/l3/users/sbond/nt-voip-2011-06/linux8.unix.nt/root :1 :2 :3 :4 :5 :6 :7 :8 :9 :10 :11 :12 :13 :14 :15 :16 :17 :18 :19 :20 :21 :22 :23 :24 :25 :26 :27 :28 :29 :30 :31 :32 :33 :34 :35 :36 :37 :38 :39 |
|
#!v
[general] context=default allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no tcpbindaddr=0.0.0.0 srvlookup=yes register => lvov:password@192.168.107.1/kiev register => lvov:password@192.168.103.1/paris [2401] ... language=de callgroup=100 pickugroup=100 [2402] username=lvov canreinvite=no context=gr4 callgroup=100 pickugroup=100 type=friend 24,0-1 Top |
#!v
![]() register => lvov:password@192.168.107.1/kiev register => lvov:password@192.168.103.1/paris [2401] type=friend secret=password host=dynamic user=2401 callerid="VoIP Phone <2401>" context=gr3isk/sip.conf" 58L, 866C mailbox=2401@default ... secret=password [paris]namic [ ] username=lvov canreinvite=no context=gr4 callgroup=100 pickugrou=30 =30 "sip.conf" 58L, 858C written |
#!v
![]() [general] context=default allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no tcpbindaddr=0.0.0.0 srvlookup=yes register => lvov:password@192.168.107.1/kiev register => lvov:password@192.168.103.1/paris [2401] ... [paris]namic [ ] username=lvov canreinvite=no context=gr4 callgroup=100 pickugrou=30 =30 "sip.conf" 58L, 858C written callgroup=100 |
#!v
[general] context=default allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no tcpbindaddr=0.0.0.0 srvlookup=yes "/etc/asterisk/sip.conf" 58L, 866C canreinvite=no context=gr4 ... =30 "sip.conf" 58L, 858C written secret=password host=dynamic user=2402 callerid="Softphone <2402>" context=gr3 mailbox=2402@default language=de callgroup=100 |
#vim /etc/asterisk/extensions.conf
![]() |
#vim /etc/asterisk/extensions.conf
![]() --- /tmp/l3-saved-22920.8386.18634 2011-06-24 10:35:37.000000000 +0300 +++ /etc/asterisk/extensions.conf 2011-06-24 10:36:11.000000000 +0300 @@ -152,7 +152,7 @@ exten => _24XX,n,Monitor(wav,${MON_FILENAME},m) ;exten => _24XX,n,System(/usr/local/bin/asterisk-jabber-logger Vam zvonyat s ${CALLERID(num)}) ;exten => _24XX,n,SetMusicOnHold(${DB(moh/${EXTEN})}) -exten => _24XX,n,Dial(SIP/${EXTEN},10) +exten => _24XX,n,Dial(SIP/${EXTEN},20) exten => _24XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?missed_call) exten => _24XX,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?missed_call) exten => _24XX,n,Hangup() |
#!v
![]() [ ] [2402] type=friend secret=pa=20 =20 host=dynamic [2402]402 [ ] callerid="Softphone <2402>" context=gr3 mailbox=2402@default language=de pickugroup=30 =20 =20 [kiev] [kiev] type=friend "sip.conf" 54L, 804C written |
#!v
![]() [ ] [2402] type=friend secret=pa=20 =20 host=dynamic [2402]402 [ ] callerid="Softphone <2402>" context=gr3 mailbox=2402@default language=de pickugroup=30 =20 =20 [kiev] [kiev] type=friend "sip.conf" 54L, 804C written |
#!v
![]() [ ] [2402] type=friend secret=pa=20 =20 host=dynamic [2402]402 [ ] callerid="Softphone <2402>" context=gr3 mailbox=2402@default language=de pickugroup=30 =20 =20 [kiev] [kiev] type=friend "sip.conf" 54L, 804C written |
#!v
[ ] [2402] type=friend secret=pa=20 =20 host=dynamic [2402]402 [ ] callerid="Softphone <2402>" context=gr3 mailbox=2402@default language=de pickugroup=30 =20 =20 [kiev] [kiev] type=friend "sip.conf" 54L, 804C written |
#!v
![]() upgroup=20 [kiev] [2402] [ ] secret=password host=dynamic username=lvov canreinvite=no upgroup=20 context=gr4 [kiev] [paris] [paris] secret=password [paris] host=dynamic username=lvov canreinvite=no "sip.conf" 54L, 806C written |
#!v
![]() upgroup=20 [kiev] [2402] [ ] secret=password host=dynamic username=lvov canreinvite=no upgroup=20 context=gr4 [kiev] [paris] [paris] secret=password [paris] host=dynamic username=lvov canreinvite=no "sip.conf" 54L, 806C written |
#!v
![]() upgroup=20 [kiev] [2402] [ ] secret=password host=dynamic username=lvov canreinvite=no upgroup=20 context=gr4 [kiev] [paris] [paris] secret=password [paris] host=dynamic username=lvov canreinvite=no "sip.conf" 54L, 806C written |
#!v
upgroup=20 [kiev] [2402] [ ] secret=password host=dynamic username=lvov canreinvite=no upgroup=20 context=gr4 [kiev] [paris] [paris] secret=password [paris] host=dynamic username=lvov canreinvite=no "sip.conf" 54L, 806C written |
#!v
![]() context=gr3isk/sip.conf" 54L, 806C mailbox=2401@default language=de callgroup=20 pickupgroup=20 [2401] [ ] [2402] type=friend secret=password ;perehvat ;perehvat host=dynamic user=2402 callerid="Softphone <2402>" context=gr3 mailbox=2402@default language=de "sip.conf" 54L, 846C written |
#!v
![]() context=gr3isk/sip.conf" 54L, 806C mailbox=2401@default language=de callgroup=20 pickupgroup=20 [2401] [ ] [2402] type=friend secret=password ;perehvat ;perehvat host=dynamic user=2402 callerid="Softphone <2402>" context=gr3 mailbox=2402@default language=de "sip.conf" 54L, 846C written |
#!v
![]() [general] context=default allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no tcpbindaddr=0.0.0.0 srvlookup=yes register => lvov:password@192.168.107.1/kiev register => lvov:password@192.168.103.1/paris [2401] ... type=friend secret=password ;perehvat ;perehvat host=dynamic user=2402 callerid="Softphone <2402>" context=gr3 mailbox=2402@default language=de "sip.conf" 54L, 846C written |
#!v
[general] context=default allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no tcpbindaddr=0.0.0.0 srvlookup=yes register => lvov:password@192.168.107.1/kiev register => lvov:password@192.168.103.1/paris [2401] ... secret=password host=dynamic ;perehvat ;perehvat user=2402 callerid="Softphone <2402>" context=gr3 mailbox=2402@default language=de callgroup=20 "sip.conf" 54L, 846C written |
#vim /etc/asterisk/queues.conf
![]() --- /tmp/l3-saved-22920.11980.28882 2011-06-24 11:20:04.000000000 +0300 +++ /etc/asterisk/queues.conf 2011-06-24 11:20:54.000000000 +0300 @@ -498,3 +498,6 @@ ;member => Agent/:1,1 ; Any agent in group 1, wait for first ; available, but consider with penalty +[queue1] +member => SIP/2401 +member => SIP/2402 |
#vim /etc/asterisk/extensions.conf
--- /tmp/l3-saved-27557.21090.13847 2011-06-24 11:21:13.000000000 +0300 +++ /etc/asterisk/extensions.conf 2011-06-24 11:25:44.000000000 +0300 @@ -145,6 +145,12 @@ [local] + +exten => 2405,1,Answer +exten => 2405,n,SetMusicOnHold(default) +exten => 2405,n,Queue(queue1) + + ;-----------------------------------LOCAL DIALPLAN exten => _24XX,1,Answer ;exten => _24XX,n,Set(MON_FILENAME=record-${STRFTIME(${EPOCH},,%Y-%m-%d-%H-%M-%S)}-${EXTEN}-${CALLER ID(num)}) |
#!v
![]() [lang-menu] [ ] exten => s,1,Answer exten => s,n(startmenu),Background(/var/tmp/asterisk/record10) ;hi exten => s,n,WaitExten(1) exten => 7,1,Set(MENU_LANG=ru) exten => 7,n,Goto(menu,s,1) exten => 8,1,Set(MENU_LANG=uk) exten => 8,n,Goto(menu,s,1) exten => i,1,Background(/var/tmp/asterisk/record05) ;error (i-znachit ne izvestnii) ... ;-----------------------------------LOCAL DIALPLAN exten => _24XX,1,Answer ;exten => _24XX,n,Set(MON_FILENAME=record-${STRFTIME(${EPOCH},,%Y-%m-%d-%H-%M-%S)}-${EXTEN}-${CALLER ID(num)}) exten => _24XX,n,Set(MON_FILENAME=record-${EXTEN}-${CALLERID(num)}-${STRFTIME(${EPOCH},,%Y-%m-%d-%H-%M-%S)}) exten => _24XX,n,Monitor(wav,${MON_FILENAME},m) ;exten => _24XX,n,System(/usr/local/bin/asterisk-jabber-logger Vam zvonyat s ${CALLERID(num)}) ;exten => _24XX,n,SetMusicOnHold(${DB(moh/${EXTEN})}) exten => _24XX,n,Dial(SIP/${EXTEN},20) exten => _24XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?missed_call) "/etc/asterisk/extensions.conf" 217L, 7147C written |
#!v
exten => s,n,WaitExten(3) exten => s,n,Background(/var/tmp/asterisk/record09) ; vy slihkom dolgo zhdete exten => i,1,Background(/var/tmp/asterisk/record05) ; oshibka exten => i,n,WaitExten(1) exten => i,n,Goto(menu,s,start-menu) ;exten => 1,1,AGI(/usr/local/bin/festival-script.pl, /var/tmp/ru-kiev.txt) exten => 1,1,Goto(kievmenu,s,1) exten => 2,1,Goto(submenu,s,1) exten => 3,1,AGI(/usr/local/bin/festival-script.pl, /var/tmp/ru-asterisk.txt) exten => 4,1,Playback(/var/tmp/asterisk/record06) ;svyaz s operatorom ... ;-----------------------------------LOCAL DIALPLAN exten => _24XX,1,Answer ;exten => _24XX,n,Set(MON_FILENAME=record-${STRFTIME(${EPOCH},,%Y-%m-%d-%H-%M-%S)}-${EXTEN}-${CALLER ID(num)}) exten => _24XX,n,Set(MON_FILENAME=record-${EXTEN}-${CALLERID(num)}-${STRFTIME(${EPOCH},,%Y-%m-%d-%H-%M-%S)}) exten => _24XX,n,Monitor(wav,${MON_FILENAME},m) ;exten => _24XX,n,System(/usr/local/bin/asterisk-jabber-logger Vam zvonyat s ${CALLERID(num)}) ;exten => _24XX,n,SetMusicOnHold(${DB(moh/${EXTEN})}) exten => _24XX,n,Dial(SIP/${EXTEN},20) exten => _24XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?missed_call) "/etc/asterisk/extensions.conf" 217L, 7147C written |
#member => SIP/1102
![]() 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 linux8 (pid = 19127) ... == Using SIP RTP CoS mark 5 [Jun 24 11:28:38] WARNING[23269]: chan_sip.c:5467 create_addr: No such host: 2405 [Jun 24 11:28:38] WARNING[23269]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [2405@gr4:5] GotoIf("SIP/kiev-00000091", "0?missed_call") in new stack -- Executing [2405@gr4:6] GotoIf("SIP/kiev-00000091", "0?missed_call") in new stack -- Executing [2405@gr4:7] Hangup("SIP/kiev-00000091", "") in new stack == Spawn extension (gr4, 2405, 7) exited non-zero on 'SIP/kiev-00000091' linux8*CLI> exit Executing last minute cleanups |
#member => SIP/1102
![]() 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 linux8 (pid = 19127) ... == Using SIP RTP CoS mark 5 [Jun 24 11:28:38] WARNING[23269]: chan_sip.c:5467 create_addr: No such host: 2405 [Jun 24 11:28:38] WARNING[23269]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [2405@gr4:5] GotoIf("SIP/kiev-00000091", "0?missed_call") in new stack -- Executing [2405@gr4:6] GotoIf("SIP/kiev-00000091", "0?missed_call") in new stack -- Executing [2405@gr4:7] Hangup("SIP/kiev-00000091", "") in new stack == Spawn extension (gr4, 2405, 7) exited non-zero on 'SIP/kiev-00000091' linux8*CLI> exit Executing last minute cleanups |
#member => SIP/1102
![]() 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 linux8 (pid = 19127) ... == Using SIP RTP CoS mark 5 [Jun 24 11:28:38] WARNING[23269]: chan_sip.c:5467 create_addr: No such host: 2405 [Jun 24 11:28:38] WARNING[23269]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [2405@gr4:5] GotoIf("SIP/kiev-00000091", "0?missed_call") in new stack -- Executing [2405@gr4:6] GotoIf("SIP/kiev-00000091", "0?missed_call") in new stack -- Executing [2405@gr4:7] Hangup("SIP/kiev-00000091", "") in new stack == Spawn extension (gr4, 2405, 7) exited non-zero on 'SIP/kiev-00000091' linux8*CLI> exit Executing last minute cleanups |
#member => SIP/1102
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 linux8 (pid = 19127) ... == Using SIP RTP CoS mark 5 [Jun 24 11:28:38] WARNING[23269]: chan_sip.c:5467 create_addr: No such host: 2405 [Jun 24 11:28:38] WARNING[23269]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [2405@gr4:5] GotoIf("SIP/kiev-00000091", "0?missed_call") in new stack -- Executing [2405@gr4:6] GotoIf("SIP/kiev-00000091", "0?missed_call") in new stack -- Executing [2405@gr4:7] Hangup("SIP/kiev-00000091", "") in new stack == Spawn extension (gr4, 2405, 7) exited non-zero on 'SIP/kiev-00000091' linux8*CLI> exit Executing last minute cleanups |
#/etc/init.d/asterisk restart
![]() Stopping Asterisk PBX: asterisk. Starting Asterisk PBX: asterisk. |
#/etc/init.d/asterisk restart
![]() Stopping Asterisk PBX: asterisk. Starting Asterisk PBX: asterisk. |
#/etc/init.d/asterisk restart
![]() Stopping Asterisk PBX: asterisk. Starting Asterisk PBX: asterisk. |
#/etc/init.d/asterisk restart
Stopping Asterisk PBX: asterisk. Starting Asterisk PBX: asterisk. |
#vim /etc/asterisk/agents.conf
--- /tmp/l3-saved-27557.25322.32739 2011-06-24 11:40:12.000000000 +0300 +++ /etc/asterisk/agents.conf 2011-06-24 11:40:22.000000000 +0300 @@ -109,5 +109,5 @@ ; ; agent => agentid,agentpassword,name ; -;agent => 1001,4321,Mark Spencer -;agent => 1002,4321,Will Meadows +agent => 1001,4321,Mark Spencer +agent => 1002,4321,Will Meadows |
#vim /etc/asterisk/extensions.conf
![]() --- /tmp/l3-saved-27557.19183.16089 2011-06-24 11:41:23.000000000 +0300 +++ /etc/asterisk/extensions.conf 2011-06-24 11:42:18.000000000 +0300 @@ -150,6 +150,8 @@ exten => 2405,n,SetMusicOnHold(default) exten => 2405,n,Queue(queue1) +exten => 2406,1,AgentLogin(1001) +exten => 2406,1,AgentLogin(1002) ;-----------------------------------LOCAL DIALPLAN exten => _24XX,1,Answer |
#vim /etc/asterisk/queues.conf
![]() --- /tmp/l3-saved-27557.21338.30766 2011-06-24 11:42:29.000000000 +0300 +++ /etc/asterisk/queues.conf 2011-06-24 11:42:58.000000000 +0300 @@ -499,5 +499,6 @@ ; available, but consider with penalty [queue1] -member => SIP/2401 -member => SIP/2402 +member => Agent/1101 +;member => SIP/2401 +;member => SIP/2402 |
#!v
![]() [general] ; ; Global settings for call queues ; "/etc/asterisk/queues.conf" 504L, 22521C ; Persistent Members ; Store each dynamic member in each queue in the astdb so that ; when asterisk is restarted, each member will be automatically ; read into their recorded queues. Default is 'yes'. ; ... ; probably more along the lines of how a queue should work and ; in most cases, you will want to enable this behavior. If you ; do not specify or comment out this option, it will default to no ; to keep backward compatibility with the old behavior. ; autofill = yes ; ; Monitor Type ; By setting monitor-type = MixMonitor, when specifying monitor-format ; to enable recording of queue member conversations, app_queue will |
#!v
; were trying to get to. The next waiting caller in line then ; becomes the head caller, and they are then connected with the ; next available member and all available members and waiting callers ; waits while this happens. The new behavior, enabled by setting ; autofill=yes makes sure that when the waiting callers are connecting ; with available members in a parallel fashion until there are ; no more available members or no more waiting callers. This is ; probably more along the lines of how a queue should work and ; in most cases, you will want to enable this behavior. If you ; do not specify or comment out this option, it will default to no ; to keep backward compatibility with the old behavior. ; autofill = yes ; ; Monitor Type ; By setting monitor-type = MixMonitor, when specifying monitor-format ; to enable recording of queue member conversations, app_queue will |
#;member => SIP/2402
![]() exten => 2405,1,Answer exten => 2405,n,SetMusicOnHold(default)/record12) ; incorrect password exten => 2405,n,Queue(queue1) exten => 2406,1,AgentLogin(1001) exten => 2406,1,AgentLogin(1002) ;-----------------------------------LOCAL DIALPLAN exten => _24XX,1,Answer ;exten => _24XX,n,Set(MON_FILENAME=record-${STRFTIME(${EPOCH},,%Y-%m-%d-%H-%M-%S)}-${EXTEN}-${CALLER ID(num)}) exten => _24XX,n,Set(MON_FILENAME=record-${EXTEN}-${CALLERID(num)}-${STRFTIME(${EPOCH},,%Y-%m-%d-%H-%M-%S)}) exten => _24XX,n,Monitor(wav,${MON_FILENAME},m) ;exten => _24XX,n,System(/usr/local/bin/asterisk-jabber-logger Vam zvonyat s ${CALLERID(num)}) ;exten => _24XX,n,SetMusicOnHold(${DB(moh/${EXTEN})}) exten => _24XX,n,Dial(SIP/${EXTEN},20) exten => _24XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?missed_call) exten => _24XX,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?missed_call) |
#;member => SIP/2402
exten => 2405,1,Answer exten => 2405,n,SetMusicOnHold(default)/record12) ; incorrect password exten => 2405,n,Queue(queue1) exten => 2406,1,AgentLogin(1001) exten => 2406,1,AgentLogin(1002) ;-----------------------------------LOCAL DIALPLAN exten => _24XX,1,Answer ;exten => _24XX,n,Set(MON_FILENAME=record-${STRFTIME(${EPOCH},,%Y-%m-%d-%H-%M-%S)}-${EXTEN}-${CALLER ID(num)}) exten => _24XX,n,Set(MON_FILENAME=record-${EXTEN}-${CALLERID(num)}-${STRFTIME(${EPOCH},,%Y-%m-%d-%H-%M-%S)}) exten => _24XX,n,Monitor(wav,${MON_FILENAME},m) ;exten => _24XX,n,System(/usr/local/bin/asterisk-jabber-logger Vam zvonyat s ${CALLERID(num)}) ;exten => _24XX,n,SetMusicOnHold(${DB(moh/${EXTEN})}) exten => _24XX,n,Dial(SIP/${EXTEN},20) exten => _24XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?missed_call) exten => _24XX,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?missed_call) |
#exten => _24XX,n,Hangup()
![]() ; in that the queue will make all waiting callers wait in the queue ; even if there is more than one available member ready to take ; calls until the head caller is connected with the member they ; were trying to get to. The next waiting caller in line then ; becomes the head caller, and they are then connected with the ; next available member and all available members and waiting callers ; waits while this happens. The new behavior, enabled by setting ; autofill=yes makes sure that when the waiting callers are connecting ; with available members in a parallel fashion until there are ; no more available members or no more waiting callers. This is ... member => Agent/11012 ;member => SIP/240100@default,0,John Smith,SIP/1000 ; ; Note that using agent groups is probably not what you want. Strategies do ; not propagate down to the Agent system so if you want round robin, least ; recent, etc, you should list all the agents in this file individually and not ; use agent groups. ; ;member => Agent/@1 ; Any agent in group 1 ;member => Agent/:1,1 ; Any agent in group 1, wait for first |
#exten => _24XX,n,Hangup()
[general] ; ; Global settings for call queues ; ; Persistent Members ; Store each dynamic member in each queue in the astdb so that ; when asterisk is restarted, each member will be automatically ; read into their recorded queues. Default is 'yes'. ; persistentmembers = yes ... member => Agent/11012 ;member => SIP/240100@default,0,John Smith,SIP/1000 ; ; Note that using agent groups is probably not what you want. Strategies do ; not propagate down to the Agent system so if you want round robin, least ; recent, etc, you should list all the agents in this file individually and not ; use agent groups. ; ;member => Agent/@1 ; Any agent in group 1 ;member => Agent/:1,1 ; Any agent in group 1, wait for first |
#;member => SIP/2402
![]() ;recordformat=gsm ; ; The text to be added to the name of the recording. Allows forming a url link. ;urlprefix=http://localhost/calls/ ; ; The optional directory to save the conversations in. The default is ; /var/spool/asterisk/monitor ;savecallsin=/var/calls ; ; An optional custom beep sound file to play to always-connected agents. ;custom_beep=beep ; ; -------------------------------------------------- ; ; This section contains the agent definitions, in the form: ; ; agent => agentid,agentpassword,name ; agent => 1001,4321,Mark Spencer |
#;member => SIP/2402
; ;group=3 ;group=1,2 ;group= ; ; -------------------------------------------------- ; This section is devoted to recording agent's calls ; The keywords are global to the chan_agent channel driver ; ; Enable recording calls addressed to agents. It's turned off by default. ... ; An optional custom beep sound file to play to always-connected agents. ;custom_beep=beep ; ; -------------------------------------------------- ; ; This section contains the agent definitions, in the form: ; ; agent => agentid,agentpassword,name ; agent => 1001,4321,Mark Spencer |
#agent => 1002,4321,Will Meadows
![]() /etc/init.d/asterisk restart Stopping Asterisk PBX: asterisk. Starting Asterisk PBX: asterisk. |
#agent => 1002,4321,Will Meadows
/etc/init.d/asterisk restart Stopping Asterisk PBX: asterisk. Starting Asterisk PBX: asterisk. |
#asterisk -rvvvvvvvvvvvv
![]() 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 linux8 (pid = 23649) ... -- Music class requested but no musiconhold loaded. == Agent '1001' logged out == Spawn extension (gr3, 2406, 1) exited non-zero on 'SIP/2401-00000002' -- Remote UNIX connection == Spawn extension (gr3, 2405, 3) exited non-zero on 'SIP/2402-00000003' Executing last minute cleanups == Destroying musiconhold processes linux8*CLI> Disconnected from Asterisk server Executing last minute cleanups |
#asterisk -rvvvvvvvvvvvv
![]() 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 linux8 (pid = 23649) ... -- Music class requested but no musiconhold loaded. == Agent '1001' logged out == Spawn extension (gr3, 2406, 1) exited non-zero on 'SIP/2401-00000002' -- Remote UNIX connection == Spawn extension (gr3, 2405, 3) exited non-zero on 'SIP/2402-00000003' Executing last minute cleanups == Destroying musiconhold processes linux8*CLI> Disconnected from Asterisk server Executing last minute cleanups |
#asterisk -rvvvvvvvvvvvv
![]() 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 linux8 (pid = 23649) ... -- Music class requested but no musiconhold loaded. == Agent '1001' logged out == Spawn extension (gr3, 2406, 1) exited non-zero on 'SIP/2401-00000002' -- Remote UNIX connection == Spawn extension (gr3, 2405, 3) exited non-zero on 'SIP/2402-00000003' Executing last minute cleanups == Destroying musiconhold processes linux8*CLI> Disconnected from Asterisk server Executing last minute cleanups |
#asterisk -rvvvvvvvvvvvv
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 linux8 (pid = 23649) ... -- Music class requested but no musiconhold loaded. == Agent '1001' logged out == Spawn extension (gr3, 2406, 1) exited non-zero on 'SIP/2401-00000002' -- Remote UNIX connection == Spawn extension (gr3, 2405, 3) exited non-zero on 'SIP/2402-00000003' Executing last minute cleanups == Destroying musiconhold processes linux8*CLI> Disconnected from Asterisk server Executing last minute cleanups |
#exten => _24XX,n,Hangup()
![]() ;-----------------------------------LOCAL DIALPLAN exten => _24XX,1,Answer ;exten => _24XX,n,Set(MON_FILENAME=record-${STRFTIME(${EPOCH},,%Y-%m-%d-%H-%M-%S)}-${EXTEN}-${CALLER ID(num)}) exten => _24XX,n,Set(MON_FILENAME=record-${EXTEN}-${CALLERID(num)}-${STRFTIME(${EPOCH},,%Y-%m-%d-%H-%M-%S)}) (1001) ;agenti dlya ocheredei exten => _24XX,n,Monitor(wav,${MON_FILENAME},m) ;exten => _24XX,n,System(/usr/local/bin/asterisk-jabber-logger Vam zvonyat s ${CALLERID(num)}) ;exten => _24XX,n,SetMusicOnHold(${DB(moh/${EXTEN})}) exten => _24XX,n,Dial(SIP/${EXTEN},20) exten => _24XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?missed_call) exten => _24XX,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?missed_call) exten => _24XX,n,Hangup() D(num)}) ;exten => _24XX,n(missed_call),System(/usr/local/bin/asterisk-jabber-logger Propuschennyi vyzov s ${CAL exten => _24XX,n,Wait(1) ;LERID(num)}) exten => _24XX,n,Dial(SIP/kiev/${DB(mobile/${EXTEN})},10,m) exten => _24XX,n(missed_call),System(/usr/local/bin/asterisk-jabber-logger ${DB(jid/${EXTEN})} Propuschennij zvonok s ${CALLERII exten => _24XX,n(voicemail),Voicemail(${EXTEN}@default) "/etc/asterisk/extensions.conf" 219L, 7243C written |
#exten => _24XX,n,Hangup()
;-----------------------------------LOCAL DIALPLAN exten => _24XX,1,Answer ;exten => _24XX,n,Set(MON_FILENAME=record-${STRFTIME(${EPOCH},,%Y-%m-%d-%H-%M-%S)}-${EXTEN}-${CALLER ID(num)}) exten => _24XX,n,Set(MON_FILENAME=record-${EXTEN}-${CALLERID(num)}-${STRFTIME(${EPOCH},,%Y-%m-%d-%H-%M-%S)}) (1001) ;agenti dlya ocheredei exten => _24XX,n,Monitor(wav,${MON_FILENAME},m) ;exten => _24XX,n,System(/usr/local/bin/asterisk-jabber-logger Vam zvonyat s ${CALLERID(num)}) ;exten => _24XX,n,SetMusicOnHold(${DB(moh/${EXTEN})}) exten => _24XX,n,Dial(SIP/${EXTEN},20) exten => _24XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?missed_call) exten => _24XX,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?missed_call) exten => _24XX,n,Hangup() D(num)}) ;exten => _24XX,n(missed_call),System(/usr/local/bin/asterisk-jabber-logger Propuschennyi vyzov s ${CAL exten => _24XX,n,Wait(1) ;LERID(num)}) exten => _24XX,n,Dial(SIP/kiev/${DB(mobile/${EXTEN})},10,m) exten => _24XX,n(missed_call),System(/usr/local/bin/asterisk-jabber-logger ${DB(jid/${EXTEN})} Propuschennij zvonok s ${CALLERII exten => _24XX,n(voicemail),Voicemail(${EXTEN}@default) "/etc/asterisk/extensions.conf" 219L, 7243C written |
#vim /etc/asterisk/queues.conf
![]() --- /tmp/l3-saved-27557.11568.21188 2011-06-24 11:49:46.000000000 +0300 +++ /etc/asterisk/queues.conf 2011-06-24 11:50:03.000000000 +0300 @@ -499,6 +499,6 @@ ; available, but consider with penalty [queue1] -member => Agent/1101 +member => Agent/2406 ;member => SIP/2401 ;member => SIP/2402 |
#!/
![]() /etc/init.d/asterisk restart Stopping Asterisk PBX: asterisk. Starting Asterisk PBX: asterisk. |
#!/
/etc/init.d/asterisk restart Stopping Asterisk PBX: asterisk. Starting Asterisk PBX: asterisk. |
#asterisk -rvvvvvvvvvvvv
![]() 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 linux8 (pid = 23847) ... -- Executing [2405@gr3:2] SetMusicOnHold("SIP/2402-00000008", "default") in new stack -- Executing [2405@gr3:3] Queue("SIP/2402-00000008", "queue1") in new stack -- Music class requested but no musiconhold loaded. -- Remote UNIX connection == Spawn extension (gr3, 2405, 3) exited non-zero on 'SIP/2402-00000008' Executing last minute cleanups == Destroying musiconhold processes linux8*CLI> Disconnected from Asterisk server Executing last minute cleanups |
#asterisk -rvvvvvvvvvvvv
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 linux8 (pid = 23847) ... -- Executing [2405@gr3:2] SetMusicOnHold("SIP/2402-00000008", "default") in new stack -- Executing [2405@gr3:3] Queue("SIP/2402-00000008", "queue1") in new stack -- Music class requested but no musiconhold loaded. -- Remote UNIX connection == Spawn extension (gr3, 2405, 3) exited non-zero on 'SIP/2402-00000008' Executing last minute cleanups == Destroying musiconhold processes linux8*CLI> Disconnected from Asterisk server Executing last minute cleanups |
Время первой команды журнала | 09:28:44 2011- 6-24 | |||||||||||||||||||||||||||||||||||||||
Время последней команды журнала | 10:50:05 2011- 6-24 | |||||||||||||||||||||||||||||||||||||||
Количество командных строк в журнале | 101 | |||||||||||||||||||||||||||||||||||||||
Процент команд с ненулевым кодом завершения, % | 0.00 | |||||||||||||||||||||||||||||||||||||||
Процент синтаксически неверно набранных команд, % | 0.00 | |||||||||||||||||||||||||||||||||||||||
Суммарное время работы с терминалом *, час | 0.79 | |||||||||||||||||||||||||||||||||||||||
Количество командных строк в единицу времени, команда/мин | 2.14 | |||||||||||||||||||||||||||||||||||||||
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) |