Журнал лабораторных работ

Содержание

Журнал

Понедельник (10/17/11)

/dev/tty2
13:15:27
$su -
Password:
l3-agent is already running: pid=3298; pidfile=/root/.lilalo/l3-agent.pid
13:15:41
##otlichno

13:15:51
#apt-
apt-cache             apt-config            apt-ftparchive        apt-key               apt-mark
apt-cdrom             apt-extracttemplates  apt-get               apt-listchanges       apt-sortpkgs
13:15:51
#apt-cache search asterisk
asterisk-mobile - bluetooth mobile devices support for Asterisk
asterisk-mp3 - MP3 format support (format_mp3) for the Asterisk PBX
asterisk-mysql - MySQL support for the Asterisk PBX (cdr mainly)
asterisk-ooh323c - H.323 protocol support for Asterisk (ooh323c stack)
asterisk-chan-capi - Common ISDN API 2.0 implementation for Asterisk
asterisk-core-sounds-en-g722 - asterisk PBX sound files - English/g722
asterisk-core-sounds-en-gsm - asterisk PBX sound files - English/gsm
asterisk-core-sounds-en-wav - asterisk PBX sound files - English/wav
asterisk-core-sounds-es-g722 - asterisk PBX sound files - Spanish/g722
asterisk-core-sounds-es-gsm - asterisk PBX sound files - Spanish/gsm
...
asterisk - Open Source Private Branch Exchange (PBX)
dahdi-linux - DAHDI telephony interface - Linux userspace parts
dahdi-source - DAHDI telephony interface - source code for kernel driver
dahdi - utilities for using the DAHDI kernel modules
iaxmodem - software modem with IAX2 connectivity
libasterisk-agi-perl - Collections of Perl modules to be used with Asterisk PBX AGI
libnetsds-perl - Service Delivery Suite framework
op-panel - switchboard type application for the Asterisk PBX
libopenr2-3 - MFC/R2 (telephony) call setup library
python-asterisk - Asterisk Manager API interface module for Python
13:17:47
#apt-get install asterisk asterisk-config asterisk-sounds-main asterisk-sounds-extra
Package configuration
 ┌──────────────────────────────────────────────────┤ Configuring libvpb0 ├──────────────────────────────────────────────────┐
 │ This is the numeric code for the region your phone system will be operating in (eg. 61 for Australia or 33 for France).   │
 │ It is used to configure the default regional standards that Voicetronix telephony hardware should comply with.            │
 │                                                                                                                           │
 │ ITU-T telephone code:                                                                                                     │
 │                                                                                                                           │
 â”3801_______________________________________________________________________________________________________________________ │
 │                                                                                                                           │
 │                                                        <Ok>k>                                                             │
...
Setting up unixodbc (2.2.14p2-1) ...
Setting up asterisk (1:1.6.2.9-2+squeeze3) ...
Adding system user for Asterisk
Adding user `asterisk' to group `dialout' ...
Adding user asterisk to group dialout
Done.
Adding user `asterisk' to group `audio' ...
Adding user asterisk to group audio
Done.
Starting Asterisk PBX: asterisk.
13:23:43
#/etc/init.d/asterisk restart
Stopping Asterisk PBX: asterisk.
Starting Asterisk PBX: asterisk.
прошло 19 минут
13:43:37
#ps aux | grep ast
asterisk  7811  1.2  0.2  48484 18208 ?        Ssl  14:43   0:00 /usr/sbin/asterisk -p -U asterisk
asterisk  7812  0.0  0.0   2936   564 ?        S    14:43   0:00 astcanary /var/run/asterisk/alt.asterisk.canary.tweet.tweet.tweet 7811
root      7861  0.0  0.0   3908   724 pts/4    S+   14:44   0:00 grep ast
13:44:11
#asterisk -rvv
Asterisk 1.6.2.9-2+squeeze3, 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+squeeze3 currently running on linux5 (pid = 7811)
...
group        gtalk        hangup       help         http         iax2         indication   jabber       jingle
keys         local        logger       manager      meetme       mfcr2        mgcp         minivm       mixmonitor
module       moh          no           odbc         originate    parkedcalls  phoneprov    pri          queue
realtime     reload       rtcp         rtp          say          sip          skinny       sla          sqlite
ss7          stun         timing       transcoder   udptl        ulimit       unistim      voicemail
linux5*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]
linux5*CLI> quit
Executing last minute cleanups
13:51:55
#vim /etc/network/interfaces
прошло 17 минут
14:09:02
#~
# and how to activate them. For more information, see interfaces(5).
# The loopback network interface
auto lo
iface l address 192.168.50.1
        netmask 255.255.255.0
        network 192.168.50.0
        broadcast 192.168.50.255
# The p gateway 192.168.50.254
        # dns-* options are implemented by the resolvconf package, if installed
allow-h dns-nameservers 10.0.35.1
...
~
~
~
~
~
~
~
~
~
"/etc/network/interfaces" 18L, 516C written
14:09:47
#vim /etc/network/interfaces
14:10:04
#~
SIOCDELRT: No such process
14:11:14
#ifconfig
eth0      Link encap:Ethernet  HWaddr 2c:27:d7:29:6d:0d
          inet addr:192.168.50.1  Bcast:192.168.50.255  Mask:255.255.255.0
          inet6 addr: fe80::2e27:d7ff:fe29:6d0d/64 Scope:Link
          UP BROADCAST MULTICAST  MTU:1500  Metric:1
          RX packets:564022 errors:0 dropped:0 overruns:0 frame:0
          TX packets:278491 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:784187912 (747.8 MiB)  TX bytes:20622600 (19.6 MiB)
          Interrupt:20 Memory:fe400000-fe420000
lo        Link encap:Local Loopback
          inet addr:127.0.0.1  Mask:255.0.0.0
          inet6 addr: ::1/128 Scope:Host
          UP LOOPBACK RUNNING  MTU:16436  Metric:1
          RX packets:25 errors:0 dropped:0 overruns:0 frame:0
          TX packets:25 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:1938 (1.8 KiB)  TX bytes:1938 (1.8 KiB)
14:11:34
#ifdonw
bash: ifdonw: command not found
14:11:43
#ifdonw eth0
bash: ifdonw: command not found
14:11:49
#ifdown eth0

14:11:54
#ifup eth0

14:12:09
#ifconfig
eth0      Link encap:Ethernet  HWaddr 2c:27:d7:29:6d:0d
          inet addr:192.168.50.1  Bcast:192.168.50.255  Mask:255.255.255.0
          UP BROADCAST MULTICAST  MTU:1500  Metric:1
          RX packets:564022 errors:0 dropped:0 overruns:0 frame:0
          TX packets:278491 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:784187912 (747.8 MiB)  TX bytes:20622600 (19.6 MiB)
          Interrupt:20 Memory:fe400000-fe420000
lo        Link encap:Local Loopback
          inet addr:127.0.0.1  Mask:255.0.0.0
          inet6 addr: ::1/128 Scope:Host
          UP LOOPBACK RUNNING  MTU:16436  Metric:1
          RX packets:42 errors:0 dropped:0 overruns:0 frame:0
          TX packets:42 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:3508 (3.4 KiB)  TX bytes:3508 (3.4 KiB)
14:12:19
#route
Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
default         192.168.50.254  0.0.0.0         UG    0      0        0 eth0
192.168.50.0    *               255.255.255.0   U     0      0        0 eth0
14:14:00
#route
Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
^[[Adefault         192.168.50.254  0.0.0.0         UG    0      0        0 eth0
192.168.50.0    *               255.255.255.0   U     0      0        0 eth0
14:14:18
#route
Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
^C
14:14:31
#ping 10.0.35.1
PING 10.0.35.1 (10.0.35.1) 56(84) bytes of data.
^C
--- 10.0.35.1 ping statistics ---
6 packets transmitted, 0 received, 100% packet loss, time 5039ms
14:14:42
#route
Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
default         192.168.50.254  0.0.0.0         UG    0      0        0 eth0
192.168.50.0    *               255.255.255.0   U     0      0        0 eth0
14:15:10
#route -n
Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
0.0.0.0         192.168.50.254  0.0.0.0         UG    0      0        0 eth0
192.168.50.0    0.0.0.0         255.255.255.0   U     0      0        0 eth0
14:15:13
#ping 192.168.50.1
PING 192.168.50.1 (192.168.50.1) 56(84) bytes of data.
64 bytes from 192.168.50.1: icmp_req=1 ttl=64 time=0.021 ms
64 bytes from 192.168.50.1: icmp_req=2 ttl=64 time=0.014 ms
^C
--- 192.168.50.1 ping statistics ---
2 packets transmitted, 2 received, 0% packet loss, time 999ms
rtt min/avg/max/mdev = 0.014/0.017/0.021/0.005 ms
14:15:25
#ping 192.168.50.254
PING 192.168.50.254 (192.168.50.254) 56(84) bytes of data.
64 bytes from 192.168.50.254: icmp_req=1 ttl=64 time=0.984 ms
64 bytes from 192.168.50.254: icmp_req=2 ttl=64 time=0.639 ms
^C
--- 192.168.50.254 ping statistics ---
2 packets transmitted, 2 received, 0% packet loss, time 1001ms
rtt min/avg/max/mdev = 0.639/0.811/0.984/0.174 ms
14:15:32
#route
Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
^C
14:16:39
#route
Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
default         192.168.50.254  0.0.0.0         UG    0      0        0 eth0
192.168.50.0    *               255.255.255.0   U     0      0        0 eth0
14:16:49
#ping 8.8.8.8
PING 8.8.8.8 (8.8.8.8) 56(84) bytes of data.
64 bytes from 8.8.8.8: icmp_req=1 ttl=54 time=53.8 ms
64 bytes from 8.8.8.8: icmp_req=2 ttl=54 time=39.3 ms
64 bytes from 8.8.8.8: icmp_req=3 ttl=54 time=39.0 ms
64 bytes from 8.8.8.8: icmp_req=4 ttl=54 time=39.1 ms
64 bytes from 8.8.8.8: icmp_req=5 ttl=54 time=39.2 ms
^C
--- 8.8.8.8 ping statistics ---
5 packets transmitted, 5 received, 0% packet loss, time 4005ms
rtt min/avg/max/mdev = 39.023/42.133/53.898/5.888 ms
14:17:30
#route
Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
default         192.168.50.254  0.0.0.0         UG    0      0        0 eth0
192.168.50.0    *               255.255.255.0   U     0      0        0 eth0
14:17:38
#route -n
Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
0.0.0.0         192.168.50.254  0.0.0.0         UG    0      0        0 eth0
192.168.50.0    0.0.0.0         255.255.255.0   U     0      0        0 eth0
14:17:41
#route
Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
default         192.168.50.254  0.0.0.0         UG    0      0        0 eth0
192.168.50.0    *               255.255.255.0   U     0      0        0 eth0
14:20:52
#apt-cache
.aptitude/         .bash_history      .bashrc            l3prompt           .lilalo/           .ssh/
.asterisk_history  .bash_profile      l3bashrc           .l3rc              .profile           .viminfo
14:20:52
#apt-cache
apt 0.8.10.3 for i386 compiled on Apr 15 2011 21:02:14
Usage: apt-cache [options] command
       apt-cache [options] add file1 [file2 ...]
       apt-cache [options] showpkg pkg1 [pkg2 ...]
       apt-cache [options] showsrc pkg1 [pkg2 ...]
apt-cache is a low-level tool used to manipulate APT's binary
cache files, and query information from them
Commands:
   add - Add a package file to the source cache
   gencaches - Build both the package and source cache
...
   policy - Show policy settings
Options:
  -h   This help text.
  -p=? The package cache.
  -s=? The source cache.
  -q   Disable progress indicator.
  -i   Show only important deps for the unmet command.
  -c=? Read this configuration file
  -o=? Set an arbitrary configuration option, eg -o dir::cache=/tmp
See the apt-cache(8) and apt.conf(5) manual pages for more information.
прошло 37 минут
14:58:38
#apt-cache dhcp
E: Invalid operation dhcp
14:58:58
#apt-cache dshow hcp
E: Invalid operation dshow
14:59:03
#apt-cache show dhcp
N: Can't select versions from package 'dhcp' as it purely virtual
N: No packages found
14:59:08
#apt-cache show
E: No packages found
14:59:20
#apt-get install dhcp3
Reading package lists... Done
Building dependency tree
Reading state information... Done
E: Unable to locate package dhcp3
14:59:36
#apt-get install dhcp3-server
Reading package lists... Done
Building dependency tree
Reading state information... Done
dhcp3-server is already the newest version.
0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded.
14:59:54
#apt-get install dhcp3-server
Reading package lists... Done
Building dependency tree
Reading state information... Done
dhcp3-server is already the newest version.
0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded.
15:00:17
#vim /etc/dhcp/dhcp.conf
15:02:00
#~
#authoritative;
# Use this to send dhcp log messages to a different log file (you also
# have to hack syslog.conf to complete the redirection).
"/etc/dhcp/dhcpd.conf" 107L, 3491C
log-facility local7;
# No service will be given on this subnet, but declaring it helps the
# DHCP server to understand the network topology.
#subnet 10.152.187.0 netmask 255.255.255.0 {
#}
# This is a very basic subnet declaration.
...
# allocated dynamically (if possible), but the host-specific information
# will still come from the host declarati{n.
#host passacaglia {
#  hardware ethernet 0:0:c0:5d:bd:95;
#} filename "vmuni{.passacaglia";
                  {
#  server-name "toccata.fugue.com";
#}
 }
"/etc/dhcp/dhcpd.conf" 107L, 3460C written
15:06:18
#less /etc/dhcp/dhcpd.conf
15:08:32
#/etc/init.d/isc-dhcp-server
.aptitude/         .bash_history      .bashrc            l3prompt           .lilalo/           .ssh/
.asterisk_history  .bash_profile      l3bashrc           .l3rc              .profile           .viminfo
15:08:32
#/etc/init.d/isc-dhcp-server restart
Stopping ISC DHCP server: dhcpd failed!
Starting ISC DHCP server: dhcpd.
15:13:06
#tail -f /var/lib/d
dbus/                defoma/              dhcp/                dictionaries-common/ dpkg/
15:13:06
#tail -f /var/lib/d
dbus/                defoma/              dhcp/                dictionaries-common/ dpkg/
15:13:06
#tail -f /var/lib/dhcp/dhcpd.leases
lease 192.168.50.200 {
  starts 1 2011/10/17 13:12:27;
  ends 1 2011/10/17 13:22:27;
  cltt 1 2011/10/17 13:12:27;
  binding state active;
  next binding state free;
  hardware ethernet c8:0a:a9:dc:73:aa;
  uid "\001\310\012\251\334s\252";
  client-hostname "Tech-notebook";
}
^[[A^[[A:
:q
^C
15:14:00
#ping 192.168.50.1
PING 192.168.50.1 (192.168.50.1) 56(84) bytes of data.
64 bytes from 192.168.50.1: icmp_req=1 ttl=64 time=0.028 ms
64 bytes from 192.168.50.1: icmp_req=2 ttl=64 time=0.017 ms
^C
--- 192.168.50.1 ping statistics ---
2 packets transmitted, 2 received, 0% packet loss, time 999ms
rtt min/avg/max/mdev = 0.017/0.022/0.028/0.007 ms
прошло 14 минут
15:28:57
#ping 192.168.50.200
PING 192.168.50.200 (192.168.50.200) 56(84) bytes of data.
64 bytes from 192.168.50.200: icmp_req=1 ttl=128 time=0.856 ms
64 bytes from 192.168.50.200: icmp_req=2 ttl=128 time=0.484 ms
^C
--- 192.168.50.200 ping statistics ---
2 packets transmitted, 2 received, 0% packet loss, time 999ms
rtt min/avg/max/mdev = 0.484/0.670/0.856/0.186 ms
15:29:01
#ping 192.168.50.201
PING 192.168.50.201 (192.168.50.201) 56(84) bytes of data.
64 bytes from 192.168.50.201: icmp_req=2 ttl=64 time=0.955 ms
64 bytes from 192.168.50.201: icmp_req=3 ttl=64 time=0.895 ms
64 bytes from 192.168.50.201: icmp_req=4 ttl=64 time=0.934 ms
64 bytes from 192.168.50.201: icmp_req=5 ttl=64 time=0.946 ms
^C
--- 192.168.50.201 ping statistics ---
5 packets transmitted, 4 received, 20% packet loss, time 4010ms
rtt min/avg/max/mdev = 0.895/0.932/0.955/0.038 ms
15:29:07
#apt-get install nc
Reading package lists... Done
Building dependency tree
Reading state information... Done
E: Unable to locate package nc
15:31:55
#apt-cache search nc
4g8 - Packet Capture and Interception for Switched Networks
a2ps - GNU a2ps - 'Anything to PostScript' converter and pretty-printer
liba52-0.7.4-dev - library for decoding ATSC A/52 streams (development)
liba52-0.7.4 - library for decoding ATSC A/52 streams
a7xpg-data - chase action game - game data
a7xpg - chase action game
aa3d - ASCII art stereogram generator
aaphoto - Auto Adjust Photo, automatic color correction of photos
abcde - A Better CD Encoder
abe-data - Side-scrolling game named "Abe's Amazing Adventure"
...
samba - SMB/CIFS file, print, and login server for Unix
smbclient - command-line SMB/CIFS clients for Unix
swat - Samba Web Administration Tool
winbind - Samba nameservice integration server
libsvn-java - Java bindings for Subversion
libsvn1 - Shared libraries used by Subversion
subversion-tools - Assorted tools related to Subversion
subversion - Advanced version control system
mahara-mediaplayer - Electronic portfolio, weblog, and resume builder - internal media player
busybox - Tiny utilities for small and embedded systems
15:32:20
#apt-cache search mc
libace-rmcast-5.7.7 - ACE reliable multicast library
libace-rmcast-dev - ACE reliable multicast library development files
libace-tmcast-5.7.7 - ACE transactional multicast library
libace-tmcast-dev - ACE transactional multicast library development files
alsa-oss - ALSA wrapper for OSS applications
amule-emc - lists ed2k links inside emulecollection files
ap-utils - Access Point SNMP Utils for Linux
apel - portable library for emacsen
apmd - Utilities for Advanced Power Management (APM)
ardour - digital audio workstation (graphical gtk2 interface)
...
xfce4-settings - graphical application for managing Xfce settings
xli - command line tool for viewing images in X11
xloadimage - Graphics file viewer under X11
xmcd - X11 based CD player
xmpuzzles - collection of puzzles for X (Motif version)
xserver-xorg-video-intel - X.Org X server -- Intel i8xx, i9xx display driver
xserver-xorg-video-openchrome - X.Org X server -- VIA display driver
xsmc-calc - Smith Chart calculator for X
xtermcontrol - dynamic configuration of xterm properties
yorick-yao - a Yorick-based adaptive optics system simulator
15:32:27
#apt-get install mc
Reading package lists... Done
Building dependency tree
Reading state information... Done
Suggested packages:
  zip unzip arj dbview odt2txt gv catdvi djvulibre-bin python-boto python-tz
The following NEW packages will be installed:
  mc
0 upgraded, 1 newly installed, 0 to remove and 0 not upgraded.
Need to get 2,173 kB of archives.
After this operation, 6,603 kB of additional disk space will be used.
Get:1 http://10.0.35.1/debian/ squeeze/main mc i386 3:4.7.0.9-1 [2,173 kB]
Fetched 2,173 kB in 0s (9,666 kB/s)
Selecting previously deselected package mc.
(Reading database ... 115144 files and directories currently installed.)
Unpacking mc (from .../mc_3%3a4.7.0.9-1_i386.deb) ...
Processing triggers for man-db ...
Processing triggers for menu ...
Setting up mc (3:4.7.0.9-1) ...
Processing triggers for menu ...
15:32:38
#cd /etc/asterisk/

прошло 25 минут
15:57:51
#cat sip
sip.conf         sip_notify.conf
15:57:51
#cat sip.conf
;
; SIP Configuration example for Asterisk
;
; SIP dial strings
;-----------------------------------------------------------
; In the dialplan (extensions.conf) you can use several
; syntaxes for dialing SIP devices.
;        SIP/devicename
;        SIP/username@domain   (SIP uri)
;        SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
...
;[pre14-asterisk]
;type=friend
;secret=digium
;host=dynamic
;rfc2833compensate=yes          ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
                                ; You must have this turned on or DTMF reception will work improperly.
;t38pt_usertpsource=yes         ; Use the source IP address of RTP as the destination IP address for UDPTL packets
                                ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
                                ; external IP address of the remote device. If port forwarding is done at the client side
                                ; then UDPTL will flow to the remote device.
15:58:06
#cd /etc/asterisk/

16:05:01
#ls -la
total 680
drwxr-xr-x   3 asterisk asterisk  4096 Oct 17 14:23 .
drwxr-xr-x 123 root     root     12288 Oct 17 16:32 ..
-rw-r-----   1 asterisk asterisk   140 Jul  7 11:58 adsi.conf
-rw-r-----   1 asterisk asterisk   840 Jul  7 11:58 adtranvofr.conf
-rw-r-----   1 asterisk asterisk  3035 Jul  7 11:58 agents.conf
-rw-r-----   1 asterisk asterisk  2906 Jul  7 11:58 ais.conf
-rw-r-----   1 asterisk asterisk  2227 Jul  7 11:58 alarmreceiver.conf
-rw-r-----   1 asterisk asterisk  3259 Jul  7 11:58 alsa.conf
-rw-r-----   1 asterisk asterisk   767 Jul  7 11:58 amd.conf
...
-rw-r-----   1 asterisk asterisk  9135 Jul  7 11:58 skinny.conf
-rw-r-----   1 asterisk asterisk  6717 Jul  7 11:58 sla.conf
-rw-r-----   1 asterisk asterisk  2669 Jul  7 11:58 smdi.conf
-rw-r-----   1 asterisk asterisk  1384 Jul  7 11:58 telcordia-1.adsi
-rw-r-----   1 asterisk asterisk   656 Jul  7 11:58 udptl.conf
-rw-r-----   1 asterisk asterisk  4909 Jul  7 11:58 unistim.conf
-rw-r-----   1 asterisk asterisk  3216 Jul  7 11:58 usbradio.conf
-rw-r-----   1 asterisk asterisk  2011 Jul  7 11:58 users.conf
-rw-r-----   1 asterisk asterisk 17961 Jul  7 11:58 voicemail.conf
-rw-r-----   1 asterisk asterisk  5939 Jul  7 11:58 vpb.conf
16:05:04
#ls -l
total 664
-rw-r----- 1 asterisk asterisk   140 Jul  7 11:58 adsi.conf
-rw-r----- 1 asterisk asterisk   840 Jul  7 11:58 adtranvofr.conf
-rw-r----- 1 asterisk asterisk  3035 Jul  7 11:58 agents.conf
-rw-r----- 1 asterisk asterisk  2906 Jul  7 11:58 ais.conf
-rw-r----- 1 asterisk asterisk  2227 Jul  7 11:58 alarmreceiver.conf
-rw-r----- 1 asterisk asterisk  3259 Jul  7 11:58 alsa.conf
-rw-r----- 1 asterisk asterisk   767 Jul  7 11:58 amd.conf
-rw-r----- 1 asterisk asterisk  3260 Jul  7 11:58 asterisk.adsi
-rw-r----- 1 asterisk asterisk  3234 Jul  7 11:58 asterisk.conf
...
-rw-r----- 1 asterisk asterisk  9135 Jul  7 11:58 skinny.conf
-rw-r----- 1 asterisk asterisk  6717 Jul  7 11:58 sla.conf
-rw-r----- 1 asterisk asterisk  2669 Jul  7 11:58 smdi.conf
-rw-r----- 1 asterisk asterisk  1384 Jul  7 11:58 telcordia-1.adsi
-rw-r----- 1 asterisk asterisk   656 Jul  7 11:58 udptl.conf
-rw-r----- 1 asterisk asterisk  4909 Jul  7 11:58 unistim.conf
-rw-r----- 1 asterisk asterisk  3216 Jul  7 11:58 usbradio.conf
-rw-r----- 1 asterisk asterisk  2011 Jul  7 11:58 users.conf
-rw-r----- 1 asterisk asterisk 17961 Jul  7 11:58 voicemail.conf
-rw-r----- 1 asterisk asterisk  5939 Jul  7 11:58 vpb.conf
16:05:10
#ls
adsi.conf                chan_dahdi.conf         festival.conf     misdn.conf              say.conf
adtranvofr.conf          cli_aliases.conf        followme.conf     modules.conf            sip.conf
agents.conf              cli.conf                func_odbc.conf    musiconhold.conf        sip_notify.conf
ais.conf                 cli_permissions.conf    gtalk.conf        muted.conf              skinny.conf
alarmreceiver.conf       codecs.conf             h323.conf         osp.conf                sla.conf
alsa.conf                console.conf            http.conf         oss.conf                smdi.conf
amd.conf                 dbsep.conf              iax.conf          phone.conf              telcordia-1.adsi
asterisk.adsi            dnsmgr.conf             iaxprov.conf      phoneprov.conf          udptl.conf
asterisk.conf            dsp.conf                indications.conf  queuerules.conf         unistim.conf
cdr_adaptive_odbc.conf   dundi.conf              jabber.conf       queues.conf             usbradio.conf
cdr.conf                 enum.conf               jingle.conf       res_config_sqlite.conf  users.conf
cdr_custom.conf          extconfig.conf          logger.conf       res_ldap.conf           voicemail.conf
cdr_manager.conf         extensions.ael          manager.conf      res_odbc.conf           vpb.conf
cdr_odbc.conf            extensions.conf         manager.d         res_pgsql.conf
cdr_pgsql.conf           extensions.lua          meetme.conf       res_snmp.conf
cdr_sqlite3_custom.conf  extensions_minivm.conf  mgcp.conf         rpt.conf
cdr_tds.conf             features.conf           minivm.conf       rtp.conf
16:05:12
#cp sip.conf sip.conf.SAVED

16:05:37
#vim sip.conf
прошло 10 минут
16:16:09
#cat /etc/asterisk/sip.conf | sed 's/;.*//' | expand | grep -xv ' *' | head -7 less
head: cannot open `less' for reading: No such file or directory
16:16:23
#cat /etc/asterisk/sip.conf | sed 's/;.*//' | expand | grep -xv ' *' | head -7 | less
16:17:45
#cd /etc/asterisk/cat sipconf
bash: cd: /etc/asterisk/cat: No such file or directory
16:17:55
#cat sip.conf

16:18:09
#cp sip.conf.SAVED sip.conf

16:18:46
#cat sip.conf
;
; SIP Configuration example for Asterisk
;
; SIP dial strings
;-----------------------------------------------------------
; In the dialplan (extensions.conf) you can use several
; syntaxes for dialing SIP devices.
;        SIP/devicename
;        SIP/username@domain   (SIP uri)
;        SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
...
;[pre14-asterisk]
;type=friend
;secret=digium
;host=dynamic
;rfc2833compensate=yes          ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
                                ; You must have this turned on or DTMF reception will work improperly.
;t38pt_usertpsource=yes         ; Use the source IP address of RTP as the destination IP address for UDPTL packets
                                ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
                                ; external IP address of the remote device. If port forwarding is done at the client side
                                ; then UDPTL will flow to the remote device.
16:19:04
#cat /etc/asterisk/sip.conf | sed 's/;.*//' | expand | grep -xv ' *' | head -7 | less
16:20:27
#less
16:26:45
#vim sip.conf
прошло 10 минут
16:37:06
#cat sip.conf
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
[2101]
type=friend     ;friend, user, peer
secret=1234
host=dynamic
[2102]
type=friend     ;friend, user, peer
secret=1234
host=dynamic
16:37:11
#~/asterisk -rvv
bash: /root/asterisk: No such file or directory
16:37:54
#asterisk -rvv
Asterisk 1.6.2.9-2+squeeze3, 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+squeeze3 currently running on linux5 (pid = 7811)
Verbosity is at least 2
[Oct 17 17:39:44] NOTICE[7830]: chan_sip.c:21763 handle_request_register: Registration from '"2102"<sip:2102@192.168.50.1>' failed for '192.168.50.200' - No matching peer found
linux5*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]
linux5*CLI> exit
Executing last minute cleanups
16:41:45
#asterisk -rvv
Asterisk 1.6.2.9-2+squeeze3, 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+squeeze3 currently running on linux5 (pid = 7811)
Verbosity is at least 2
linux5*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]
linux5*CLI> exit
Executing last minute cleanups
16:44:50
#cat sip.conf
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
[2101]
type=friend     ;friend, user, peer
secret=1234
host=dynamic
[2102]
type=friend     ;friend, user, peer
secret=1234
host=dynamic
16:44:54
#asterisk -rvv
Asterisk 1.6.2.9-2+squeeze3, 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+squeeze3 currently running on linux5 (pid = 7811)
Verbosity is at least 2
linux5*CLI> sip reload
 Reloading SIP
  == Parsing '/etc/asterisk/sip.conf':   == Found
  == Parsing '/etc/asterisk/users.conf':   == Found
  == Parsing '/etc/asterisk/sip_notify.conf':   == Found
linux5*CLI> exit
Executing last minute cleanups
16:46:15
#asterisk -rvvvv
Asterisk 1.6.2.9-2+squeeze3, 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+squeeze3 currently running on linux5 (pid = 7811)
Verbosity was 2 and is now 4
    -- Registered SIP '2102' at 192.168.50.200 port 34984
       > Saved useragent "X-Lite 4 release 4.1 stamp 63214" for peer 2102
[Oct 17 17:46:55] NOTICE[7830]: chan_sip.c:21594 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 2102
linux5*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
2101                       (Unspecified)    D          5060     Unmonitored
2102/2102                  192.168.50.200   D          34984    Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
linux5*CLI> exit
Executing last minute cleanups
16:48:35
#ls -la
total 684
drwxr-xr-x   3 asterisk asterisk  4096 Oct 17 17:37 .
drwxr-xr-x 123 root     root     12288 Oct 17 16:32 ..
-rw-r-----   1 asterisk asterisk   140 Jul  7 11:58 adsi.conf
-rw-r-----   1 asterisk asterisk   840 Jul  7 11:58 adtranvofr.conf
-rw-r-----   1 asterisk asterisk  3035 Jul  7 11:58 agents.conf
-rw-r-----   1 asterisk asterisk  2906 Jul  7 11:58 ais.conf
-rw-r-----   1 asterisk asterisk  2227 Jul  7 11:58 alarmreceiver.conf
-rw-r-----   1 asterisk asterisk  3259 Jul  7 11:58 alsa.conf
-rw-r-----   1 asterisk asterisk   767 Jul  7 11:58 amd.conf
...
-rw-r-----   1 asterisk asterisk  9135 Jul  7 11:58 skinny.conf
-rw-r-----   1 asterisk asterisk  6717 Jul  7 11:58 sla.conf
-rw-r-----   1 asterisk asterisk  2669 Jul  7 11:58 smdi.conf
-rw-r-----   1 asterisk asterisk  1384 Jul  7 11:58 telcordia-1.adsi
-rw-r-----   1 asterisk asterisk   656 Jul  7 11:58 udptl.conf
-rw-r-----   1 asterisk asterisk  4909 Jul  7 11:58 unistim.conf
-rw-r-----   1 asterisk asterisk  3216 Jul  7 11:58 usbradio.conf
-rw-r-----   1 asterisk asterisk  2011 Jul  7 11:58 users.conf
-rw-r-----   1 asterisk asterisk 17961 Jul  7 11:58 voicemail.conf
-rw-r-----   1 asterisk asterisk  5939 Jul  7 11:58 vpb.conf
16:48:39
#ls
adsi.conf                chan_dahdi.conf         festival.conf     misdn.conf              say.conf
adtranvofr.conf          cli_aliases.conf        followme.conf     modules.conf            sip.conf
agents.conf              cli.conf                func_odbc.conf    musiconhold.conf        sip.conf.SAVED
ais.conf                 cli_permissions.conf    gtalk.conf        muted.conf              sip_notify.conf
alarmreceiver.conf       codecs.conf             h323.conf         osp.conf                skinny.conf
alsa.conf                console.conf            http.conf         oss.conf                sla.conf
amd.conf                 dbsep.conf              iax.conf          phone.conf              smdi.conf
asterisk.adsi            dnsmgr.conf             iaxprov.conf      phoneprov.conf          telcordia-1.adsi
asterisk.conf            dsp.conf                indications.conf  queuerules.conf         udptl.conf
cdr_adaptive_odbc.conf   dundi.conf              jabber.conf       queues.conf             unistim.conf
cdr.conf                 enum.conf               jingle.conf       res_config_sqlite.conf  usbradio.conf
cdr_custom.conf          extconfig.conf          logger.conf       res_ldap.conf           users.conf
cdr_manager.conf         extensions.ael          manager.conf      res_odbc.conf           voicemail.conf
cdr_odbc.conf            extensions.conf         manager.d         res_pgsql.conf          vpb.conf
cdr_pgsql.conf           extensions.lua          meetme.conf       res_snmp.conf
cdr_sqlite3_custom.conf  extensions_minivm.conf  mgcp.conf         rpt.conf
cdr_tds.conf             features.conf           minivm.conf       rtp.conf
16:48:42
#cp ext
extconfig.conf          extensions.ael          extensions.conf         extensions.lua          extensions_minivm.conf
16:48:42
#cp extensions.conf extensions.conf.SAVED

16:49:17
#cat extensions.conf
; extensions.conf - the Asterisk dial plan
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "dialplan reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI
;
...
exten => _X.,n,SayDigits(${CALLERID(ani)})      ; playback again in case of missed digit
exten => _X.,n,Return()
; For more information on applications, just type "core show applications" at your
; friendly Asterisk CLI prompt.
;
; "core show application <command>" will show details of how you
; use that particular application in this file, the dial plan.
; "core show functions" will list all dialplan functions
; "core show function <COMMAND>" will show you more information about
; one function. Remember that function names are UPPER CASE.
16:49:38
#vim extensions.conf
16:54:03
#vim extensions.conf
16:57:17
#asterisk -rvv
Asterisk 1.6.2.9-2+squeeze3, 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+squeeze3 currently running on linux5 (pid = 7811)
...
    -- Executing [2199@default:1] Playback("SIP/2102-00000003", "demo-thanks") in new stack
    -- <SIP/2102-00000003> Playing 'demo-thanks.gsm' (language 'en')
[Oct 17 18:10:36] NOTICE[10680]: rtp.c:1808 ast_rtp_read: Unknown RTP codec 126 received from '192.168.50.200'
[Oct 17 18:10:36] NOTICE[10680]: rtp.c:1808 ast_rtp_read: Unknown RTP codec 126 received from '192.168.50.200'
[Oct 17 18:10:36] NOTICE[10680]: rtp.c:1808 ast_rtp_read: Unknown RTP codec 126 received from '192.168.50.200'
    -- Executing [2199@default:2] Playback("SIP/2102-00000003", "demo-thanks") in new stack
    -- <SIP/2102-00000003> Playing 'demo-thanks.gsm' (language 'en')
    -- Auto fallthrough, channel 'SIP/2102-00000003' status is 'UNKNOWN'
linux5*CLI> exit
Executing last minute cleanups
прошло 13 минут
17:10:54
#vim extensions.conf
17:12:44
#asterisk -rvv\
> asterisk -rvv
asterisk: invalid option -- 'a'
17:12:55
#asterisk -rvvasterisk -rvv
asterisk: invalid option -- 'a'
17:12:57
#asterisk -rvvasterisk -rvvvvv
asterisk: invalid option -- 'a'
17:13:02
#asterisk -rvv
Asterisk 1.6.2.9-2+squeeze3, 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+squeeze3 currently running on linux5 (pid = 7811)
...
    -- merging incls/swits/igpats from old(app_dial_gosub_virtual_context) to new(app_dial_gosub_virtual_context) context, registrar = pbx_config
    -- Added extension 's' priority 1 to app_dial_gosub_virtual_context (0xb4d2e0e8)
    -- Time to scan old dialplan and merge leftovers back into the new: 0.000611 sec
    -- Time to restore hints and swap in new dialplan: 0.000000 sec
    -- Time to delete the old dialplan: 0.000035 sec
    -- Total time merge_contexts_delete: 0.000646 sec
  == Using SIP RTP CoS mark 5
[Oct 17 18:13:37] NOTICE[7830]: chan_sip.c:20276 handle_request_invite: Call from '2102' to extension '2101' rejected because extension not found in context 'default'.
linux5*CLI> exit
Executing last minute cleanups
17:14:02
#vim sip.conf
17:14:35
#[2102]
Asterisk 1.6.2.9-2+squeeze3, 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+squeeze3 currently running on linux5 (pid = 7811)
...
  == Using SIP RTP CoS mark 5
[Oct 17 18:17:55] NOTICE[7830]: chan_sip.c:20276 handle_request_invite: Call from '2102' to extension '2101' rejected because extension not found in context 'default'.
  == Using SIP RTP CoS mark 5
[Oct 17 18:18:22] NOTICE[7830]: chan_sip.c:20276 handle_request_invite: Call from '2102' to extension '2101' rejected because extension not found in context 'default'.
[Oct 17 18:18:46] NOTICE[7830]: chan_sip.c:21763 handle_request_register: Registration from 'sip:2201@192.168.50.1' failed for '192.168.50.201' - No matching peer found
[Oct 17 18:19:47] NOTICE[7830]: chan_sip.c:21763 handle_request_register: Registration from 'sip:2201@192.168.50.1' failed for '192.168.50.201' - No matching peer found
[Oct 17 18:20:47] NOTICE[7830]: chan_sip.c:21763 handle_request_register: Registration from 'sip:2201@192.168.50.1' failed for '192.168.50.201' - No matching peer found
[Oct 17 18:21:47] NOTICE[7830]: chan_sip.c:21763 handle_request_register: Registration from 'sip:2201@192.168.50.1' failed for '192.168.50.201' - No matching peer found
linux5*CLI> exit
Executing last minute cleanups
17:21:54
#passwordpassword
bash: passwordpassword: command not found

Вторник (10/18/11)

08:41:10
#~
Asterisk 1.6.2.9-2+squeeze3, 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+squeeze3 currently running on linux5 (pid = 7811)
...
       > Saved useragent "Cisco-CP7960G/7.5" for peer 2101
linux5*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
2101/2101                  192.168.50.201   D          5060     Unmonitored
2102/2102                  192.168.50.200   D          52600    Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
  == Using SIP RTP CoS mark 5
[Oct 18 09:45:52] NOTICE[7830]: chan_sip.c:20276 handle_request_invite: Call from '2102' to extension '2101' rejected because extension not found in context 'default'.
linux5*CLI> exit
Executing last minute cleanups
08:46:02
#vim extensions.conf
08:46:24
#asterisk -rvv
Asterisk 1.6.2.9-2+squeeze3, 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+squeeze3 currently running on linux5 (pid = 7811)
...
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
    -- Remote UNIX connection
Executing last minute cleanups
  == Destroying musiconhold processes
linux5*CLI>
Disconnected from Asterisk server
Executing last minute cleanups

Файлы

  • extensions.conf
  • sip
  • sip.conf
  • extensions.conf
    >
    ; extensions.conf - the Asterisk dial plan
    ;
    ; Static extension configuration file, used by
    ; the pbx_config module. This is where you configure all your
    ; inbound and outbound calls in Asterisk.
    ;
    ; This configuration file is reloaded
    ; - With the "dialplan reload" command in the CLI
    ; - With the "reload" command (that reloads everything) in the CLI
    ;
    ; The "General" category is for certain variables.
    ;
    [general]
    ;
    ; If static is set to no, or omitted, then the pbx_config will rewrite
    ; this file when extensions are modified.  Remember that all comments
    ; made in the file will be lost when that happens.
    ;
    ; XXX Not yet implemented XXX
    ;
    static=yes
    ;
    ; if static=yes and writeprotect=no, you can save dialplan by
    ; CLI command "dialplan save" too
    ;
    writeprotect=no
    ;
    ; If autofallthrough is set, then if an extension runs out of
    ; things to do, it will terminate the call with BUSY, CONGESTION
    ; or HANGUP depending on Asterisk's best guess. This is the default.
    ;
    ; If autofallthrough is not set, then if an extension runs out of
    ; things to do, Asterisk will wait for a new extension to be dialed
    ; (this is the original behavior of Asterisk 1.0 and earlier).
    ;
    ;autofallthrough=no
    ;
    ;
    ;
    ; If extenpatternmatchnew is set (true, yes, etc), then a new algorithm that uses
    ; a Trie to find the best matching pattern is used. In dialplans
    ; with more than about 20-40 extensions in a single context, this
    ; new algorithm can provide a noticeable speedup.
    ; With 50 extensions, the speedup is 1.32x
    ; with 88 extensions, the speedup is 2.23x
    ; with 138 extensions, the speedup is 3.44x
    ; with 238 extensions, the speedup is 5.8x
    ; with 438 extensions, the speedup is 10.4x
    ; With 1000 extensions, the speedup is ~25x
    ; with 10,000 extensions, the speedup is 374x
    ; Basically, the new algorithm provides a flat response
    ; time, no matter the number of extensions.
    ;
    ; By default, the old pattern matcher is used.
    ;
    ; ****This is a new feature! *********************
    ; The new pattern matcher is for the brave, the bold, and
    ; the desperate. If you have large dialplans (more than about 50 extensions
    ; in a context), and/or high call volume, you might consider setting
    ; this value to "yes" !!
    ; Please, if you try this out, and are forced to return to the
    ; old pattern matcher, please report your reasons in a bug report
    ; on bugs.digium.com. We have made good progress in providing something
    ; compatible with the old matcher; help us finish the job!
    ;
    ; This value can be switched at runtime using the cli command "dialplan set extenpatternmatchnew true"
    ; or "dialplan set extenpatternmatchnew false", so you can experiment to your hearts content.
    ;
    ;extenpatternmatchnew=no
    ;
    ; If clearglobalvars is set, global variables will be cleared
    ; and reparsed on a dialplan reload, or Asterisk reload.
    ;
    ; If clearglobalvars is not set, then global variables will persist
    ; through reloads, and even if deleted from the extensions.conf or
    ; one of its included files, will remain set to the previous value.
    ;
    ; NOTE: A complication sets in, if you put your global variables into
    ; the AEL file, instead of the extensions.conf file. With clearglobalvars
    ; set, a "reload" will often leave the globals vars cleared, because it
    ; is not unusual to have extensions.conf (which will have no globals)
    ; load after the extensions.ael file (where the global vars are stored).
    ; So, with "reload" in this particular situation, first the AEL file will
    ; clear and then set all the global vars, then, later, when the extensions.conf
    ; file is loaded, the global vars are all cleared, and then not set, because
    ; they are not stored in the extensions.conf file.
    ;
    clearglobalvars=no
    ;
    ; If priorityjumping is set to 'yes', then applications that support
    ; 'jumping' to a different priority based on the result of their operations
    ; will do so (this is backwards compatible behavior with pre-1.2 releases
    ; of Asterisk). Individual applications can also be requested to do this
    ; by passing a 'j' option in their arguments.
    ;
    ;priorityjumping=yes
    ;
    ; User context is where entries from users.conf are registered.  The
    ; default value is 'default'
    ;
    ;userscontext=default
    ;
    ; You can include other config files, use the #include command
    ; (without the ';'). Note that this is different from the "include" command
    ; that includes contexts within other contexts. The #include command works
    ; in all asterisk configuration files.
    ;#include "filename.conf"
    ;#include <filename.conf>
    ;#include filename.conf
    ;
    ; You can execute a program or script that produces config files, and they
    ; will be inserted where you insert the #exec command. The #exec command
    ; works on all asterisk configuration files.  However, you will need to
    ; activate them within asterisk.conf with the "execincludes" option.  They
    ; are otherwise considered a security risk.
    ;#exec /opt/bin/build-extra-contexts.sh
    ;#exec /opt/bin/build-extra-contexts.sh --foo="bar"
    ;#exec </opt/bin/build-extra-contexts.sh --foo="bar">
    ;#exec "/opt/bin/build-extra-contexts.sh --foo=\"bar\""
    ;
    ; The "Globals" category contains global variables that can be referenced
    ; in the dialplan with the GLOBAL dialplan function:
    ; ${GLOBAL(VARIABLE)}
    ; ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
    ; Unix/Linux environmental variables can be reached with the ENV dialplan
    ; function: ${ENV(VARIABLE)}
    ;
    [globals]
    CONSOLE=Console/dsp                             ; Console interface for demo
    ;CONSOLE=DAHDI/1
    ;CONSOLE=Phone/phone0
    IAXINFO=guest                                   ; IAXtel username/password
    ;IAXINFO=myuser:mypass
    TRUNK=DAHDI/G2                                  ; Trunk interface
    ;
    ; Note the 'G2' in the TRUNK variable above. It specifies which group (defined
    ; in chan_dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use
    ; in the specified group. The four possible options are:
    ;
    ; g: select the lowest-numbered non-busy DAHDI channel
    ;    (aka. ascending sequential hunt group).
    ; G: select the highest-numbered non-busy DAHDI channel
    ;    (aka. descending sequential hunt group).
    ; r: use a round-robin search, starting at the next highest channel than last
    ;    time (aka. ascending rotary hunt group).
    ; R: use a round-robin search, starting at the next lowest channel than last
    ;    time (aka. descending rotary hunt group).
    ;
    TRUNKMSD=1                                      ; MSD digits to strip (usually 1 or 0)
    ;TRUNK=IAX2/user:pass@provider
    ;FREENUMDOMAIN=mydomain.com                     ; domain to send on outbound
                                                    ; freenum calls (uses outbound-freenum
                                                    ; context)
    ;
    ; WARNING WARNING WARNING WARNING
    ; If you load any other extension configuration engine, such as pbx_ael.so,
    ; your global variables may be overridden by that file.  Please take care to
    ; use only one location to set global variables, and you will likely save
    ; yourself a ton of grief.
    ; WARNING WARNING WARNING WARNING
    ;
    ; Any category other than "General" and "Globals" represent
    ; extension contexts, which are collections of extensions.
    ;
    ; Extension names may be numbers, letters, or combinations
    ; thereof. If an extension name is prefixed by a '_'
    ; character, it is interpreted as a pattern rather than a
    ; literal.  In patterns, some characters have special meanings:
    ;
    ;   X - any digit from 0-9
    ;   Z - any digit from 1-9
    ;   N - any digit from 2-9
    ;   [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
    ;   . - wildcard, matches anything remaining (e.g. _9011. matches
    ;       anything starting with 9011 excluding 9011 itself)
    ;   ! - wildcard, causes the matching process to complete as soon as
    ;       it can unambiguously determine that no other matches are possible
    ;
    ; For example, the extension _NXXXXXX would match normal 7 digit dialings,
    ; while _1NXXNXXXXXX would represent an area code plus phone number
    ; preceded by a one.
    ;
    ; Each step of an extension is ordered by priority, which must always start
    ; with 1 to be considered a valid extension.  The priority "next" or "n" means
    ; the previous priority plus one, regardless of whether the previous priority
    ; was associated with the current extension or not.  The priority "same" or "s"
    ; means the same as the previously specified priority, again regardless of
    ; whether the previous entry was for the same extension.  Priorities may be
    ; immediately followed by a plus sign and another integer to add that amount
    ; (most useful with 's' or 'n').  Priorities may then also have an alias, or
    ; label, in parentheses after their name which can be used in goto situations.
    ;
    ; Contexts contain several lines, one for each step of each extension.  One may
    ; include another context in the current one as well, optionally with a date
    ; and time.  Included contexts are included in the order they are listed.
    ; Switches may also be included within a context.  The order of matching within
    ; a context is always exact extensions, pattern match extensions, includes, and
    ; switches.  Includes are always processed depth-first.  So for example, if you
    ; would like a switch "A" to match before context "B", simply put switch "A" in
    ; an included context "C", where "C" is included in your original context
    ; before "B".
    ;
    ;[context]
    ;exten => someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,...)
    ;
    ; Timing list for includes is
    ;
    ;   <time range>,<days of week>,<days of month>,<months>[,<timezone>]
    ;
    ; Note that ranges may be specified to wrap around the ends.  Also, minutes are
    ; fine-grained only down to the closest even minute.
    ;
    ;include => daytime,9:00-17:00,mon-fri,*,*
    ;include => weekend,*,sat-sun,*,*
    ;include => weeknights,17:02-8:58,mon-fri,*,*
    ;
    ; ignorepat can be used to instruct drivers to not cancel dialtone upon receipt
    ; of a particular pattern.  The most commonly used example is of course '9'
    ; like this:
    ;
    ;ignorepat => 9
    ;
    ; so that dialtone remains even after dialing a 9.  Please note that ignorepat
    ; only works with channels which receive dialtone from the PBX, such as DAHDI,
    ; Phone, and VPB.  Other channels, such as SIP and MGCP, which generate their
    ; own dialtone and converse with the PBX only after a number is complete, are
    ; generally unaffected by ignorepat (unless DISA or another method is used to
    ; generate a dialtone after answering the channel).
    ;
    ;
    ; Sample entries for extensions.conf
    ;
    ;
    [dundi-e164-canonical]
    ;include => stdexten
    ;
    ; List canonical entries here
    ;
    ;exten => 12564286000,1,Gosub(6000,stdexten(IAX2/foo))
    ;exten => 12564286000,n,Goto(default,s,1)       ; exited Voicemail
    ;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})
    [dundi-e164-customers]
    ;
    ; If you are an ITSP or Reseller, list your customers here.
    ;
    ;exten => _12564286000,1,Dial(SIP/customer1)
    ;exten => _12564286001,1,Dial(IAX2/customer2)
    [dundi-e164-via-pstn]
    ;
    ; If you are freely delivering calls to the PSTN, list them here
    ;
    ;exten => _1256428XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428
    ;exten => _1256325XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Ditto for 256-325
    [dundi-e164-local]
    ;
    ; Context to put your dundi IAX2 or SIP user in for
    ; full access
    ;
    include => dundi-e164-canonical
    include => dundi-e164-customers
    include => dundi-e164-via-pstn
    [dundi-e164-switch]
    ;
    ; Just a wrapper for the switch
    ;
    switch => DUNDi/e164
    [dundi-e164-lookup]
    ;
    ; Locally to lookup, try looking for a local E.164 solution
    ; then try DUNDi if we don't have one.
    ;
    include => dundi-e164-local
    include => dundi-e164-switch
    ;
    ; DUNDi can also be implemented as a Macro instead of using
    ; the Local channel driver.
    ;
    [macro-dundi-e164]
    ;
    ; ARG1 is the extension to Dial
    ;
    ; Extension "s" is not a wildcard extension that matches "anything".
    ; In macros, it is the start extension. In most other cases,
    ; you have to goto "s" to execute that extension.
    ;
    ; For wildcard matches, see above - all pattern matches start with
    ; an underscore.
    exten => s,1,Goto(${ARG1},1)
    include => dundi-e164-lookup
    ;
    ; Here are the entries you need to participate in the IAXTEL
    ; call routing system.  Most IAXTEL numbers begin with 1-700, but
    ; there are exceptions.  For more information, and to sign
    ; up, please go to www.gnophone.com or www.iaxtel.com
    ;
    [iaxtel700]
    exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)
    ;
    ; The SWITCH statement permits a server to share the dialplan with
    ; another server. Use with care: Reciprocal switch statements are not
    ; allowed (e.g. both A -> B and B -> A), and the switched server needs
    ; to be on-line or else dialing can be severly delayed.
    ;
    [iaxprovider]
    ;switch => IAX2/user:[key]@myserver/mycontext
    [trunkint]
    ;
    ; International long distance through trunk
    ;
    exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
    exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${FILTER(0-9,${EXTEN:${GLOBAL(TRUNKMSD)}})})
    [trunkld]
    ;
    ; Long distance context accessed through trunk
    ;
    exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
    exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
    [trunklocal]
    ;
    ; Local seven-digit dialing accessed through trunk interface
    ;
    exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
    [trunktollfree]
    ;
    ; Long distance context accessed through trunk interface
    ;
    exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
    exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
    exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
    exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
    [international]
    ;
    ; Master context for international long distance
    ;
    ignorepat => 9
    include => longdistance
    include => trunkint
    [longdistance]
    ;
    ; Master context for long distance
    ;
    ignorepat => 9
    include => local
    include => trunkld
    [local]
    ;
    ; Master context for local, toll-free, and iaxtel calls only
    ;
    ignorepat => 9
    include => default
    include => trunklocal
    include => iaxtel700
    include => trunktollfree
    include => iaxprovider
    ;Include parkedcalls (or the context you define in features conf)
    ;to enable call parking.
    include => parkedcalls
    ;
    ; You can use an alternative switch type as well, to resolve
    ; extensions that are not known here, for example with remote
    ; IAX switching you transparently get access to the remote
    ; Asterisk PBX
    ;
    ; switch => IAX2/user:password@bigserver/local
    ;
    ; An "lswitch" is like a switch but is literal, in that
    ; variable substitution is not performed at load time
    ; but is passed to the switch directly (presumably to
    ; be substituted in the switch routine itself)
    ;
    ; lswitch => Loopback/12${EXTEN}@othercontext
    ;
    ; An "eswitch" is like a switch but the evaluation of
    ; variable substitution is performed at runtime before
    ; being passed to the switch routine.
    ;
    ; eswitch => IAX2/context@${CURSERVER}
    ; The following two contexts are a template to enable the ability to dial
    ; ISN numbers. For more information about what an ISN number is, please see
    ; http://www.freenum.org.
    ;
    ; This is the dialing hook.  use:
    ; include => outbound-freenum
    [outbound-freenum]
    ; We'll add more digits as needed. The purpose is to dial things
    ; like extension numbers at domains (ITAD number) so we're matching
    ; on lengths of 1 through 6 prior to the separator (the asterisk [*])
    ;
    exten => _X*X!,1,Goto(outbound-freenum2,${EXTEN},1)
    exten => _XX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
    exten => _XXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
    exten => _XXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
    exten => _XXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
    exten => _XXXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
    [outbound-freenum2]
    ; This is the handler which performs the dialing logic. It is called
    ; from the [outbound-freenum] context
    ;
    exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
    same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)})                                ; make sure the suffix is all digits as well
    same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION,1)
                                                                            ; filter out bad characters per the README-SERIOUSLY.best-practices.txt document
    same => n,Set(TIMEOUT(absolute)=10800)
    same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)})     ; perform our lookup with freenum.org
    same => n,GotoIf($["${isnresult}" != ""]?from)
    same => n,Set(DIALSTATUS=CONGESTION)
    same => n,Goto(fn-CONGESTION,1)
    same => n(from),Set(SIPFROMUSER=${CALLERID(num)})
    same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial)               ; check if we set the FREENUMDOMAIN global variable in [global]
    same => n,Set(SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})                   ;    if we did set it, then we'll use it for our outbound dialing domain
    same => n(dial),Dial(SIP/${isnresult},40)
    same => n,Goto(fn-${DIALSTATUS},1)
    exten => fn-BUSY,1,Busy()
    exten => _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
    same => n,Congestion()
    [macro-trunkdial]
    ;
    ; Standard trunk dial macro (hangs up on a dialstatus that should
    ; terminate call)
    ;   ${ARG1} - What to dial
    ;
    exten => s,1,Dial(${ARG1})
    exten => s,n,Goto(s-${DIALSTATUS},1)
    exten => s-NOANSWER,1,Hangup
    exten => s-BUSY,1,Hangup
    exten => _s-.,1,NoOp
    [stdexten]
    ;
    ; Standard extension subroutine:
    ;   ${EXTEN} - Extension
    ;   ${ARG1} - Device(s) to ring
    ;   ${ARG2} - Optional context in Voicemail (if empty, then "default")
    ;
    ; Note that the current version will drop through to the next priority in the
    ; case of their pressing '#'.  This gives more flexibility in what do to next:
    ; you can prompt for a new extension, or drop the call, or send them to a
    ; general delivery mailbox, or...
    ;
    ; The use of the LOCAL() function is purely for convenience.  Any variable
    ; initially declared as LOCAL() will disappear when the innermost Gosub context
    ; in which it was declared returns.  Note also that you can declare a LOCAL()
    ; variable on top of an existing variable, and its value will revert to its
    ; previous value (before being declared as LOCAL()) upon Return.
    ;
    exten => _X.,50000(stdexten),NoOp(Start stdexten)
    exten => _X.,n,Set(LOCAL(ext)=${EXTEN})
    exten => _X.,n,Set(LOCAL(dev)=${ARG1})
    exten => _X.,n,Set(LOCAL(cntx)=${ARG2})
    exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
    exten => _X.,n,Dial(${dev},20)                  ; Ring the interface, 20 seconds maximum
    exten => _X.,n,Goto(stdexten-${DIALSTATUS},1)           ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
    exten => stdexten-NOANSWER,1,Voicemail(${mbx},u)        ; If unavailable, send to voicemail w/ unavail announce
    exten => stdexten-NOANSWER,n,NoOp(Finish stdexten NOANSWER)
    exten => stdexten-NOANSWER,n,Return()                   ; If they press #, return to start
    exten => stdexten-BUSY,1,Voicemail(${mbx},b)
                                                    ; If busy, send to voicemail w/ busy announce
    exten => stdexten-BUSY,n,NoOp(Finish stdexten BUSY)
    exten => stdexten-BUSY,n,Return()                       ; If they press #, return to start
    exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1)    ; Treat anything else as no answer
    exten => a,1,VoicemailMain(${mbx})              ; If they press *, send the user into VoicemailMain
    exten => a,n,Return()
    [stdPrivacyexten]
    ;
    ; Standard extension subroutine:
    ;   ${ARG1} - Extension
    ;   ${ARG2} - Device(s) to ring
    ;   ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 extension-priority)
    ;   ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 extension-priority)`
    ;   ${ARG5} - Context in voicemail (if empty, then "default")
    ;
    ; See above note in stdexten about priority handling on exit.
    ;
    exten => _X.,60000(stdPrivacyexten),NoOp(Start stdPrivacyexten)
    exten => _X.,n,Set(LOCAL(ext)=${ARG1})
    exten => _X.,n,Set(LOCAL(dev)=${ARG2})
    exten => _X.,n,Set(LOCAL(dontcntx)=${ARG3})
    exten => _X.,n,Set(LOCAL(tortcntx)=${ARG4})
    exten => _X.,n,Set(LOCAL(cntx)=${ARG5})
    exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
    exten => _X.,n,Dial(${dev},20,p)                        ; Ring the interface, 20 seconds maximum, call screening
                                                    ; option (or use P for databased call _X.creening)
    exten => _X.,n,Goto(stdexten-${DIALSTATUS},1)           ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
    exten => stdexten-NOANSWER,1,Voicemail(${mbx},u)        ; If unavailable, send to voicemail w/ unavail announce
    exten => stdexten-NOANSWER,n,NoOp(Finish stdPrivacyexten NOANSWER)
    exten => stdexten-NOANSWER,n,Return()                   ; If they press #, return to start
    exten => stdexten-BUSY,1,Voicemail(${mbx},b)            ; If busy, send to voicemail w/ busy announce
    exten => stdexten-BUSY,n,NoOp(Finish stdPrivacyexten BUSY)
    exten => stdexten-BUSY,n,Return()                       ; If they press #, return to start
    exten => stdexten-DONTCALL,1,Goto(${dontcntx},s,1)      ; Callee chose to send this call to a polite "Don't call again" script.
    exten => stdexten-TORTURE,1,Goto(${tortcntx},s,1)       ; Callee chose to send this call to a telemarketer torture script.
    exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1)    ; Treat anything else as no answer
    exten => a,1,VoicemailMain(${mbx})              ; If they press *, send the user into VoicemailMain
    exten => a,n,Return
    [macro-page];
    ;
    ; Paging macro:
    ;
    ;       Check to see if SIP device is in use and DO NOT PAGE if they are
    ;
    ;   ${ARG1} - Device to page
    exten => s,1,ChanIsAvail(${ARG1},s)                     ; s is for ANY call
    exten => s,n,GoToIf([${AVAILORIGCHAN} = ""]?fail:autoanswer)
    exten => s,n(autoanswer),Set(_ALERT_INFO="RA")                  ; This is for the PolyComs
    exten => s,n,SIPAddHeader(Call-Info: Answer-After=0)    ; This is for the Grandstream, Snoms, and Others
    exten => s,n,NoOp()                                     ; Add others here and Post on the Wiki!!!!
    exten => s,n,Dial(${ARG1})
    exten => s,n(fail),Hangup
    [demo]
    include => stdexten
    ;
    ; We start with what to do when a call first comes in.
    ;
    exten => s,1,Wait(1)                    ; Wait a second, just for fun
    exten => s,n,Answer                     ; Answer the line
    exten => s,n,Set(TIMEOUT(digit)=5)      ; Set Digit Timeout to 5 seconds
    exten => s,n,Set(TIMEOUT(response)=10)  ; Set Response Timeout to 10 seconds
    exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message
    exten => s,n(instruct),BackGround(demo-instruct)        ; Play some instructions
    exten => s,n,WaitExten                  ; Wait for an extension to be dialed.
    exten => 2,1,BackGround(demo-moreinfo)  ; Give some more information.
    exten => 2,n,Goto(s,instruct)
    exten => 3,1,Set(LANGUAGE()=fr)         ; Set language to french
    exten => 3,n,Goto(s,restart)            ; Start with the congratulations
    exten => 1000,1,Goto(default,s,1)
    ;
    ; We also create an example user, 1234, who is on the console and has
    ; voicemail, etc.
    ;
    exten => 1234,1,Playback(transfer,skip)         ; "Please hold while..."
                                            ; (but skip if channel is not up)
    exten => 1234,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}))
    exten => 1234,n,Goto(default,s,1)               ; exited Voicemail
    exten => 1235,1,Voicemail(1234,u)               ; Right to voicemail
    exten => 1236,1,Dial(Console/dsp)               ; Ring forever
    exten => 1236,n,Voicemail(1234,b)               ; Unless busy
    ;
    ; # for when they're done with the demo
    ;
    exten => #,1,Playback(demo-thanks)      ; "Thanks for trying the demo"
    exten => #,n,Hangup                     ; Hang them up.
    ;
    ; A timeout and "invalid extension rule"
    ;
    exten => t,1,Goto(#,1)                  ; If they take too long, give up
    exten => i,1,Playback(invalid)          ; "That's not valid, try again"
    ;
    ; Create an extension, 500, for dialing the
    ; Asterisk demo.
    ;
    exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
    exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default)        ; Call the Asterisk demo
    exten => 500,n,Playback(demo-nogo)      ; Couldn't connect to the demo site
    exten => 500,n,Goto(s,6)                ; Return to the start over message.
    ;
    ; Create an extension, 600, for evaluating echo latency.
    ;
    exten => 600,1,Playback(demo-echotest)  ; Let them know what's going on
    exten => 600,n,Echo                     ; Do the echo test
    exten => 600,n,Playback(demo-echodone)  ; Let them know it's over
    exten => 600,n,Goto(s,6)                ; Start over
    ;
    ;       You can use the Macro Page to intercom a individual user
    exten => 76245,1,Macro(page,SIP/Grandstream1)
    ; or if your peernames are the same as extensions
    exten => _7XXX,1,Macro(page,SIP/${EXTEN})
    ;
    ;
    ; System Wide Page at extension 7999
    ;
    exten => 7999,1,Set(TIMEOUT(absolute)=60)
    exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n,d)
    ; Give voicemail at extension 8500
    ;
    exten => 8500,1,VoicemailMain
    exten => 8500,n,Goto(s,6)
    ;
    ; Here's what a phone entry would look like (IXJ for example)
    ;
    ;exten => 1265,1,Dial(Phone/phone0,15)
    ;exten => 1265,n,Goto(s,5)
    ;
    ;       The page context calls up the page macro that sets variables needed for auto-answer
    ;       It is in is own context to make calling it from the Page() application as simple as
    ;       Local/{peername}@page
    ;
    [page]
    exten => _X.,1,Macro(page,SIP/${EXTEN})
    ;[mainmenu]
    ;
    ; Example "main menu" context with submenu
    ;
    ;exten => s,1,Answer
    ;exten => s,n,Background(thanks)                ; "Thanks for calling press 1 for sales, 2 for support, ..."
    ;exten => s,n,WaitExten
    ;exten => 1,1,Goto(submenu,s,1)
    ;exten => 2,1,Hangup
    ;include => default
    ;
    ;[submenu]
    ;exten => s,1,Ringing                                   ; Make them comfortable with 2 seconds of ringback
    ;exten => s,n,Wait,2
    ;exten => s,n,Background(submenuopts)   ; "Thanks for calling the sales department.  Press 1 for steve, 2 for..."
    ;exten => s,n,WaitExten
    ;exten => 1,1,Goto(default,steve,1)
    ;exten => 2,1,Goto(default,mark,2)
    [default]
    ;
    ; By default we include the demo.  In a production system, you
    ; probably don't want to have the demo there.
    ;
    include => demo
    ;
    ; An extension like the one below can be used for FWD, Nikotel, sipgate etc.
    ; Note that you must have a [sipprovider] section in sip.conf
    ;
    ;exten => _41X.,1,Dial(SIP/${FILTER(0-9,${EXTEN:2})}@sipprovider,,r)
    ; Real extensions would go here. Generally you want real extensions to be
    ; 4 or 5 digits long (although there is no such requirement) and start with a
    ; single digit that is fairly large (like 6 or 7) so that you have plenty of
    ; room to overlap extensions and menu options without conflict.  You can alias
    ; them with names, too, and use global variables
    ;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1(Joe Schmoe) ; Channel hints for presence
    ;exten => 6245,1,Dial(SIP/Grandstream1,20,rt)   ; permit transfer
    ;exten => 6245,n(dial),Dial(${HINT},20,rtT)     ; Use hint as listed
    ;exten => 6245,n,Voicemail(6245,u)              ; Voicemail (unavailable)
    ;exten => 6245,s+1,Hangup                       ; s+1, same as n
    ;exten => 6245,dial+101,Voicemail(6245,b)       ; Voicemail (busy)
    ;exten => 6361,1,Dial(IAX2/JaneDoe,,rm)         ; ring without time limit
    ;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14)
    ;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
    ;exten => 6391,1,Dial(JINGLE/asterisk@digium.com/mogorman@astjab.org) ;Dial via jingle using asterisk as the transport and calling mogorman.
    ;exten => 6394,1,Dial(Local/6275/n)             ; this will dial ${MARK}
    ;exten => 6275,1,Gosub(${EXTEN},stdexten(${MARK}))
                                                    ; assuming ${MARK} is something like DAHDI/2
    ;exten => 6275,n,Goto(default,s,1)              ; exited Voicemail
    ;exten => mark,1,Goto(6275,1)                   ; alias mark to 6275
    ;exten => 6536,1,Gosub(${EXTEN},stdexten(${WIL}))
                                                    ; Ditto for wil
    ;exten => 6536,n,Goto(default,s,1)              ; exited Voicemail
    ;exten => wil,1,Goto(6236,1)
    ;If you want to subscribe to the status of a parking space, this is
    ;how you do it. Subscribe to extension 6600 in sip, and you will see
    ;the status of the first parking lot with this extensions' help
    ;exten => 6600,hint,park:701@parkedcalls
    ;exten => 6600,1,noop
    ;
    ; Some other handy things are an extension for checking voicemail via
    ; voicemailmain
    ;
    ;exten => 8500,1,VoicemailMain
    ;exten => 8500,n,Hangup
    ;
    ; Or a conference room (you'll need to edit meetme.conf to enable this room)
    ;
    ;exten => 8600,1,Meetme(1234)
    ;
    ; Or playing an announcement to the called party, as soon it answers
    ;
    ;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
    ;
    ; example of a compartmentalized company called "acme"
    ;
    ; this is the context that your incoming IAX/SIP trunk dumps you in...
    ;[acme-incoming]
    ;exten => s,1,Wait(1)
    ;exten => s,n,Answer()
    ;exten => s,n(menu),Playback(acme/vm-brief-menu)
    ;exten => s,n(exten),Background(vm-enter-num-to-call)
    ;exten => s,n,WaitExten(5)
    ;exten => s,n(goodbye),Playback(vm-goodbye)
    ;exten => s,n(end),Hangup()
    ;
    ;include  => acme-extens
    ;
    ;exten => i,1,Playback(vm-invalid)
    ;exten => i,n,Goto(s,exten)                     ; optionally, transfer to operator
    ;
    ;exten => t,1,Goto(s,goodbye)
    ;
    ; this is the context our internal SIP hardphones use (see sip.conf)
    ;
    ;[acme-internal]
    ;exten => s,1,Answer()
    ;exten => s,n(exten),Background(vm-enter-num-to-call)
    ;exten => s,n,WaitExten(5)
    ;exten => s,n(goodbye),Playback(vm-goodbye)
    ;exten => s,n(end),Hangup()
    ;
    ;include => trunkint
    ;include => trunkld
    ;include => trunklocal
    ;
    ;include => acme-extens
    ;
    ; you can test what your system sounds like to outside callers by dialing this
    ;exten => 777,1,DISA(no-password,acme-incoming)
    ;
    ; grouping of acme's extensions... never used directly, always included.
    ;
    ;[acme-extens]
    ;include => stdexten
    ;exten => 111,1,Gosub(111,stdexten(SIP/pete_1,acme))
    ;exten => 111,n,Goto(s,exten)
    ;
    ;exten => 112,1,Gosub(112,stdexten(SIP/nancy_1,acme))
    ;exten => 112,n,Goto(s,end)
    ;
    ; end of acme example
    ;
    ; Time context: you can patch this in via the following.
    ;
    ; [acme-internal]
    ; ...
    ; exten => 777,1,Gosub(time)
    ; exten => 777,n,Hangup()
    ;
    ; ...
    ; include => time
    ;
    ; Note: if you're geographically spread out, you can have SIP extensions
    ; specify their own local timezone in sip.conf as:
    ;
    ; [boi]
    ; type=friend
    ; context=acme-internal
    ; callerid="Boise Ofc. <2083451111>"
    ; ...
    ; ; use system-wide default timezone of MST7MDT
    ;
    ; [lws]
    ; type=friend
    ; context=acme-internal
    ; callerid="Lewiston Ofc. <2087431111>"
    ; ...
    ; setvar=timezone=PST8PDT
    ;
    ; "timezone" isn't a 'reserved' name in any way, and other places where
    ; the timezone is significant (e.g. calls to "SayUnixTime()", etc) will
    ; require modification as well.  Note that voicemail.conf already has
    ; a mechanism for timezones.
    ;
    [time]
    exten => _X.,30000(time),NoOp(Time: ${EXTEN} ${timezone})
    exten => _X.,n,Wait(0.25)
    exten => _X.,n,Answer()
    ; the amount of delay is set for English; you may need to adjust this time
    ; for other languages if there's no pause before the synchronizing beep.
    exten => _X.,n,Set(FUTURETIME=$[${EPOCH} + 12])
    exten => _X.,n,SayUnixTime(${FUTURETIME},Zulu,HNS)
    exten => _X.,n,SayPhonetic(z)
    ; use the timezone associated with the extension (sip only), or system-wide
    ; default if one hasn't been set.
    exten => _X.,n,SayUnixTime(${FUTURETIME},${timezone},HNS)
    exten => _X.,n,Playback(spy-local)
    exten => _X.,n,WaitUntil(${FUTURETIME})
    exten => _X.,n,Playback(beep)
    exten => _X.,n,Return()
    ;
    ; ANI context: use in the same way as "time" above
    ;
    [ani]
    exten => _X.,40000(ani),NoOp(ANI: ${EXTEN})
    exten => _X.,n,Wait(0.25)
    exten => _X.,n,Answer()
    exten => _X.,n,Playback(vm-from)
    exten => _X.,n,SayDigits(${CALLERID(ani)})
    exten => _X.,n,Wait(1.25)
    exten => _X.,n,SayDigits(${CALLERID(ani)})      ; playback again in case of missed digit
    exten => _X.,n,Return()
    ; For more information on applications, just type "core show applications" at your
    ; friendly Asterisk CLI prompt.
    ;
    ; "core show application <command>" will show details of how you
    ; use that particular application in this file, the dial plan.
    ; "core show functions" will list all dialplan functions
    ; "core show function <COMMAND>" will show you more information about
    ; one function. Remember that function names are UPPER CASE.
    
    sip
    >
    sip.conf         sip_notify.conf
    
    sip.conf
    >
    [general]
    context=default
    allowoverlap=no
    udpbindaddr=0.0.0.0
    tcpenable=no
    tcpbindaddr=0.0.0.0
    srvlookup=yes
    [2101]
    type=friend     ;friend, user, peer
    secret=1234
    host=dynamic
    [2102]
    type=friend     ;friend, user, peer
    secret=1234
    host=dynamic
    

    Статистика

    Время первой команды журнала13:15:27 2011-10-17
    Время последней команды журнала08:46:24 2011-10-18
    Количество командных строк в журнале100
    Процент команд с ненулевым кодом завершения, %12.00
    Процент синтаксически неверно набранных команд, % 4.00
    Суммарное время работы с терминалом *, час 3.57
    Количество командных строк в единицу времени, команда/мин 0.47
    Частота использования команд
    asterisk12|==========| 10.17%
    route10|========| 8.47%
    vim10|========| 8.47%
    cat10|========| 8.47%
    apt-cache9|=======| 7.63%
    ping7|=====| 5.93%
    apt-get6|=====| 5.08%
    ls5|====| 4.24%
    cp4|===| 3.39%
    less4|===| 3.39%
    grep4|===| 3.39%
    ~4|===| 3.39%
    '3|==| 2.54%
    expand3|==| 2.54%
    cd3|==| 2.54%
    head3|==| 2.54%
    tail3|==| 2.54%
    sed3|==| 2.54%
    ifconfig2|=| 1.69%
    ifdonw2|=| 1.69%
    /etc/init.d/isc-dhcp-server2|=| 1.69%
    ps1|| 0.85%
    passwordpassword1|| 0.85%
    #otlichno1|| 0.85%
    ifup1|| 0.85%
    /etc/init.d/asterisk1|| 0.85%
    apt-1|| 0.85%
    ifdown1|| 0.85%
    [2102]1|| 0.85%
    su1|| 0.85%
    ____
    *) Интервалы неактивности длительностью 30 минут и более не учитываются

    Справка

    Для того чтобы использовать LiLaLo, не нужно знать ничего особенного: всё происходит само собой. Однако, чтобы ведение и последующее использование журналов было как можно более эффективным, желательно иметь в виду следующее:
    1. В журнал автоматически попадают все команды, данные в любом терминале системы.

    2. 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.

    3. Команды, при наборе которых были допущены синтаксические ошибки, выводятся перечёркнутым текстом:
      $ l s-l
      bash: l: command not found
      

    4. Если код завершения команды равен нулю, команда была выполнена без ошибок. Команды, код завершения которых отличен от нуля, выделяются цветом.
      $ test 5 -lt 4
      Обратите внимание на то, что код завершения команды может быть отличен от нуля не только в тех случаях, когда команда была выполнена с ошибкой. Многие команды используют код завершения, например, для того чтобы показать результаты проверки

    5. 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
      

    6. Команды, выполненные с привилегиями суперпользователя, выделяются слева красной чертой.
      # id
      uid=0(root) gid=0(root) Gruppen=0(root)
      

    7. 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
      2a3,5
      >    if [ -f /usr/local/etc/bash_completion ]; then
      >         . /usr/local/etc/bash_completion
      >        fi
      

    8. 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
      В данном случае изменения применяются к файлу ~/.bashrc

    9. 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.

    10. 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.

    11. Время ввода команды, показанное в журнале, соответствует времени начала ввода командной строки, которое равно тому моменту, когда на терминале появилось приглашение интерпретатора

    12. 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.

    13. 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.

    14. 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?

    15. 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
      
      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.
      Для разделения нескольких абзацев между собой используйте символ "-", один в строке.

    16. Комментарии, не относящиеся непосредственно ни к какой из команд, добавляются точно таким же способом, только вместо симолов #^ или #v нужно использовать символы #=

    17. Содержимое файла может быть показано в журнале. Для этого его нужно вывести с помощью программы cat. Если вывод команды отметить симоволами #!, содержимое файла будет показано в журнале в специально отведённой для этого секции.
    18. Для того чтобы вставить скриншот интересующего вас окна в журнал, нужно воспользоваться командой l3shot. После того как команда вызвана, нужно с помощью мыши выбрать окно, которое должно быть в журнале.
    19. Команды в журнале расположены в хронологическом порядке. Если две команды давались одна за другой, но на разных терминалах, в журнале они будут рядом, даже если они не имеют друг к другу никакого отношения.
      1
          2
      3   
          4
      
      Группы 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.

    О программе

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

    $Id$