 18 101 (32) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=rtpmap:18 G729/8000 (21) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=fmtp:101 0-15 (15) --- (13 headers 11 lines)--- Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:13253 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 192.168.0.200 : 5060 (no NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Found description format rtpmap:0 PCMU/8000 for ID 101 Found description format rtpmap:8 PCMA/8000 for ID 101 Found description format rtpmap:18 G729/8000 for ID 101 Found description format rtpmap:101 telephone-event/8000 for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264)/video=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264), 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 0.0.0.0:27686 Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4545 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:12193 handle_request_invite: Got a SIP re-invite for call 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:12282 handle_request_invite: SIP/706-0e7b: New call is UP.... Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:5306 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:5307 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 90.0.0.62 port 17250 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:5452 add_sdp: -- Done with adding codecs to SDP Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:5491 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) Reliably Transmitting (no NAT) to 192.168.0.200:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK76e45681;received=192.168.0.200 From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 To: ;tag=as0388523b Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 269 v=0 o=root 21502 21503 IN IP4 90.0.0.62 s=session c=IN IP4 90.0.0.62 t=0 0 m=audio 17250 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=recvonly --- Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:1816 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #22 <-- SIP read from 192.168.0.200:52426: ACK sip:705@90.0.0.62:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK157d5706 From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 To: ;tag=as0388523b Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 Date: Tue, 20 Jun 2006 17:22:28 GMT CSeq: 103 ACK User-Agent: CSCO/7 Proxy-Authorization: Digest username="706",realm="asterisk",uri="sip:90.0.0.62",response="188c68047438dccce1fd1b0b4f20b030",nonce="2bec2a08",algorithm=md5 Content-Length: 0 Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 0: ACK sip:705@90.0.0.62:5060 SIP/2.0 (34) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK157d5706 (58) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 2: From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 (70) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 3: To: ;tag=as0388523b (38) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 4: Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 (58) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 5: Date: Tue, 20 Jun 2006 17:22:28 GMT (35) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 6: CSeq: 103 ACK (13) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 7: User-Agent: CSCO/7 (18) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 8: Proxy-Authorization: Digest username="706",realm="asterisk",uri="sip:90.0.0.62",response="188c68047438dccce1fd1b0b4f20b030",nonce="2bec2a08",algorithm=md5 (154) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 9: Content-Length: 0 (17) Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 10: (0) --- (10 headers 0 lines)--- Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:13253 handle_request: **** Received ACK (6) - Command in SIP ACK Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:1909 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #22 Jun 20 18:22:29 DEBUG[21518]: chan_sip.c:1919 __sip_ack: Stopping retransmission on '00078599-3d3d03d2-51e05994-3286d044@192.168.0.200' of Response 103: Match Not Found RTCP SR transmission error, rtcp halted No such file or directory Jun 20 18:22:29 NOTICE[21518]: sched.c:287 ast_sched_del: Attempted to delete nonexistent schedule entry 19! <-- SIP read from 192.168.0.200:52426: INVITE sip:705@90.0.0.62:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK7da8d5f4 From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 To: ;tag=as0388523b Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 Date: Tue, 20 Jun 2006 17:22:32 GMT CSeq: 104 INVITE User-Agent: CSCO/7 Contact: Content-Type: application/sdp Content-Length: 247 Remote-Party-ID: "jmls" ;party=calling;id-type=subscriber;privacy=off;screen=no Proxy-Authorization: Digest username="706",realm="asterisk",uri="sip:90.0.0.62",response="188c68047438dccce1fd1b0b4f20b030",nonce="2bec2a08",algorithm=md5 v=0 o=Cisco-SIPUA 861 20866 IN IP4 192.168.0.200 s=SIP Call c=IN IP4 192.168.0.200 t=0 0 m=audio 27686 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 0: INVITE sip:705@90.0.0.62:5060 SIP/2.0 (37) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK7da8d5f4 (58) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 2: From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 (70) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 3: To: ;tag=as0388523b (38) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 4: Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 (58) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 5: Date: Tue, 20 Jun 2006 17:22:32 GMT (35) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 6: CSeq: 104 INVITE (16) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 7: User-Agent: CSCO/7 (18) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 8: Contact: (37) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 9: Content-Type: application/sdp (29) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 10: Content-Length: 247 (19) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 11: Remote-Party-ID: "jmls" ;party=calling;id-type=subscriber;privacy=off;screen=no (102) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 12: Proxy-Authorization: Digest username="706",realm="asterisk",uri="sip:90.0.0.62",response="188c68047438dccce1fd1b0b4f20b030",nonce="2bec2a08",algorithm=md5 (154) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 13: (0) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: v=0 (3) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: o=Cisco-SIPUA 861 20866 IN IP4 192.168.0.200 (44) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: s=SIP Call (10) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: c=IN IP4 192.168.0.200 (22) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: t=0 0 (5) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: m=audio 27686 RTP/AVP 0 8 18 101 (32) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=rtpmap:18 G729/8000 (21) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4162 parse_request: Line: a=fmtp:101 0-15 (15) --- (13 headers 11 lines)--- Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:13253 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 192.168.0.200 : 5060 (no NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Found description format rtpmap:0 PCMU/8000 for ID 101 Found description format rtpmap:8 PCMA/8000 for ID 101 Found description format rtpmap:18 G729/8000 for ID 101 Found description format rtpmap:101 telephone-event/8000 for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264)/video=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264), 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 192.168.0.200:27686 Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4545 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:12193 handle_request_invite: Got a SIP re-invite for call 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:12282 handle_request_invite: SIP/706-0e7b: New call is UP.... Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:5306 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:5307 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 90.0.0.62 port 17250 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:5452 add_sdp: -- Done with adding codecs to SDP Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:5491 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) Reliably Transmitting (no NAT) to 192.168.0.200:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK7da8d5f4;received=192.168.0.200 From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 To: ;tag=as0388523b Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 269 v=0 o=root 21502 21504 IN IP4 90.0.0.62 s=session c=IN IP4 90.0.0.62 t=0 0 m=audio 17250 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:1816 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #23 Jun 20 18:22:33 DEBUG[21534]: rtp.c:2236 ast_rtp_raw_write: Difference is 35536, ms is 4462 <-- SIP read from 192.168.0.200:52426: ACK sip:705@90.0.0.62:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK694a37e4 From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 To: ;tag=as0388523b Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 Date: Tue, 20 Jun 2006 17:22:33 GMT CSeq: 104 ACK User-Agent: CSCO/7 Proxy-Authorization: Digest username="706",realm="asterisk",uri="sip:90.0.0.62",response="188c68047438dccce1fd1b0b4f20b030",nonce="2bec2a08",algorithm=md5 Content-Length: 0 Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 0: ACK sip:705@90.0.0.62:5060 SIP/2.0 (34) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK694a37e4 (58) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 2: From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 (70) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 3: To: ;tag=as0388523b (38) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 4: Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 (58) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 5: Date: Tue, 20 Jun 2006 17:22:33 GMT (35) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 6: CSeq: 104 ACK (13) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 7: User-Agent: CSCO/7 (18) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 8: Proxy-Authorization: Digest username="706",realm="asterisk",uri="sip:90.0.0.62",response="188c68047438dccce1fd1b0b4f20b030",nonce="2bec2a08",algorithm=md5 (154) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 9: Content-Length: 0 (17) Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 10: (0) --- (10 headers 0 lines)--- Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:13253 handle_request: **** Received ACK (6) - Command in SIP ACK Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:1909 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #23 Jun 20 18:22:33 DEBUG[21518]: chan_sip.c:1919 __sip_ack: Stopping retransmission on '00078599-3d3d03d2-51e05994-3286d044@192.168.0.200' of Response 104: Match Not Found Jun 20 18:22:34 DEBUG[21534]: rtp.c:2236 ast_rtp_raw_write: Difference is 2360, ms is 315 Jun 20 18:22:34 DEBUG[21534]: rtp.c:2236 ast_rtp_raw_write: Difference is 3584, ms is 468 <-- SIP read from 192.168.0.200:52426: BYE sip:705@90.0.0.62:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK56ada1be From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 To: ;tag=as0388523b Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 Date: Tue, 20 Jun 2006 17:22:40 GMT CSeq: 105 BYE User-Agent: CSCO/7 Content-Length: 0 Proxy-Authorization: Digest username="706",realm="asterisk",uri="sip:90.0.0.62",response="a559b1498b9828a63413b8d83a58062b",nonce="2bec2a08",algorithm=md5 Jun 20 18:22:40 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 0: BYE sip:705@90.0.0.62:5060 SIP/2.0 (34) Jun 20 18:22:40 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK56ada1be (58) Jun 20 18:22:40 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 2: From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 (70) Jun 20 18:22:40 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 3: To: ;tag=as0388523b (38) Jun 20 18:22:40 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 4: Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 (58) Jun 20 18:22:40 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 5: Date: Tue, 20 Jun 2006 17:22:40 GMT (35) Jun 20 18:22:40 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 6: CSeq: 105 BYE (13) Jun 20 18:22:40 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 7: User-Agent: CSCO/7 (18) Jun 20 18:22:40 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 8: Content-Length: 0 (17) Jun 20 18:22:40 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 9: Proxy-Authorization: Digest username="706",realm="asterisk",uri="sip:90.0.0.62",response="a559b1498b9828a63413b8d83a58062b",nonce="2bec2a08",algorithm=md5 (154) Jun 20 18:22:40 DEBUG[21518]: chan_sip.c:4130 parse_request: Header 10: (0) --- (10 headers 0 lines)--- Jun 20 18:22:40 DEBUG[21518]: chan_sip.c:13253 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 192.168.0.200 : 5060 (no NAT) Transmitting (no NAT) to 192.168.0.200:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK56ada1be;received=192.168.0.200 From: "jmls" ;tag=000785993d3d051c3b460e40-6d9307c2 To: ;tag=as0388523b Call-ID: 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200 CSeq: 105 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- -- Stopped music on hold on SIP/706-0e7b Jun 20 18:22:40 DEBUG[21534]: channel.c:2487 set_format: Set channel SIP/706-0e7b to write format ulaw Jun 20 18:22:40 DEBUG[21534]: channel.c:1792 ast_settimeout: Scheduling timer at 0 sample intervals Jun 20 18:22:40 DEBUG[21534]: pbx.c:2258 __ast_pbx_run: Spawn extension (from-sip,705,2) exited non-zero on 'SIP/706-0e7b' Jun 20 18:22:40 DEBUG[21534]: pbx.c:1675 pbx_extension_helper: Launching 'NoOp' -- Executing [h@from-sip:1] NoOp("SIP/706-0e7b", "Reached Hangup Extension") in new stack Jun 20 18:22:40 DEBUG[21534]: pbx.c:1675 pbx_extension_helper: Launching 'Hangup' -- Executing [h@from-sip:2] Hangup("SIP/706-0e7b", "") in new stack Jun 20 18:22:40 DEBUG[21534]: pbx.c:2378 __ast_pbx_run: Spawn extension (from-sip,h,2) exited non-zero on 'SIP/706-0e7b' Jun 20 18:22:40 DEBUG[21534]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '"Julian Lyndon-Smith" <706>' Jun 20 18:22:40 DEBUG[21534]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '706' Jun 20 18:22:40 DEBUG[21534]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '705' Jun 20 18:22:40 DEBUG[21534]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'from-sip' Jun 20 18:22:40 DEBUG[21534]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'SIP/706-0e7b' Jun 20 18:22:40 DEBUG[21534]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '' Jun 20 18:22:40 DEBUG[21534]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'Hangup' Jun 20 18:22:40 DEBUG[21534]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '' Jun 20 18:22:40 DEBUG[21534]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '2006-06-20 18:22:14' Jun 20 18:22:40 DEBUG[21534]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '2006-06-20 18:22:14' Jun 20 18:22:40 DEBUG[21534]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '2006-06-20 18:22:40' Jun 20 18:22:40 DEBUG[21534]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '26' Jun 20 18:22:40 DEBUG[21534]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '26' Jun 20 18:22:40 DEBUG[21534]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' Jun 20 18:22:40 DEBUG[21534]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' Jun 20 18:22:40 DEBUG[21534]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '' Jun 20 18:22:40 DEBUG[21534]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '1150824134.0' Jun 20 18:22:40 DEBUG[21534]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '' Jun 20 18:22:40 DEBUG[21534]: channel.c:1492 ast_hangup: Hanging up channel 'SIP/706-0e7b' Jun 20 18:22:40 DEBUG[21534]: chan_sip.c:3043 sip_hangup: Hangup call SIP/706-0e7b, SIP callid 00078599-3d3d03d2-51e05994-3286d044@192.168.0.200) Jun 20 18:22:40 DEBUG[21534]: chan_sip.c:3051 sip_hangup: update_call_counter(706) - decrement call limit counter on hangup Jun 20 18:22:40 DEBUG[21534]: chan_sip.c:2743 update_call_counter: Updating call counter for incoming call Jun 20 18:22:40 DEBUG[21506]: chan_sip.c:13820 sip_devicestate: Checking device state for peer 706 Jun 20 18:22:40 DEBUG[21506]: devicestate.c:189 do_state_change: Changing state for SIP/706 - state 1 (Not in use) Jun 20 18:22:40 DEBUG[21537]: app_queue.c:557 changethread: Device 'SIP/706' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Really destroying SIP dialog '00078599-3d3d03d2-51e05994-3286d044@192.168.0.200' Method: BYE stop now Beginning asterisk shutdown.... Executing last minute cleanups == Destroying musiconhold processes Asterisk cleanly ending (0). [root@foxtrot asterisk]#