/l3/users/olef-alex/ais-voip-2011-jun/linux1.unix.ais/root :1 :2 :3 :4 :5 :6 :7 :8 :9 :10 :11 :12 :13 |
|
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- Executing [1103@grp1:3] GotoIf("SIP/1102-0000008b", "0)?voicemail") in new stack -- Executing [1103@grp1:4] GotoIf("SIP/1102-0000008b", "0)?voicemail") in new stack -- Executing [1103@grp1:5] GotoIf("SIP/1102-0000008b", "1)?voicemail") in new stack -- Goto (grp1,1103,7) -- Executing [1103@grp1:7] VoiceMail("SIP/1102-0000008b", "1103@default") in new stack [Jun 29 11:00:34] WARNING[8928]: app_voicemail.c:5260 leave_voicemail: No entry in voicemail config file for '1103' -- Executing [1103@grp1:8] Hangup("SIP/1102-0000008b", "") in new stack == Spawn extension (grp1, 1103, 8) exited non-zero on 'SIP/1102-0000008b' linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- Executing [1103@grp1:3] GotoIf("SIP/1102-0000008b", "0)?voicemail") in new stack -- Executing [1103@grp1:4] GotoIf("SIP/1102-0000008b", "0)?voicemail") in new stack -- Executing [1103@grp1:5] GotoIf("SIP/1102-0000008b", "1)?voicemail") in new stack -- Goto (grp1,1103,7) -- Executing [1103@grp1:7] VoiceMail("SIP/1102-0000008b", "1103@default") in new stack [Jun 29 11:00:34] WARNING[8928]: app_voicemail.c:5260 leave_voicemail: No entry in voicemail config file for '1103' -- Executing [1103@grp1:8] Hangup("SIP/1102-0000008b", "") in new stack == Spawn extension (grp1, 1103, 8) exited non-zero on 'SIP/1102-0000008b' linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- Executing [1103@grp1:3] GotoIf("SIP/1102-0000008b", "0)?voicemail") in new stack -- Executing [1103@grp1:4] GotoIf("SIP/1102-0000008b", "0)?voicemail") in new stack -- Executing [1103@grp1:5] GotoIf("SIP/1102-0000008b", "1)?voicemail") in new stack -- Goto (grp1,1103,7) -- Executing [1103@grp1:7] VoiceMail("SIP/1102-0000008b", "1103@default") in new stack [Jun 29 11:00:34] WARNING[8928]: app_voicemail.c:5260 leave_voicemail: No entry in voicemail config file for '1103' -- Executing [1103@grp1:8] Hangup("SIP/1102-0000008b", "") in new stack == Spawn extension (grp1, 1103, 8) exited non-zero on 'SIP/1102-0000008b' linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- Executing [1103@grp1:3] GotoIf("SIP/1102-0000008b", "0)?voicemail") in new stack -- Executing [1103@grp1:4] GotoIf("SIP/1102-0000008b", "0)?voicemail") in new stack -- Executing [1103@grp1:5] GotoIf("SIP/1102-0000008b", "1)?voicemail") in new stack -- Goto (grp1,1103,7) -- Executing [1103@grp1:7] VoiceMail("SIP/1102-0000008b", "1103@default") in new stack [Jun 29 11:00:34] WARNING[8928]: app_voicemail.c:5260 leave_voicemail: No entry in voicemail config file for '1103' -- Executing [1103@grp1:8] Hangup("SIP/1102-0000008b", "") in new stack == Spawn extension (grp1, 1103, 8) exited non-zero on 'SIP/1102-0000008b' linux1*CLI> exit Executing last minute cleanups |
#vim voicemail.conf
--- /tmp/l3-saved-6655.4395.18548 2011-06-29 11:02:55.000000000 +0400 +++ voicemail.conf 2011-06-29 11:06:51.000000000 +0400 @@ -329,6 +329,7 @@ [default] 1102 => 4242,Olefirenko,user@linux1.unix.ais +1103 => 4242,Olefirenko,user@linux1.unix.ais ;4200 => 9855,Mark Spencer,markster@linux-support.net,mypager@digium.com,attach=no|serveremail=myaddy@digium.com|tz=central|maxmsg=10 ;4300 => 3456,Ben Rigas,ben@american-computer.net |
#vim sip.conf
--- /tmp/l3-saved-6655.22581.175 2011-06-29 11:07:12.000000000 +0400 +++ sip.conf 2011-06-29 11:08:02.000000000 +0400 @@ -16,6 +16,7 @@ canreinvite=no context=grp1 language=ru +mailbox=1102 callerid = " Alex Olefirenko <1102>" [1103] @@ -26,6 +27,8 @@ callerid = "Alex Olefirenko <1103>" context=grp2 language=ru +mailbox=1103 + [1104] type=friend |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... == Parsing '/etc/asterisk/voicemail.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found [Jun 29 11:08:26] NOTICE[1675]: chan_sip.c:21599 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1102 linux1*CLI> sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found == Parsing '/etc/asterisk/sip_notify.conf': == Found linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... == Parsing '/etc/asterisk/voicemail.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found [Jun 29 11:08:26] NOTICE[1675]: chan_sip.c:21599 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1102 linux1*CLI> sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found == Parsing '/etc/asterisk/sip_notify.conf': == Found linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... == Parsing '/etc/asterisk/voicemail.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found [Jun 29 11:08:26] NOTICE[1675]: chan_sip.c:21599 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1102 linux1*CLI> sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found == Parsing '/etc/asterisk/sip_notify.conf': == Found linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... == Parsing '/etc/asterisk/voicemail.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found [Jun 29 11:08:26] NOTICE[1675]: chan_sip.c:21599 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1102 linux1*CLI> sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found == Parsing '/etc/asterisk/sip_notify.conf': == Found linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... == Parsing '/etc/asterisk/voicemail.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found [Jun 29 11:08:26] NOTICE[1675]: chan_sip.c:21599 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1102 linux1*CLI> sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found == Parsing '/etc/asterisk/sip_notify.conf': == Found linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... == Parsing '/etc/asterisk/voicemail.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found [Jun 29 11:08:26] NOTICE[1675]: chan_sip.c:21599 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1102 linux1*CLI> sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found == Parsing '/etc/asterisk/sip_notify.conf': == Found linux1*CLI> exit Executing last minute cleanups |
#~
[ ] exten =>8000,1,Playback(limit-simul-calls) exten => _11XX,1,AnswerIP/${EXTEN},15) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"])?voicemail) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "CONGESTION"])?voicemail) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"])?voicemail) exten => _11XX,n,Hangup exten => _11XX,n,Hangup exten => _11XX,n(voicemail),Voicemail(${EXTEN}@default) [national] [international] exten => _21XX,1,Dial(SIP/msk/${EXTEN}) "extensions.conf" 49L, 961C записано |
#~
[ ] [xten => ]11XX,n,GotoIf($["${DIALSTATUS}" = "CONGESTION"])?voicemail) [ ] exten => _11XX,n,Hangup exten => _11XX,n(voicemail),Voicemail(${EXTEN}@default) local] exten => _11XX,1,Answer 18C [ ] exten => _11XX,n,Dial(SIP/${EXTEN},15) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"])?voicemail) ... exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"])?voicemail) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "CONGESTION"])?voicemail) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"])?voicemail) exten => _11XX,n,Hangup exten => _11XX,n,Hangup exten => _11XX,n(voicemail),Voicemail(${EXTEN}@default) [national] [international] exten => _21XX,1,Dial(SIP/msk/${EXTEN}) "extensions.conf" 49L, 961C запиÑано |
#~
exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "CONGESTION"])?voicemail) exten => _11XX,n,Hangup exten => _11XX,n,Hangup [national] exten => _12XX,1,Dial(SIP/msk/${EXTEN}) exten => _32XX,1,Dial(SIP/msk/${EXTEN}) [grp1] include => local exten => _31XX,1,Dial(SIP/msk/${EXTEN}) 33,1 32% [grp2] [grp2] include => nationalional 17,1 64% [ ] exten =>8000,1,Playback(limit-simul-calls) exten => _11XX,1,AnswerIP/${EXTEN},15) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"])?voicemail) exten => _11XX,n(voicemail),Voicemail(${EXTEN}@default) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"])?voicemail) [international] exten => _21XX,1,Dial(SIP/msk/${EXTEN}) "extensions.conf" 49L, 961C записано |
#~
[ ] exten =>8000,1,Playback(limit-simul-calls) exten => _11XX,1,AnswerIP/${EXTEN},15) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"])?voicemail) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "CONGESTION"])?voicemail) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"])?voicemail) exten => _11XX,n,Hangup exten => _11XX,n,Hangup exten => _11XX,n(voicemail),Voicemail(${EXTEN}@default) [national] [international] exten => _21XX,1,Dial(SIP/msk/${EXTEN}) "extensions.conf" 49L, 961C записано |
#~
exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"])?voicemail) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"])?voicemail) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "CONGESTION"])?voicemail) exten => _11XX,n,Hangup exten => _11XX,n(voicemail),Voicemail(${EXTEN}@default) exten => _11XX,n,Hangup "extensions.conf" 49L, 918C [national] exten => _12XX,1,Dial(SIP/msk/${EXTEN}) [ ] ... exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"])?voicemail) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "CONGESTION"])?voicemail) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"])?voicemail) exten => _11XX,n,Hangup exten => _11XX,n,Hangup exten => _11XX,n(voicemail),Voicemail(${EXTEN}@default) [national] [international] exten => _21XX,1,Dial(SIP/msk/${EXTEN}) "extensions.conf" 49L, 961C записано |
#~
exten => _11XX,n,Hangup exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "CONGESTION"])?voicemail) exten => _11XX,n,Hangup exten => _11XX,n(voicemail),Voicemail(${EXTEN}@default) exten => _11XX,n,Hangup [ ] -- ВСТАВКА -- 21,1 Наверху exten => _21XX,1,Dial(SIP/msk/${EXTEN}) [ ] 27,1 8% exten => _22XX,1,Dial(SIP/msk/${EXTEN}) ... exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"])?voicemail) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "CONGESTION"])?voicemail) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"])?voicemail) exten => _11XX,n,Hangup exten => _11XX,n,Hangup exten => _11XX,n(voicemail),Voicemail(${EXTEN}@default) [national] [international] exten => _21XX,1,Dial(SIP/msk/${EXTEN}) "extensions.conf" 49L, 961C записано |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- merging incls/swits/igpats from old(ael-dundi-e164-canonical) to new(ael-dundi-e164-canonical) context, registrar = pbx_config -- Time to scan old dialplan and merge leftovers back into the new: 0.000487 sec -- Time to restore hints and swap in new dialplan: 0.000001 sec -- Time to delete the old dialplan: 0.000050 sec -- Total time merge_contexts_delete: 0.000538 sec [Jun 29 11:16:47] WARNING[9027]: pbx.c:9553 ast_context_verify_includes: Context 'grp3' tries to include nonexistent context 'local' [Jun 29 11:16:47] WARNING[9027]: pbx.c:9553 ast_context_verify_includes: Context 'grp2' tries to include nonexistent context 'local' [Jun 29 11:16:47] WARNING[9027]: pbx.c:9553 ast_context_verify_includes: Context 'grp1' tries to include nonexistent context 'local' linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- merging incls/swits/igpats from old(ael-dundi-e164-canonical) to new(ael-dundi-e164-canonical) context, registrar = pbx_config -- Time to scan old dialplan and merge leftovers back into the new: 0.000487 sec -- Time to restore hints and swap in new dialplan: 0.000001 sec -- Time to delete the old dialplan: 0.000050 sec -- Total time merge_contexts_delete: 0.000538 sec [Jun 29 11:16:47] WARNING[9027]: pbx.c:9553 ast_context_verify_includes: Context 'grp3' tries to include nonexistent context 'local' [Jun 29 11:16:47] WARNING[9027]: pbx.c:9553 ast_context_verify_includes: Context 'grp2' tries to include nonexistent context 'local' [Jun 29 11:16:47] WARNING[9027]: pbx.c:9553 ast_context_verify_includes: Context 'grp1' tries to include nonexistent context 'local' linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- merging incls/swits/igpats from old(ael-dundi-e164-canonical) to new(ael-dundi-e164-canonical) context, registrar = pbx_config -- Time to scan old dialplan and merge leftovers back into the new: 0.000487 sec -- Time to restore hints and swap in new dialplan: 0.000001 sec -- Time to delete the old dialplan: 0.000050 sec -- Total time merge_contexts_delete: 0.000538 sec [Jun 29 11:16:47] WARNING[9027]: pbx.c:9553 ast_context_verify_includes: Context 'grp3' tries to include nonexistent context 'local' [Jun 29 11:16:47] WARNING[9027]: pbx.c:9553 ast_context_verify_includes: Context 'grp2' tries to include nonexistent context 'local' [Jun 29 11:16:47] WARNING[9027]: pbx.c:9553 ast_context_verify_includes: Context 'grp1' tries to include nonexistent context 'local' linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- merging incls/swits/igpats from old(ael-dundi-e164-canonical) to new(ael-dundi-e164-canonical) context, registrar = pbx_config -- Time to scan old dialplan and merge leftovers back into the new: 0.000487 sec -- Time to restore hints and swap in new dialplan: 0.000001 sec -- Time to delete the old dialplan: 0.000050 sec -- Total time merge_contexts_delete: 0.000538 sec [Jun 29 11:16:47] WARNING[9027]: pbx.c:9553 ast_context_verify_includes: Context 'grp3' tries to include nonexistent context 'local' [Jun 29 11:16:47] WARNING[9027]: pbx.c:9553 ast_context_verify_includes: Context 'grp2' tries to include nonexistent context 'local' [Jun 29 11:16:47] WARNING[9027]: pbx.c:9553 ast_context_verify_includes: Context 'grp1' tries to include nonexistent context 'local' linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- merging incls/swits/igpats from old(ael-dundi-e164-canonical) to new(ael-dundi-e164-canonical) context, registrar = pbx_config -- Time to scan old dialplan and merge leftovers back into the new: 0.000487 sec -- Time to restore hints and swap in new dialplan: 0.000001 sec -- Time to delete the old dialplan: 0.000050 sec -- Total time merge_contexts_delete: 0.000538 sec [Jun 29 11:16:47] WARNING[9027]: pbx.c:9553 ast_context_verify_includes: Context 'grp3' tries to include nonexistent context 'local' [Jun 29 11:16:47] WARNING[9027]: pbx.c:9553 ast_context_verify_includes: Context 'grp2' tries to include nonexistent context 'local' [Jun 29 11:16:47] WARNING[9027]: pbx.c:9553 ast_context_verify_includes: Context 'grp1' tries to include nonexistent context 'local' linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- merging incls/swits/igpats from old(ael-dundi-e164-canonical) to new(ael-dundi-e164-canonical) context, registrar = pbx_config -- Time to scan old dialplan and merge leftovers back into the new: 0.000487 sec -- Time to restore hints and swap in new dialplan: 0.000001 sec -- Time to delete the old dialplan: 0.000050 sec -- Total time merge_contexts_delete: 0.000538 sec [Jun 29 11:16:47] WARNING[9027]: pbx.c:9553 ast_context_verify_includes: Context 'grp3' tries to include nonexistent context 'local' [Jun 29 11:16:47] WARNING[9027]: pbx.c:9553 ast_context_verify_includes: Context 'grp2' tries to include nonexistent context 'local' [Jun 29 11:16:47] WARNING[9027]: pbx.c:9553 ast_context_verify_includes: Context 'grp1' tries to include nonexistent context 'local' linux1*CLI> exit Executing last minute cleanups |
#vim extensions.conf
--- /tmp/l3-saved-6655.13510.24914 2011-06-29 11:18:13.000000000 +0400 +++ extensions.conf 2011-06-29 11:18:31.000000000 +0400 @@ -5,7 +5,8 @@ exten => _81XX,1,VoicemailMain(11${EXTEN:2}) -local] +[local] + exten => _11XX,1,Answer exten => _11XX,n,Dial(SIP/${EXTEN},15) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"])?voicemail) |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- Time to delete the old dialplan: 0.000050 sec -- Total time merge_contexts_delete: 0.000497 sec == Using SIP RTP CoS mark 5 -- Executing [8101@grp1:1] VoiceMailMain("SIP/1102-0000008d", "1101") in new stack -- <SIP/1102-0000008d> Playing 'vm-login.gsm' (language 'ru') -- <SIP/1102-0000008d> Playing 'vm-password.gsm' (language 'ru') -- Incorrect password '' for user '1102' (context = default) -- <SIP/1102-0000008d> Playing 'vm-incorrect-mailbox.gsm' (language 'ru') linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- Time to delete the old dialplan: 0.000050 sec -- Total time merge_contexts_delete: 0.000497 sec == Using SIP RTP CoS mark 5 -- Executing [8101@grp1:1] VoiceMailMain("SIP/1102-0000008d", "1101") in new stack -- <SIP/1102-0000008d> Playing 'vm-login.gsm' (language 'ru') -- <SIP/1102-0000008d> Playing 'vm-password.gsm' (language 'ru') -- Incorrect password '' for user '1102' (context = default) -- <SIP/1102-0000008d> Playing 'vm-incorrect-mailbox.gsm' (language 'ru') linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- Time to delete the old dialplan: 0.000050 sec -- Total time merge_contexts_delete: 0.000497 sec == Using SIP RTP CoS mark 5 -- Executing [8101@grp1:1] VoiceMailMain("SIP/1102-0000008d", "1101") in new stack -- <SIP/1102-0000008d> Playing 'vm-login.gsm' (language 'ru') -- <SIP/1102-0000008d> Playing 'vm-password.gsm' (language 'ru') -- Incorrect password '' for user '1102' (context = default) -- <SIP/1102-0000008d> Playing 'vm-incorrect-mailbox.gsm' (language 'ru') linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- Time to delete the old dialplan: 0.000050 sec -- Total time merge_contexts_delete: 0.000497 sec == Using SIP RTP CoS mark 5 -- Executing [8101@grp1:1] VoiceMailMain("SIP/1102-0000008d", "1101") in new stack -- <SIP/1102-0000008d> Playing 'vm-login.gsm' (language 'ru') -- <SIP/1102-0000008d> Playing 'vm-password.gsm' (language 'ru') -- Incorrect password '' for user '1102' (context = default) -- <SIP/1102-0000008d> Playing 'vm-incorrect-mailbox.gsm' (language 'ru') linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- Time to delete the old dialplan: 0.000050 sec -- Total time merge_contexts_delete: 0.000497 sec == Using SIP RTP CoS mark 5 -- Executing [8101@grp1:1] VoiceMailMain("SIP/1102-0000008d", "1101") in new stack -- <SIP/1102-0000008d> Playing 'vm-login.gsm' (language 'ru') -- <SIP/1102-0000008d> Playing 'vm-password.gsm' (language 'ru') -- Incorrect password '' for user '1102' (context = default) -- <SIP/1102-0000008d> Playing 'vm-incorrect-mailbox.gsm' (language 'ru') linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- Time to delete the old dialplan: 0.000050 sec -- Total time merge_contexts_delete: 0.000497 sec == Using SIP RTP CoS mark 5 -- Executing [8101@grp1:1] VoiceMailMain("SIP/1102-0000008d", "1101") in new stack -- <SIP/1102-0000008d> Playing 'vm-login.gsm' (language 'ru') -- <SIP/1102-0000008d> Playing 'vm-password.gsm' (language 'ru') -- Incorrect password '' for user '1102' (context = default) -- <SIP/1102-0000008d> Playing 'vm-incorrect-mailbox.gsm' (language 'ru') linux1*CLI> exit Executing last minute cleanups |
#[international]
; If you need to have an external program, i.e. /usr/bin/myapp ; called when a user changes her voicemail password, uncomment this: ;externpasscheck=/usr/bin/myapp ; Arguments for this script are: ; mailbox context oldpass newpass ; For the directory, you can override the intro file if you want ;directoryintro=dir-intro ; The character set for voicemail messages can be specified here ;charset=ISO-8859-1 ; The ADSI feature descriptor number to download to ... 301,1-8 80% ; ; Mailboxes may be organized into multiple contexts for ; voicemail virtualhosting ; [other] [other] ;directoryintro=dir-company2 1234 => 5678,Company2 User,root@localhost ;The intro can be customized on a per-context basis' IMp 352,0-1 95% |
#[international]
; Should the email contain the voicemail as an attachment attach=yes ; Maximum number of messages per folder. If not specified, a default value ; (100) is used. Maximum value for this option is 9999. ;maxmsg=100 ; Maximum length of a voicemail message in seconds ;maxsecs=180 ; Minimum length of a voicemail message in seconds for the message to be kept ; The default is no minimum. ;minsecs=3 ... ; ; Mailboxes may be organized into multiple contexts for ; voicemail virtualhosting ; [other] [other] eastern=America/New_York|'vm-received' Q 'digits/at' IMp 348,1 94% ;directoryintro=dir-company2 1234 => 5678,Company2 User,root@localhost 352,0-1 95% |
#[international]
;externnotify=/usr/bin/myapp ; If you would also like to enable SMDI notification then set smdienable to yes. ; You will also need to make sure smdiport is set to a valid port as specified in ; smdi.conf. ;smdienable=yes ;smdiport=/dev/ttyS0 ; If you need to have an external program, i.e. /usr/bin/myapp ; called when a voicemail password is changed, uncomment this: ; Note: If this is set, the password will NOT be changed in voicemail.conf ; If you would like to also change the password in voicemail.conf, use ... 285,1 75% ; ; Mailboxes may be organized into multiple contexts for ; voicemail virtualhosting ; [other] [other] ;directoryintro=dir-company2 1234 => 5678,Company2 User,root@localhost ;The intro can be customized on a per-context basis' IMp 352,0-1 95% |
#[international]
; not currently work with the "#include <file>" directive for Asterisk ; configuration files, nor when using realtime static configuration. ; Do not use them with this configuration file. ; [general] ; Formats for writing Voicemail. Note that when using IMAP storage for "voicemail.conf" 368L, 18008C ; voicemail, only the first format specified will be used. ;format=g723sf|wav49|wav format=wav49|gsm|wav ... 320,1 86% ; ; Mailboxes may be organized into multiple contexts for ; voicemail virtualhosting ; [other] [other] ;directoryintro=dir-company2 1234 => 5678,Company2 User,root@localhost ;The intro can be customized on a per-context basis 352,0-1 95% |
#[international]
; The default is no minimum. ;minsecs=3 ; Maximum length of greetings in seconds ;maxgreet=60 ; How many milliseconds to skip forward/back when rew/ff in message playback skipms=3000 ; How many seconds of silence before we end the recording maxsilence=10 ; Silence threshold (what we consider silence: the lower, the more sensitive) silencethreshold=128 ... 257,1-8 67% ; ; Mailboxes may be organized into multiple contexts for ; voicemail virtualhosting ; [other] [other] ;directoryintro=dir-company2 1234 => 5678,Company2 User,root@localhost ;The intro can be customized on a per-context basis' IMp 352,0-1 95% |
#[international]
; If you need to have an external program, i.e. /usr/bin/myapp ; called when a voicemail password is changed, uncomment this: ; Note: If this is set, the password will NOT be changed in voicemail.conf ; If you would like to also change the password in voicemail.conf, use ; the externpassnotify option below instead. ;externpass=/usr/bin/myapp ;externpassnotify=/usr/bin/myapp ; If you need to have an external program, i.e. /usr/bin/myapp ; called when a user changes her voicemail password, uncomment this: ;externpasscheck=/usr/bin/myapp ... 293,1-8 78% ; ; Mailboxes may be organized into multiple contexts for ; voicemail virtualhosting ; [other] [other] ;directoryintro=dir-company2 1234 => 5678,Company2 User,root@localhost ;The intro can be customized on a per-context basis' IMp 352,0-1 95% |
#vim extensions.conf
--- /tmp/l3-saved-6655.24574.13290 2011-06-29 11:50:43.000000000 +0400 +++ extensions.conf 2011-06-29 11:55:38.000000000 +0400 @@ -8,6 +8,9 @@ [local] exten => _11XX,1,Answer +exten => _11XX,n,Set(CALLFILENAME=${EXTEN}-${STRFTIME(${EPOCH},Europe/Moscow,"%Y-%m +-%d-%H:%M:%S")}) +exten => _11XX,n,Monitor(wav,${CALLFILENAME}) exten => _11XX,n,Dial(SIP/${EXTEN},15) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"])?voicemail) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"])?voicemail) |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- Executing [1102@grp2:3] Monitor("SIP/1103-00000093", "wav,1102-"2011-%") in new stack -- Executing [1102@grp2:4] Dial("SIP/1103-00000093", "SIP/1102,15") in new stack == Using SIP RTP CoS mark 5 -- Called 1102 [Jun 29 11:56:20] NOTICE[9236]: channel.c:3066 __ast_read: Dropping incompatible voice frame on SIP/1103-00000093 of format ulaw since our native format has changed to 0x8 (alaw) -- SIP/1102-00000094 is ringing -- SIP/1102-00000094 answered SIP/1103-00000093 == Spawn extension (grp2, 1102, 4) exited non-zero on 'SIP/1103-00000093' linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- Executing [1102@grp2:3] Monitor("SIP/1103-00000093", "wav,1102-"2011-%") in new stack -- Executing [1102@grp2:4] Dial("SIP/1103-00000093", "SIP/1102,15") in new stack == Using SIP RTP CoS mark 5 -- Called 1102 [Jun 29 11:56:20] NOTICE[9236]: channel.c:3066 __ast_read: Dropping incompatible voice frame on SIP/1103-00000093 of format ulaw since our native format has changed to 0x8 (alaw) -- SIP/1102-00000094 is ringing -- SIP/1102-00000094 answered SIP/1103-00000093 == Spawn extension (grp2, 1102, 4) exited non-zero on 'SIP/1103-00000093' linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- Executing [1102@grp2:3] Monitor("SIP/1103-00000093", "wav,1102-"2011-%") in new stack -- Executing [1102@grp2:4] Dial("SIP/1103-00000093", "SIP/1102,15") in new stack == Using SIP RTP CoS mark 5 -- Called 1102 [Jun 29 11:56:20] NOTICE[9236]: channel.c:3066 __ast_read: Dropping incompatible voice frame on SIP/1103-00000093 of format ulaw since our native format has changed to 0x8 (alaw) -- SIP/1102-00000094 is ringing -- SIP/1102-00000094 answered SIP/1103-00000093 == Spawn extension (grp2, 1102, 4) exited non-zero on 'SIP/1103-00000093' linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- Executing [1102@grp2:3] Monitor("SIP/1103-00000093", "wav,1102-"2011-%") in new stack -- Executing [1102@grp2:4] Dial("SIP/1103-00000093", "SIP/1102,15") in new stack == Using SIP RTP CoS mark 5 -- Called 1102 [Jun 29 11:56:20] NOTICE[9236]: channel.c:3066 __ast_read: Dropping incompatible voice frame on SIP/1103-00000093 of format ulaw since our native format has changed to 0x8 (alaw) -- SIP/1102-00000094 is ringing -- SIP/1102-00000094 answered SIP/1103-00000093 == Spawn extension (grp2, 1102, 4) exited non-zero on 'SIP/1103-00000093' linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- Executing [1102@grp2:3] Monitor("SIP/1103-00000093", "wav,1102-"2011-%") in new stack -- Executing [1102@grp2:4] Dial("SIP/1103-00000093", "SIP/1102,15") in new stack == Using SIP RTP CoS mark 5 -- Called 1102 [Jun 29 11:56:20] NOTICE[9236]: channel.c:3066 __ast_read: Dropping incompatible voice frame on SIP/1103-00000093 of format ulaw since our native format has changed to 0x8 (alaw) -- SIP/1102-00000094 is ringing -- SIP/1102-00000094 answered SIP/1103-00000093 == Spawn extension (grp2, 1102, 4) exited non-zero on 'SIP/1103-00000093' linux1*CLI> exit Executing last minute cleanups |
#asterisk -rvvvvv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) ... -- Executing [1102@grp2:3] Monitor("SIP/1103-00000093", "wav,1102-"2011-%") in new stack -- Executing [1102@grp2:4] Dial("SIP/1103-00000093", "SIP/1102,15") in new stack == Using SIP RTP CoS mark 5 -- Called 1102 [Jun 29 11:56:20] NOTICE[9236]: channel.c:3066 __ast_read: Dropping incompatible voice frame on SIP/1103-00000093 of format ulaw since our native format has changed to 0x8 (alaw) -- SIP/1102-00000094 is ringing -- SIP/1102-00000094 answered SIP/1103-00000093 == Spawn extension (grp2, 1102, 4) exited non-zero on 'SIP/1103-00000093' linux1*CLI> exit Executing last minute cleanups |
#vim extensions.conf
--- /tmp/l3-saved-6655.31903.17933 2011-06-29 11:57:23.000000000 +0400 +++ extensions.conf 2011-06-29 12:02:27.000000000 +0400 @@ -8,8 +8,7 @@ [local] exten => _11XX,1,Answer -exten => _11XX,n,Set(CALLFILENAME=${EXTEN}-${STRFTIME(${EPOCH},Europe/Moscow,"%Y-%m --%d-%H:%M:%S")}) +exten => _11XX,n,Set(CALLFILENAME=${EXTEN}-${STRFTIME(${EPOCH},Europe/Moscow,"%Y-%m-%d-%H:%M:%S")}) exten => _11XX,n,Monitor(wav,${CALLFILENAME}) exten => _11XX,n,Dial(SIP/${EXTEN},15) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"])?voicemail) |
#asterisk -rv
Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux1 (pid = 1629) Verbosity is at least 5 linux1*CLI> ... [Jun 29 12:00:38] NOTICE[9452]: channel.c:3066 __ast_read: Dropping incompatible voice frame on SIP/1103-00000095 of format ulaw since our native format has changed to 0x8 (alaw) -- SIP/1102-00000096 is ringing -- Nobody picked up in 15000 ms -- Executing [1102@grp2:5] GotoIf("SIP/1103-00000095", "0)?voicemail") in new stack -- Executing [1102@grp2:6] GotoIf("SIP/1103-00000095", "1)?voicemail") in new stack -- Goto (grp2,1102,9) -- Executing [1102@grp2:9] VoiceMail("SIP/1103-00000095", "1102@default") in new stack -- <SIP/1103-00000095> Playing 'vm-intro.gsm' (language 'ru') linux1*CLI> quit Executing last minute cleanups |
#less /etc/asterisk/extensions.conf
|
#{EXTEN})
adsi.conf cdr_manager.conf dnsmgr.conf followme.conf manager.d queuerules.conf sip_notify.conf adtranvofr.conf cdr_odbc.conf dsp.conf func_odbc.conf meetme.conf queues.conf skinny.conf agents.conf cdr_pgsql.conf dundi.conf gtalk.conf mgcp.conf res_config_sqlite.conf sla.conf ais.conf cdr_sqlite3_custom.conf enum.conf h323.conf minivm.conf res_ldap.conf smdi.conf alarmreceiver.conf cdr_tds.conf extconfig.conf http.conf misdn.conf res_odbc.conf telcordia-1.adsi alsa.conf chan_dahdi.conf extensions.ael iax.conf modules.conf res_pgsql.conf udptl.conf amd.conf cli_aliases.conf extensions.conf iaxprov.conf musiconhold.conf res_snmp.conf unistim.conf asterisk.adsi cli.conf extensions.conf.SAVE indications.conf muted.conf rpt.conf usbradio.conf asterisk.conf cli_permissions.conf extensions.lua jabber.conf osp.conf rtp.conf users.conf cdr_adaptive_odbc.conf codecs.conf extensions_minivm.conf jingle.conf oss.conf say.conf voicemail.conf cdr.conf console.conf features.conf logger.conf phone.conf sip.conf vpb.conf cdr_custom.conf dbsep.conf festival.conf manager.conf phoneprov.conf sip.conf.SAVE |
#{EXTEN})
adsi.conf cdr_manager.conf dnsmgr.conf followme.conf manager.d queuerules.conf sip_notify.conf adtranvofr.conf cdr_odbc.conf dsp.conf func_odbc.conf meetme.conf queues.conf skinny.conf agents.conf cdr_pgsql.conf dundi.conf gtalk.conf mgcp.conf res_config_sqlite.conf sla.conf ais.conf cdr_sqlite3_custom.conf enum.conf h323.conf minivm.conf res_ldap.conf smdi.conf alarmreceiver.conf cdr_tds.conf extconfig.conf http.conf misdn.conf res_odbc.conf telcordia-1.adsi alsa.conf chan_dahdi.conf extensions.ael iax.conf modules.conf res_pgsql.conf udptl.conf amd.conf cli_aliases.conf extensions.conf iaxprov.conf musiconhold.conf res_snmp.conf unistim.conf asterisk.adsi cli.conf extensions.conf.SAVE indications.conf muted.conf rpt.conf usbradio.conf asterisk.conf cli_permissions.conf extensions.lua jabber.conf osp.conf rtp.conf users.conf cdr_adaptive_odbc.conf codecs.conf extensions_minivm.conf jingle.conf oss.conf say.conf voicemail.conf cdr.conf console.conf features.conf logger.conf phone.conf sip.conf vpb.conf cdr_custom.conf dbsep.conf festival.conf manager.conf phoneprov.conf sip.conf.SAVE |
#{EXTEN})
adsi.conf cdr_manager.conf dnsmgr.conf followme.conf manager.d queuerules.conf sip_notify.conf adtranvofr.conf cdr_odbc.conf dsp.conf func_odbc.conf meetme.conf queues.conf skinny.conf agents.conf cdr_pgsql.conf dundi.conf gtalk.conf mgcp.conf res_config_sqlite.conf sla.conf ais.conf cdr_sqlite3_custom.conf enum.conf h323.conf minivm.conf res_ldap.conf smdi.conf alarmreceiver.conf cdr_tds.conf extconfig.conf http.conf misdn.conf res_odbc.conf telcordia-1.adsi alsa.conf chan_dahdi.conf extensions.ael iax.conf modules.conf res_pgsql.conf udptl.conf amd.conf cli_aliases.conf extensions.conf iaxprov.conf musiconhold.conf res_snmp.conf unistim.conf asterisk.adsi cli.conf extensions.conf.SAVE indications.conf muted.conf rpt.conf usbradio.conf asterisk.conf cli_permissions.conf extensions.lua jabber.conf osp.conf rtp.conf users.conf cdr_adaptive_odbc.conf codecs.conf extensions_minivm.conf jingle.conf oss.conf say.conf voicemail.conf cdr.conf console.conf features.conf logger.conf phone.conf sip.conf vpb.conf cdr_custom.conf dbsep.conf festival.conf manager.conf phoneprov.conf sip.conf.SAVE |
#{EXTEN})
adsi.conf cdr_manager.conf dnsmgr.conf followme.conf manager.d queuerules.conf sip_notify.conf adtranvofr.conf cdr_odbc.conf dsp.conf func_odbc.conf meetme.conf queues.conf skinny.conf agents.conf cdr_pgsql.conf dundi.conf gtalk.conf mgcp.conf res_config_sqlite.conf sla.conf ais.conf cdr_sqlite3_custom.conf enum.conf h323.conf minivm.conf res_ldap.conf smdi.conf alarmreceiver.conf cdr_tds.conf extconfig.conf http.conf misdn.conf res_odbc.conf telcordia-1.adsi alsa.conf chan_dahdi.conf extensions.ael iax.conf modules.conf res_pgsql.conf udptl.conf amd.conf cli_aliases.conf extensions.conf iaxprov.conf musiconhold.conf res_snmp.conf unistim.conf asterisk.adsi cli.conf extensions.conf.SAVE indications.conf muted.conf rpt.conf usbradio.conf asterisk.conf cli_permissions.conf extensions.lua jabber.conf osp.conf rtp.conf users.conf cdr_adaptive_odbc.conf codecs.conf extensions_minivm.conf jingle.conf oss.conf say.conf voicemail.conf cdr.conf console.conf features.conf logger.conf phone.conf sip.conf vpb.conf cdr_custom.conf dbsep.conf festival.conf manager.conf phoneprov.conf sip.conf.SAVE |
#{EXTEN})
adsi.conf cdr_manager.conf dnsmgr.conf followme.conf manager.d queuerules.conf sip_notify.conf adtranvofr.conf cdr_odbc.conf dsp.conf func_odbc.conf meetme.conf queues.conf skinny.conf agents.conf cdr_pgsql.conf dundi.conf gtalk.conf mgcp.conf res_config_sqlite.conf sla.conf ais.conf cdr_sqlite3_custom.conf enum.conf h323.conf minivm.conf res_ldap.conf smdi.conf alarmreceiver.conf cdr_tds.conf extconfig.conf http.conf misdn.conf res_odbc.conf telcordia-1.adsi alsa.conf chan_dahdi.conf extensions.ael iax.conf modules.conf res_pgsql.conf udptl.conf amd.conf cli_aliases.conf extensions.conf iaxprov.conf musiconhold.conf res_snmp.conf unistim.conf asterisk.adsi cli.conf extensions.conf.SAVE indications.conf muted.conf rpt.conf usbradio.conf asterisk.conf cli_permissions.conf extensions.lua jabber.conf osp.conf rtp.conf users.conf cdr_adaptive_odbc.conf codecs.conf extensions_minivm.conf jingle.conf oss.conf say.conf voicemail.conf cdr.conf console.conf features.conf logger.conf phone.conf sip.conf vpb.conf cdr_custom.conf dbsep.conf festival.conf manager.conf phoneprov.conf sip.conf.SAVE |
#{EXTEN})
adsi.conf cdr_manager.conf dnsmgr.conf followme.conf manager.d queuerules.conf sip_notify.conf adtranvofr.conf cdr_odbc.conf dsp.conf func_odbc.conf meetme.conf queues.conf skinny.conf agents.conf cdr_pgsql.conf dundi.conf gtalk.conf mgcp.conf res_config_sqlite.conf sla.conf ais.conf cdr_sqlite3_custom.conf enum.conf h323.conf minivm.conf res_ldap.conf smdi.conf alarmreceiver.conf cdr_tds.conf extconfig.conf http.conf misdn.conf res_odbc.conf telcordia-1.adsi alsa.conf chan_dahdi.conf extensions.ael iax.conf modules.conf res_pgsql.conf udptl.conf amd.conf cli_aliases.conf extensions.conf iaxprov.conf musiconhold.conf res_snmp.conf unistim.conf asterisk.adsi cli.conf extensions.conf.SAVE indications.conf muted.conf rpt.conf usbradio.conf asterisk.conf cli_permissions.conf extensions.lua jabber.conf osp.conf rtp.conf users.conf cdr_adaptive_odbc.conf codecs.conf extensions_minivm.conf jingle.conf oss.conf say.conf voicemail.conf cdr.conf console.conf features.conf logger.conf phone.conf sip.conf vpb.conf cdr_custom.conf dbsep.conf festival.conf manager.conf phoneprov.conf sip.conf.SAVE |
#vim extensions.conf
--- /tmp/l3-saved-6655.5372.18900 2011-06-29 12:05:04.000000000 +0400 +++ extensions.conf 2011-06-29 12:05:48.000000000 +0400 @@ -9,7 +9,7 @@ exten => _11XX,1,Answer exten => _11XX,n,Set(CALLFILENAME=${EXTEN}-${STRFTIME(${EPOCH},Europe/Moscow,"%Y-%m-%d-%H:%M:%S")}) -exten => _11XX,n,Monitor(wav,${CALLFILENAME}) +exten => _11XX,n,Monitor(wav,${CALLFILENAME}, m) exten => _11XX,n,Dial(SIP/${EXTEN},15) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"])?voicemail) exten => _11XX,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"])?voicemail) |
Время первой команды журнала | 09:00:03 2011- 6-29 | |||||||||||||||||||||
Время последней команды журнала | 10:03:43 2011- 6-29 | |||||||||||||||||||||
Количество командных строк в журнале | 101 | |||||||||||||||||||||
Процент команд с ненулевым кодом завершения, % | 5.94 | |||||||||||||||||||||
Процент синтаксически неверно набранных команд, % | 0.00 | |||||||||||||||||||||
Суммарное время работы с терминалом *, час | 1.06 | |||||||||||||||||||||
Количество командных строк в единицу времени, команда/мин | 1.59 | |||||||||||||||||||||
Частота использования команд |
|
В журнал автоматически попадают все команды, данные в любом терминале системы.
Для того чтобы убедиться, что журнал на текущем терминале ведётся, и команды записываются, дайте команду w. В поле WHAT, соответствующем текущему терминалу, должна быть указана программа script.
Команды, при наборе которых были допущены синтаксические ошибки, выводятся перечёркнутым текстом:
$ l s-l bash: l: command not found |
Если код завершения команды равен нулю, команда была выполнена без ошибок. Команды, код завершения которых отличен от нуля, выделяются цветом.
$ test 5 -lt 4 |
Команды, ход выполнения которых был прерван пользователем, выделяются цветом.
$ find / -name abc find: /home/devi-orig/.gnome2: Keine Berechtigung find: /home/devi-orig/.gnome2_private: Keine Berechtigung find: /home/devi-orig/.nautilus/metafiles: Keine Berechtigung find: /home/devi-orig/.metacity: Keine Berechtigung find: /home/devi-orig/.inkscape: Keine Berechtigung ^C |
Команды, выполненные с привилегиями суперпользователя, выделяются слева красной чертой.
# id uid=0(root) gid=0(root) Gruppen=0(root) |
Изменения, внесённые в текстовый файл с помощью редактора, запоминаются и показываются в журнале в формате ed. Строки, начинающиеся символом "<", удалены, а строки, начинающиеся символом ">" -- добавлены.
$ vi ~/.bashrc
|
Для того чтобы изменить файл в соответствии с показанными в диффшоте изменениями, можно воспользоваться командой patch. Нужно скопировать изменения, запустить программу patch, указав в качестве её аргумента файл, к которому применяются изменения, и всавить скопированный текст:
$ patch ~/.bashrc |
Для того чтобы получить краткую справочную информацию о команде, нужно подвести к ней мышь. Во всплывающей подсказке появится краткое описание команды.
Если справочная информация о команде есть, команда выделяется голубым фоном, например: vi. Если справочная информация отсутствует, команда выделяется розовым фоном, например: notepad.exe. Справочная информация может отсутствовать в том случае, если (1) команда введена неверно; (2) если распознавание команды LiLaLo выполнено неверно; (3) если информация о команде неизвестна LiLaLo. Последнее возможно для редких команд.
Большие, в особенности многострочные, всплывающие подсказки лучше всего показываются браузерами KDE Konqueror, Apple Safari и Microsoft Internet Explorer. В браузерах Mozilla и Firefox они отображаются не полностью, а вместо перевода строки выводится специальный символ.
Время ввода команды, показанное в журнале, соответствует времени начала ввода командной строки, которое равно тому моменту, когда на терминале появилось приглашение интерпретатора
Имя терминала, на котором была введена команда, показано в специальном блоке. Этот блок показывается только в том случае, если терминал текущей команды отличается от терминала предыдущей.
Вывод не интересующих вас в настоящий момент элементов журнала, таких как время, имя терминала и других, можно отключить. Для этого нужно воспользоваться формой управления журналом вверху страницы.
Небольшие комментарии к командам можно вставлять прямо из командной строки. Комментарий вводится прямо в командную строку, после символов #^ или #v. Символы ^ и v показывают направление выбора команды, к которой относится комментарий: ^ - к предыдущей, v - к следующей. Например, если в командной строке было введено:
$ whoami
user
$ #^ Интересно, кто я?в журнале это будет выглядеть так:
$ whoami
user
Интересно, кто я? |
Если комментарий содержит несколько строк, его можно вставить в журнал следующим образом:
$ whoami
user
$ cat > /dev/null #^ Интересно, кто я?
Программа whoami выводит имя пользователя, под которым мы зарегистрировались в системе. - Она не может ответить на вопрос о нашем назначении в этом мире.В журнале это будет выглядеть так:
$ whoami user
|
Комментарии, не относящиеся непосредственно ни к какой из команд, добавляются точно таким же способом, только вместо симолов #^ или #v нужно использовать символы #=
1 2 3 4Группы команд, выполненных на разных терминалах, разделяются специальной линией. Под этой линией в правом углу показано имя терминала, на котором выполнялись команды. Для того чтобы посмотреть команды только одного сенса, нужно щёкнуть по этому названию.
LiLaLo (L3) расшифровывается как Live Lab Log.
Программа разработана для повышения эффективности обучения Unix/Linux-системам.
(c) Игорь Чубин, 2004-2008