12:51:39

#asterisk -rvv

Asterisk 1.8.13.1~dfsg1-3+deb7u3, Copyright (C) 1999 - 2012 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 1.8.13.1~dfsg1-3+deb7u3 currently running on debian5 (pid = 13517)
...
    -- SIP/2202-000000d9 answered SIP/2201-000000d8
    -- Started music on hold, class 'default', on SIP/2201-000000d8
    -- Stopped music on hold on SIP/2201-000000d8
    -- Started music on hold, class 'default', on SIP/2201-000000d8
    -- Stopped music on hold on SIP/2201-000000d8
    -- Started music on hold, class 'default', on SIP/2201-000000d8
  == Spawn extension (gr1, 2202, 4) exited non-zero on 'SIP/2201-000000d8'
    -- Stopped music on hold on SIP/2201-000000d8
debian5*CLI> exit
Executing last minute cleanups

/dev/pts/0

12:51:39

#asterisk -rvv

Asterisk 1.8.13.1~dfsg1-3+deb7u3, Copyright (C) 1999 - 2012 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 1.8.13.1~dfsg1-3+deb7u3 currently running on debian5 (pid = 13517)
...
    -- SIP/2202-000000d9 answered SIP/2201-000000d8
    -- Started music on hold, class 'default', on SIP/2201-000000d8
    -- Stopped music on hold on SIP/2201-000000d8
    -- Started music on hold, class 'default', on SIP/2201-000000d8
    -- Stopped music on hold on SIP/2201-000000d8
    -- Started music on hold, class 'default', on SIP/2201-000000d8
  == Spawn extension (gr1, 2202, 4) exited non-zero on 'SIP/2201-000000d8'
    -- Stopped music on hold on SIP/2201-000000d8
debian5*CLI> exit
Executing last minute cleanups

/dev/pts/3

12:53:13

#vim /etc/asterisk/musiconhold.conf

--- /tmp/l3-saved-25685.29822.6605	2014-11-26 13:53:16.771641179 +0200
+++ /etc/asterisk/musiconhold.conf	2014-11-26 13:53:26.311641291 +0200
@@ -44,7 +44,7 @@
 
 [default]
 mode=files
-directory=/var/lib/asterisk/moh/
+directory=/var/lib/asterisk/moh
 ;
 ;[native-random]
 ;mode=files

/dev/pts/0

12:53:13

#vim /etc/asterisk/musiconhold.conf

/dev/pts/3

12:54:13

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:

2201-2201-2014-11-26-13:18:38.wav  2201-2201-2014-11-26-13:22:29.wav

/dev/pts/0

12:54:13

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:

2201-2201-2014-11-26-13:18:38.wav  2201-2201-2014-11-26-13:22:29.wav

12:54:13

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13:22:29.wav

play WARN alsa: can't encode 0-bit Unknown or not applicable
/var/spool/asterisk/monitor/2201-2201-2014-11-26-13:22:29.wav:
 File Size: 12.2k     Bit Rate: 128k
  Encoding: Signed PCM
  Channels: 1 @ 16-bit
Samplerate: 8000Hz
Replaygain: off
  Duration: 00:00:00.76
In:100%  00:00:00.76 [00:00:00.00] Out:6.08k [   -==|==-   ]        Clip:0
Done.

/dev/pts/3

12:54:13

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13:22:29.wav

play WARN alsa: can't encode 0-bit Unknown or not applicable
/var/spool/asterisk/monitor/2201-2201-2014-11-26-13:22:29.wav:
 File Size: 12.2k     Bit Rate: 128k
  Encoding: Signed PCM
  Channels: 1 @ 16-bit
Samplerate: 8000Hz
Replaygain: off
  Duration: 00:00:00.76
In:100%  00:00:00.76 [00:00:00.00] Out:6.08k [   -==|==-   ]        Clip:0
Done.

12:56:00

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13:53:33.wav

play WARN alsa: can't encode 0-bit Unknown or not applicable
play FAIL formats: can't open input file `/var/spool/asterisk/monitor/2201-2201-2014-11-26-13:53:33.wav': No such file or directory

/dev/pts/0

12:56:00

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13:53:33.wav

play WARN alsa: can't encode 0-bit Unknown or not applicable
play FAIL formats: can't open input file `/var/spool/asterisk/monitor/2201-2201-2014-11-26-13:53:33.wav': No such file or directory

/dev/pts/3

12:56:17

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13:52:22.wav

play WARN alsa: can't encode 0-bit Unknown or not applicable
play FAIL formats: can't open input file `/var/spool/asterisk/monitor/2201-2201-2014-11-26-13:52:22.wav': No such file or directory

/dev/pts/0

12:56:17

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13:52:22.wav

play WARN alsa: can't encode 0-bit Unknown or not applicable
play FAIL formats: can't open input file `/var/spool/asterisk/monitor/2201-2201-2014-11-26-13:52:22.wav': No such file or directory

/dev/pts/3

12:57:31

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:

2201-2201-2014-11-26-13:18:38.wav  2201-2201-2014-11-26-13:22:29.wav

12:57:31

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:

2201-2201-2014-11-26-13:18:38.wav  2201-2201-2014-11-26-13:22:29.wav

/dev/pts/0

12:57:31

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:

2201-2201-2014-11-26-13:18:38.wav  2201-2201-2014-11-26-13:22:29.wav

12:57:31

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:

2201-2201-2014-11-26-13:18:38.wav  2201-2201-2014-11-26-13:22:29.wav

/dev/pts/3

12:57:31

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:22\:29.wav

play WARN alsa: can't encode 0-bit Unknown or not applicable
/var/spool/asterisk/monitor/2201-2201-2014-11-26-13:22:29.wav:
 File Size: 12.2k     Bit Rate: 128k
  Encoding: Signed PCM
  Channels: 1 @ 16-bit
Samplerate: 8000Hz
Replaygain: off
  Duration: 00:00:00.76
In:100%  00:00:00.76 [00:00:00.00] Out:6.08k [   -==|==-   ]        Clip:0
Done.

/dev/pts/0

12:57:31

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:22\:29.wav

play WARN alsa: can't encode 0-bit Unknown or not applicable
/var/spool/asterisk/monitor/2201-2201-2014-11-26-13:22:29.wav:
 File Size: 12.2k     Bit Rate: 128k
  Encoding: Signed PCM
  Channels: 1 @ 16-bit
Samplerate: 8000Hz
Replaygain: off
  Duration: 00:00:00.76
In:100%  00:00:00.76 [00:00:00.00] Out:6.08k [   -==|==-   ]        Clip:0
Done.

/dev/pts/3

12:57:53

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:

2201-2201-2014-11-26-13:18:38.wav  2201-2201-2014-11-26-13:22:29.wav

12:57:53

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:

2201-2201-2014-11-26-13:18:38.wav  2201-2201-2014-11-26-13:22:29.wav

12:57:53

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:

2201-2201-2014-11-26-13:18:38.wav  2201-2201-2014-11-26-13:22:29.wav

/dev/pts/0

12:57:53

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:

2201-2201-2014-11-26-13:18:38.wav  2201-2201-2014-11-26-13:22:29.wav

12:57:53

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:

2201-2201-2014-11-26-13:18:38.wav  2201-2201-2014-11-26-13:22:29.wav

12:57:53

#play /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:

2201-2201-2014-11-26-13:18:38.wav  2201-2201-2014-11-26-13:22:29.wav

/dev/pts/3

12:58:38

#rm /var/spool/asterisk/monitor/202-.wav

/dev/pts/0

12:58:38

#rm /var/spool/asterisk/monitor/202-.wav

/dev/pts/3

12:58:42

#rm /var/spool/asterisk/monitor/2202-.wav

/dev/pts/0

12:58:42

#rm /var/spool/asterisk/monitor/2202-.wav

/dev/pts/3

12:58:53

#rm /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:

2201-2201-2014-11-26-13:18:38.wav  2201-2201-2014-11-26-13:22:29.wav

12:58:53

#rm /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:

2201-2201-2014-11-26-13:18:38.wav  2201-2201-2014-11-26-13:22:29.wav

/dev/pts/0

12:58:53

#rm /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:

2201-2201-2014-11-26-13:18:38.wav  2201-2201-2014-11-26-13:22:29.wav

12:58:53

#rm /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:

2201-2201-2014-11-26-13:18:38.wav  2201-2201-2014-11-26-13:22:29.wav

/dev/pts/3

12:58:53

#rm /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:18\:38.wav

/dev/pts/0

12:58:53

#rm /var/spool/asterisk/monitor/2201-2201-2014-11-26-13\:18\:38.wav

/dev/pts/3

13:07:11

#m-a a-i dahdi

        ┌────────────────────┤ Updating cached package data ├─────────────────────┐
        │ Reading apt-cache output...                                             │
        │                                                                         │
        │                                                                         │
        │                                100%                                     │
        │                                                                         │
        └─────────────────────────────────────────────────────────────────────────┘
Updated infos about 1 packages
unpack
Extracting the package tarball, /usr/src/dahdi.tar.bz2, please wait...
"/usr/share/modass/packages/default.sh" build KVERS=3.2.0-4-amd64 KSRC=/lib/modules/3.2.0-4-amd64/build KDREV=3.2.63-2+deb7u1 kdist_image

/dev/pts/0

13:07:11

#m-a a-i dahdi

        ┌────────────────────┤ Updating cached package data ├─────────────────────┐
        │ Reading apt-cache output...                                             │
        │                                                                         │
        │                                                                         │
        │                                100%                                     │
        │                                                                         │
        └─────────────────────────────────────────────────────────────────────────┘
Updated infos about 1 packages
unpack
Extracting the package tarball, /usr/src/dahdi.tar.bz2, please wait...
"/usr/share/modass/packages/default.sh" build KVERS=3.2.0-4-amd64 KSRC=/lib/modules/3.2.0-4-amd64/build KDREV=3.2.63-2+deb7u1 kdist_image

/dev/pts/3

13:10:35

#modprobe dahdi

/dev/pts/0

13:10:35

#modprobe dahdi

/dev/pts/3

13:11:07

#lsmod | grep dahdi

dahdi                 191056  0
crc_ccitt              12347  1 dahdi

/dev/pts/0

13:11:07

#lsmod | grep dahdi

dahdi                 191056  0
crc_ccitt              12347  1 dahdi

/dev/pts/3

13:11:18

#/etc/init.d/asterisk restart

Stopping Asterisk PBX: asterisk.
Starting Asterisk PBX: asterisk.

/dev/pts/0

13:11:18

#/etc/init.d/asterisk restart

Stopping Asterisk PBX: asterisk.
Starting Asterisk PBX: asterisk.

/dev/pts/3

13:11:48

#vim /etc/

/dev/pts/0

13:11:48

#vim /etc/

/dev/pts/3

13:11:48

#vim /etc/asterisk/e

/dev/pts/0

13:11:48

#vim /etc/asterisk/e

/dev/pts/3

13:11:48

#vim /etc/asterisk/extensions.conf

--- /tmp/l3-saved-25685.30843.26477	2014-11-26 14:14:55.095656541 +0200
+++ /etc/asterisk/extensions.conf	2014-11-26 14:16:49.883657900 +0200
@@ -24,6 +24,10 @@
 
 
 
+; создаем конференц комнаты
+; комнаты 100-109
+exten => _801X,1,MeetMe(10${EXTEN:3})
+
 
 
 ;

/dev/pts/0

13:11:48

#vim /etc/asterisk/extensions.conf

/dev/pts/3

13:19:48

#vim /etc/asterisk/meetme.conf

/dev/pts/0

13:19:48

#vim /etc/asterisk/meetme.conf

/dev/pts/3

13:24:43

#dpkg -L asterisk-dahdi | grep meet

/usr/lib/asterisk/modules/app_meetme.so

/dev/pts/0

13:24:43

#dpkg -L asterisk-dahdi | grep meet

/usr/lib/asterisk/modules/app_meetme.so

/dev/pts/3

13:27:21

#asterisk -rx 'meetme restart'

No such command 'meetme restart' (type 'core show help meetme restart' for other possible commands)

/dev/pts/0

13:27:21

#asterisk -rx 'meetme restart'

No such command 'meetme restart' (type 'core show help meetme restart' for other possible commands)

/dev/pts/12

13:35:49

#less /var/log/asterisk/messages

/dev/pts/3

13:36:09

#[ ]

Asterisk 1.8.13.1~dfsg1-3+deb7u3, Copyright (C) 1999 - 2012 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 1.8.13.1~dfsg1-3+deb7u3 currently running on debian5 (pid = 2481)
Verbosity was 0 and is now 2
  == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
debian5*CLI> exit
Executing last minute cleanups

/dev/pts/0

13:36:09

#[ ]

Asterisk 1.8.13.1~dfsg1-3+deb7u3, Copyright (C) 1999 - 2012 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 1.8.13.1~dfsg1-3+deb7u3 currently running on debian5 (pid = 2481)
Verbosity was 0 and is now 2
  == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
debian5*CLI> exit
Executing last minute cleanups

/dev/pts/12

13:36:21

#asterisk -rvv

Asterisk 1.8.13.1~dfsg1-3+deb7u3, Copyright (C) 1999 - 2012 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 1.8.13.1~dfsg1-3+deb7u3 currently running on debian5 (pid = 2481)
...
  == Using SIP RTP CoS mark 5
[Nov 26 14:45:19] WARNING[2868]: chan_sip.c:5456 create_addr: Purely numeric hostname (2290), and not a peer--rejecting!
[Nov 26 14:45:19] WARNING[2868]: app_dial.c:2274 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
[Nov 26 14:45:19] WARNING[2868]: app_voicemail.c:5660 leave_voicemail: No entry in voicemail config file for '2290'
debian5*CLI>
debian5*CLI>
debian5*CLI>
debian5*CLI> quit
Executing last minute cleanups

/dev/pts/3

13:39:13

#vim /etc/asterisk/extensions.conf

--- /tmp/l3-saved-25685.31702.23569	2014-11-26 14:39:41.519674133 +0200
+++ /etc/asterisk/extensions.conf	2014-11-26 14:44:10.311677319 +0200
@@ -8,8 +8,13 @@
 
 
 
+
+
  
-; группы звонков
+;======= группы звонков =====
+
+
+
 
 [internal]
 
@@ -23,18 +28,15 @@
 ;exten => _82XX,1,Playback(/var/tmp/asterisk/record${EXTEN:2})
 
 
-
 ; создаем конференц комнаты
 ; комнаты 100-109
 exten => _801X,1,MeetMe(10${EXTEN:3})
 
 
-
 ;
 ;exten => 8001,1,VoiceMailMain()
 
 
-
 ; сразу переход на свой голосовой почтовый ящик c запросом пароля
 exten => _83XX,1,VoiceMailMain(22${EXTEN:2}@default)
 
@@ -46,6 +48,8 @@
 
 
 
+
+
 [local]
 
 ; запись звонков
@@ -59,7 +63,6 @@
 exten => _22XX,n,Dial(SIP/${EXTEN},5)
 
 
-
 ; использование переменной ${EXTEN}
 ;exten => _22XX,1,NoOp(Starting call processing)
 ;exten => _22XX,n,NoOp(to extension {EXTEN})
@@ -71,18 +74,20 @@
 
 
 
-; здесь будет голосовое меню
-;exten => 
-
+[national]
 
+exten => _21XX,1,Dial(SIP/ny/${EXTEN})
+exten => _1XXX,1,Dial(SIP/ny/${EXTEN})
 
 
 
-[national]
-exten => _21XX,1,Dial(SIP/ny/${EXTEN})
-exten => _1XXX,1,Dial(SIP/ny/${EXTEN})
 
 [international]
+
+; международная конференция
+exten => _229X,1,NoOp(Conferences for external users)
+exten => _229X,n,MeetMe(10${EXTEN:3})
+
 exten => _3XXX,1,Dial(SIP/london/${EXTEN})
 exten => _23XX,1,Dial(SIP/london/${EXTEN})

/dev/pts/0

13:39:13

#vim /etc/asterisk/extensions.conf

/dev/pts/3

13:44:11

#asterisk -rvv

Asterisk 1.8.13.1~dfsg1-3+deb7u3, Copyright (C) 1999 - 2012 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 1.8.13.1~dfsg1-3+deb7u3 currently running on debian5 (pid = 2481)
Verbosity is at least 2
  == Using SIP RTP CoS mark 5
[Nov 26 14:44:24] WARNING[2848]: chan_sip.c:5456 create_addr: Purely numeric hostname (2290), and not a peer--rejecting!
[Nov 26 14:44:24] WARNING[2848]: app_dial.c:2274 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
[Nov 26 14:44:24] WARNING[2848]: app_voicemail.c:5660 leave_voicemail: No entry in voicemail config file for '2290'
debian5*CLI> exit
Executing last minute cleanups

/dev/pts/0

13:44:11

#asterisk -rvv

Asterisk 1.8.13.1~dfsg1-3+deb7u3, Copyright (C) 1999 - 2012 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 1.8.13.1~dfsg1-3+deb7u3 currently running on debian5 (pid = 2481)
Verbosity is at least 2
  == Using SIP RTP CoS mark 5
[Nov 26 14:44:24] WARNING[2848]: chan_sip.c:5456 create_addr: Purely numeric hostname (2290), and not a peer--rejecting!
[Nov 26 14:44:24] WARNING[2848]: app_dial.c:2274 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
[Nov 26 14:44:24] WARNING[2848]: app_voicemail.c:5660 leave_voicemail: No entry in voicemail config file for '2290'
debian5*CLI> exit
Executing last minute cleanups

/dev/pts/3

13:45:50

#;conf => 2345,9938

[local]
; запись звонков
exten => _22XX,1,NoOp(запись звонков)
;exten => _22XX,n,NoOp(to extension {EXTEN})
;exten => _22XX,n,Dial(SIP/${EXTEN},5)
; если через 5 сек не подняли трубку тогда переходим на голосовую почту
; включить головую почту, в ящик ${EXTEN}
exten => _22XX,n,VoiceMail(22${EXTEN:2}@default)
[national]
[        ]
...
include => national
[gr3]
[gr3]
include => local
; для транзитных звонков
[gr4]
; использование same
;exten => _22XX,1,NoOp(Starting call processing)
; same =>       n,NoOp(to extension {EXTEN})
include => international                                                  116,36        98%

/dev/pts/0

13:45:50

#;conf => 2345,9938

E325: ATTENTION
Found a swap file by the name "/etc/asterisk/.extensions.conf.swp"
          owned by: root   dated: Tue Nov 25 15:44:27 2014
         file name: /etc/asterisk/extensions.conf
          modified: YES
         user name: root   host name: debian5
        process ID: 18251 (still running)
While opening file "/etc/asterisk/extensions.conf"
             dated: Wed Nov 26 14:44:10 2014
;exten => _81XX,1,Record(/var/tmp/asterisk/record${EXTEN:2}:gsm)
...
include => national
[gr3]
[gr3]
include => local
; для транзитных звонков
[gr4]
; использование same
;exten => _22XX,1,NoOp(Starting call processing)
; same =>       n,NoOp(to extension {EXTEN})
include => internationaltarting call processing)                          116,36        98%

/dev/pts/3

13:50:06

#/etc/init.d/asterisk restart

Stopping Asterisk PBX: asterisk.
Starting Asterisk PBX: asterisk.

/dev/pts/0

13:50:06

#/etc/init.d/asterisk restart

Stopping Asterisk PBX: asterisk.
Starting Asterisk PBX: asterisk.

/dev/pts/4

13:52:33

#vim /etc/asterisk/extensions.conf

--- /tmp/l3-saved-3963.1695.18127	2014-11-26 14:52:41.887683373 +0200
+++ /etc/asterisk/extensions.conf	2014-11-26 15:58:16.935729946 +0200
@@ -19,6 +19,8 @@
 [internal]
 
 
+; звонок на группу номеров
+exten => 2289,1,Dial(SIP/3201&SIP/3202)
 
 ; сначала записать сообщение, затем сразу его проиграть на том-же номере
 exten => _81XX,1,Record(/var/tmp/asterisk/record${EXTEN:2}:gsm,3)
@@ -52,6 +54,7 @@
 
 [local]
 
+
 ; запись звонков
 exten => _22XX,1,NoOp(запись звонков)
 ; Monitor - программа записи взонков
@@ -60,7 +63,14 @@
 ; set- записать в переменную CALLFILENAME, все что после
 exten => _22XX,n,Set(CALLFILENAME=${EXTEN}-${CALLERID(num)}-${STRFTIME(${EPOCH},Europe/Kiev,"%Y-%m-%d-%H:%M:%S")})
 exten => _22XX,n,Monitor(wav,${CALLFILENAME},m)
-exten => _22XX,n,Dial(SIP/${EXTEN},5)
+exten => _22XX,n,Dial(SIP/${EXTEN},10)
+
+
+; международная конференция
+exten => _229X,1,NoOp(Conferences for external users)
+exten => _229X,n,MeetMe(10${EXTEN:3})
+; Hangup - гарантировано повесить трубку по окончанию конференции
+exten => _229X,n,Hangupi
 
 
 ; использование переменной ${EXTEN}
@@ -77,6 +87,7 @@
 [national]
 
 exten => _21XX,1,Dial(SIP/ny/${EXTEN})
+
 exten => _1XXX,1,Dial(SIP/ny/${EXTEN})
 
 
@@ -84,11 +95,8 @@
 
 [international]
 
-; международная конференция
-exten => _229X,1,NoOp(Conferences for external users)
-exten => _229X,n,MeetMe(10${EXTEN:3})
-
 exten => _3XXX,1,Dial(SIP/london/${EXTEN})
+
 exten => _23XX,1,Dial(SIP/london/${EXTEN})
 
 [gr1]

/dev/pts/3

14:37:21

#asterisk -rvv

Asterisk 1.8.13.1~dfsg1-3+deb7u3, Copyright (C) 1999 - 2012 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 1.8.13.1~dfsg1-3+deb7u3 currently running on debian5 (pid = 3091)
...
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/2201-0000000c' status is 'CHANUNAVAIL'
  == Using SIP RTP CoS mark 5
    -- Executing [1292@gr1:1] Dial("SIP/2201-0000000d", "SIP/ny/1292") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/ny/1292
    -- SIP/ny-0000000e answered SIP/2201-0000000d
    -- Locally bridging SIP/2201-0000000d and SIP/ny-0000000e
debian5*CLI> exit
Executing last minute cleanups

/dev/pts/0

14:37:21

#asterisk -rvv

Asterisk 1.8.13.1~dfsg1-3+deb7u3, Copyright (C) 1999 - 2012 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 1.8.13.1~dfsg1-3+deb7u3 currently running on debian5 (pid = 3091)
...
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/2201-0000000c' status is 'CHANUNAVAIL'
  == Using SIP RTP CoS mark 5
    -- Executing [1292@gr1:1] Dial("SIP/2201-0000000d", "SIP/ny/1292") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/ny/1292
    -- SIP/ny-0000000e answered SIP/2201-0000000d
    -- Locally bridging SIP/2201-0000000d and SIP/ny-0000000e
debian5*CLI> exit
Executing last minute cleanups

/dev/pts/3

14:42:19

#vim /etc/asterisk/extensions.conf

--- /tmp/l3-saved-25685.11572.18737	2014-11-26 15:42:21.243718638 +0200
+++ /etc/asterisk/extensions.conf	2014-11-26 15:42:55.603719043 +0200
@@ -54,6 +54,7 @@
 
 [local]
 
+exten => 2289,1,Dial(SIP/3201&SIP/3202)
 
 ; запись звонков
 exten => _22XX,1,NoOp(запись звонков)
@@ -92,9 +93,11 @@
 
 
 
+exten => 2289,1,Dial(SIP/3201&SIP/3202)
 
 [international]
 
+exten => 2289,1,Dial(SIP/3201&SIP/3202)
 exten => _3XXX,1,Dial(SIP/london/${EXTEN})
 
 exten => _23XX,1,Dial(SIP/london/${EXTEN})

/dev/pts/0

14:42:19

#vim /etc/asterisk/extensions.conf

/dev/pts/3

14:51:24

#asterisk -rx 'dialplan reload'

Dialplan reloaded.

/dev/pts/0

14:51:24

#asterisk -rx 'dialplan reload'

Dialplan reloaded.

/dev/pts/3

14:52:46

#asterisk -rx "features reload"

/dev/pts/0

14:52:46

#asterisk -rx "features reload"

/dev/pts/3

14:53:33

#asterisk -rvv

Asterisk 1.8.13.1~dfsg1-3+deb7u3, Copyright (C) 1999 - 2012 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 1.8.13.1~dfsg1-3+deb7u3 currently running on debian5 (pid = 3091)
...
    -- Executing [2202@gr1:3] Monitor("SIP/ny-00000029", "wav,2202-3101-"2014-11-26-15:53:59",m") in new stack
    -- Executing [2202@gr1:4] Dial("SIP/ny-00000029", "SIP/2202,10") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/2202
    -- SIP/2202-0000002a is ringing
    -- SIP/2202-0000002a is ringing
  == Using SIP RTP CoS mark 5
  == Spawn extension (gr1, 2202, 4) exited non-zero on 'SIP/ny-00000029'
debian5*CLI> exit
Executing last minute cleanups

/dev/pts/0

14:53:33

#asterisk -rvv

Asterisk 1.8.13.1~dfsg1-3+deb7u3, Copyright (C) 1999 - 2012 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 1.8.13.1~dfsg1-3+deb7u3 currently running on debian5 (pid = 3091)
...
    -- Executing [2202@gr1:3] Monitor("SIP/ny-00000029", "wav,2202-3101-"2014-11-26-15:53:59",m") in new stack
    -- Executing [2202@gr1:4] Dial("SIP/ny-00000029", "SIP/2202,10") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/2202
    -- SIP/2202-0000002a is ringing
    -- SIP/2202-0000002a is ringing
  == Using SIP RTP CoS mark 5
  == Spawn extension (gr1, 2202, 4) exited non-zero on 'SIP/ny-00000029'
debian5*CLI> exit
Executing last minute cleanups

/dev/pts/3

14:58:51

#[ ]

Asterisk 1.8.13.1~dfsg1-3+deb7u3, Copyright (C) 1999 - 2012 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 1.8.13.1~dfsg1-3+deb7u3 currently running on debian5 (pid = 3091)
...
  == Everyone is busy/congested at this time (2:0/0/2)
    -- Executing [2289@gr1:2] Set("SIP/2201-00000030", "CALLFILENAME=2289-2201-"2014-11-26-15:59:00"") in new stack
    -- Executing [2289@gr1:3] Monitor("SIP/2201-00000030", "wav,2289-2201-"2014-11-26-15:59:00",m") in new stack
    -- Executing [2289@gr1:4] Dial("SIP/2201-00000030", "SIP/2289,10") in new stack
[Nov 26 15:59:00] WARNING[4283]: chan_sip.c:5456 create_addr: Purely numeric hostname (2289), and not a peer--rejecting!
[Nov 26 15:59:00] WARNING[4283]: app_dial.c:2274 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/2201-00000030' status is 'CHANUNAVAIL'
debian5*CLI> exit
Executing last minute cleanups

/dev/pts/0

14:58:51

#[ ]

Asterisk 1.8.13.1~dfsg1-3+deb7u3, Copyright (C) 1999 - 2012 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 1.8.13.1~dfsg1-3+deb7u3 currently running on debian5 (pid = 3091)
...
  == Everyone is busy/congested at this time (2:0/0/2)
    -- Executing [2289@gr1:2] Set("SIP/2201-00000030", "CALLFILENAME=2289-2201-"2014-11-26-15:59:00"") in new stack
    -- Executing [2289@gr1:3] Monitor("SIP/2201-00000030", "wav,2289-2201-"2014-11-26-15:59:00",m") in new stack
    -- Executing [2289@gr1:4] Dial("SIP/2201-00000030", "SIP/2289,10") in new stack
[Nov 26 15:59:00] WARNING[4283]: chan_sip.c:5456 create_addr: Purely numeric hostname (2289), and not a peer--rejecting!
[Nov 26 15:59:00] WARNING[4283]: app_dial.c:2274 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/2201-00000030' status is 'CHANUNAVAIL'
debian5*CLI> exit
Executing last minute cleanups

/dev/pts/4

14:58:51

#{EXTEN:2})

;exten => _22XX,n,NoOp(to extension {EXTEN})
;exten => _22XX,n,Dial(SIP/${EXTEN},5)
; если через 5 сек не подняли трубку тогда переходим на голосовую почту
; включить головую почту, в ящик ${EXTEN}
exten => _22XX,n,VoiceMail(22${EXTEN:2}@default)
[national]
exten => _21XX,1,Dial(SIP/ny/${EXTEN})
; контекст плана звонков default
[default]
; входящий вызов по сипу на номера 21(00-99)
...
; set- записать в переменную CALLFILENAME, все что после
exten => _22XX,n,Set(CALLFILENAME=${EXTEN}-${CALLERID(num)}-${STRFTIME(${EPOCH},Europe/Kiev,"%Y-%m-%d-%H::
; международная конференция
exten => _229X,1,NoOp(Conferences for external users)
exten => _229X,n,MeetMe(10${EXTEN:3})
; Hangup - гарантировано повесить трубку по окончанию конференции
exten => _229X,n,Hangupi
; использование переменной ${EXTEN}
;exten => _22XX,1,NoOp(Starting call processing)
;exten => _22XX,n,NoOp(to extension {EXTEN})

/dev/pts/3

15:00:25

#{EXTEN:2}@default,s)

[phone]´ÐºÐ»ÑÑÐµÐ½Ð¸Ðµ Ðº Ð´ÑÑÐ³Ð¾Ð¼Ñ Ð³Ð¾ÑÐ¾Ð´Ñ
[phone]
; Ð¿Ð¾Ð´ÐºÐ»ÑÑÐ°ÑÑÑ Ðº NY
[ny](phone)»Ñ Ð¿Ð¾Ð´ÐºÐ»ÑÑÐµÐ½Ð¸Ñ
; Ð½Ð¾Ð²Ð°Ñ Ð¾Ð¿ÑÐ¸Ñ
; directmedia - Ð·Ð°Ð¿ÑÐµÑ Ð¿ÑÑÐ¼Ð¾Ð³Ð¾ ÑÐ¾ÐµÐ´Ð¸Ð½ÐµÐ½Ð¸Ñ Ð¼ÐµÐ¶Ð´Ñ Ð¿Ð¸ÑÐ°Ð¼Ð¸ (ÑÐµÐ»ÐµÑÐ¾Ð½-ÑÐµÐ»ÐµÑÐ¾Ð½)
; ÑÐ°Ð±Ð¾ÑÐ°ÑÑ ÑÐµÐ»ÐµÑÐ¾Ð½Ð°Ð¼ ÑÐµÐ·ÑÐµÑÐµÐ½Ð¾ ÑÐ¾Ð»ÑÐºÐ¾ ÑÐµÑÐµÐ· asterisk
directmedia=no
"/etc/asterisk/sip.conf" 70L, 1551C
context=gr4
...
[2201]
[2203](phone)
[2203]
[2202]
language=es
callgroup=1
pickupgroup=1
[2203](phone)
qualify=no
qualify=no                                                                57,0-1        50%

/dev/pts/0

15:00:25

#{EXTEN:2}@default,s)

[2201](phone)
context=gr1
mailbox=2201@default
[2201]
[ny]1]group=1
[ny]
[2202](phone)
[2202]t=gr1
[2202]
[london]2202@default
...
[2201]
[2203](phone)
[2203]
[2202]
language=es
callgroup=1
pickupgroup=1
[2203](phone)
qualify=no
qualify=no                                                                57,0-1        50%

/dev/pts/3

15:30:12

#vim /etc/asterisk/extensions.conf

--- /tmp/l3-saved-25685.10771.9149	2014-11-26 16:30:15.767752660 +0200
+++ /etc/asterisk/extensions.conf	2014-11-26 16:33:04.623754657 +0200
@@ -18,6 +18,10 @@
 [internal]
 
 
+; парковка вызова
+include => parkedcalls
+
+
 ; звонок на группу номеров
 exten => 2289,1,Dial(SIP/london/3201&SIP/london/3202)
 
@@ -47,6 +51,8 @@
 
 
 
+
+
 [local]
 
 ;разрешаем трансфер номера в Dial  опцию tT
@@ -80,6 +86,8 @@
 
 
 
+
+
 [national]
 
 exten => _21XX,1,Dial(SIP/ny/${EXTEN})
@@ -88,6 +96,8 @@
 
 
 
+
+
 
 [international]

/dev/pts/0

15:30:12

#vim /etc/asterisk/extensions.conf

/dev/pts/3

15:33:04

#vim /etc/asterisk/extensions.conf

/dev/pts/0

15:33:04

#vim /etc/asterisk/extensions.conf

/dev/pts/3

15:58:29

#[ ]

adele.gsm             gnash-cookies.Oyrqlu  l3-saved-18147.31619.29581  pulse-5wvCE7nRDBB6
adele.wav             gnash-cookies.QJQoPc  l3-saved-3963.30414.573     someone-called-8300
gnash-cookies.MhUKUM  gnash-cookies.RCgYiZ  orbit-user                  ssh-dR9djQdIcMIv
gnash-cookies.nJ93a2  gnash-cookies.uhQmOf  pulse-3fVjlTkJObDi          tracker-user

/dev/pts/0

15:58:29

#[ ]

adele.gsm             gnash-cookies.Oyrqlu  l3-saved-18147.31619.29581  pulse-5wvCE7nRDBB6
adele.wav             gnash-cookies.QJQoPc  l3-saved-3963.30414.573     someone-called-8300
gnash-cookies.MhUKUM  gnash-cookies.RCgYiZ  orbit-user                  ssh-dR9djQdIcMIv
gnash-cookies.nJ93a2  gnash-cookies.uhQmOf  pulse-3fVjlTkJObDi          tracker-user

/dev/pts/3

15:58:37

#vim /etc/asterisk/extensions.conf

/dev/pts/0

15:58:37

#vim /etc/asterisk/extensions.conf

/dev/pts/3

15:59:00

#vim /etc/asterisk/extensions.conf

/dev/pts/0

15:59:00

#vim /etc/asterisk/extensions.conf

/dev/pts/3

15:59:56

#vim /etc/asterisk/extensions.conf

/dev/pts/0

15:59:56

#vim /etc/asterisk/extensions.conf

/dev/pts/3

16:00:51

#[ ]

hello

/dev/pts/0

16:00:51

#[ ]

hello

16:00:55

#vim /etc/asterisk/extensions.conf

/dev/pts/3

16:00:55

#vim /etc/asterisk/extensions.conf

16:01:52

#asterisk -rvv

Asterisk 1.8.13.1~dfsg1-3+deb7u3, Copyright (C) 1999 - 2012 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 1.8.13.1~dfsg1-3+deb7u3 currently running on debian5 (pid = 3091)
Verbosity is at least 3
  == Using SIP RTP CoS mark 5
    -- Executing [8301@gr1:1] System("SIP/2201-0000006e", "echo 2201 > touch /tmp/someone-called-8301") in new stack
    -- Auto fallthrough, channel 'SIP/2201-0000006e' status is 'UNKNOWN'
debian5*CLI> exit
Executing last minute cleanups

/dev/pts/0

16:01:52

#asterisk -rvv

Asterisk 1.8.13.1~dfsg1-3+deb7u3, Copyright (C) 1999 - 2012 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 1.8.13.1~dfsg1-3+deb7u3 currently running on debian5 (pid = 3091)
Verbosity is at least 3
  == Using SIP RTP CoS mark 5
    -- Executing [8301@gr1:1] System("SIP/2201-0000006e", "echo 2201 > touch /tmp/someone-called-8301") in new stack
    -- Auto fallthrough, channel 'SIP/2201-0000006e' status is 'UNKNOWN'
debian5*CLI> exit
Executing last minute cleanups

/dev/pts/3

16:05:06

#ls /tmp/

adele.gsm             gnash-cookies.Oyrqlu  l3-saved-18147.31619.29581  pulse-5wvCE7nRDBB6
adele.wav             gnash-cookies.QJQoPc  l3-saved-3963.30414.573     someone-called-8300
gnash-cookies.MhUKUM  gnash-cookies.RCgYiZ  orbit-user                  ssh-dR9djQdIcMIv
gnash-cookies.nJ93a2  gnash-cookies.uhQmOf  pulse-3fVjlTkJObDi          tracker-user

Статистика

Справка

О программе

LiLaLo (L3) расшифровывается как Live Lab Log.
Программа разработана для повышения эффективности обучения Unix/Linux-системам.
(c) Игорь Чубин, 2004-2008

Журнал лабораторных работ

Журнал

Среда (11/26/14)

Статистика

Справка

О программе