Журнал лабораторных работ

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Журнал

Понедельник (11/24/14)

/dev/pts/2
09:08:50
#screen
/dev/pts/1
09:08:50
#screen
/dev/pts/6
09:16:04
#screen -x
/dev/pts/4
09:16:08
###### aaaaa

09:16:10
#ls -ls
total 0
прошло 11 минут
09:27:55
#шifconfig
eth0      Link encap:Ethernet  HWaddr 00:22:4d:82:f9:7a
          inet addr:192.168.12.8  Bcast:192.168.12.255  Mask:255.255.255.0
          inet6 addr: fe80::222:4dff:fe82:f97a/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:29332 errors:0 dropped:4 overruns:0 frame:0
          TX packets:17820 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:20043600 (19.1 MiB)  TX bytes:2028238 (1.9 MiB)
          Interrupt:20 Memory:fe600000-fe620000
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:183 errors:0 dropped:0 overruns:0 frame:0
          TX packets:183 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:11069 (10.8 KiB)  TX bytes:11069 (10.8 KiB)
прошло 116 минут
11:24:07
#screen -ls
11:27:38
#alias exit='echo no exit'

11:27:41
#зшping 192.168.12.253
PING 192.168.12.253 (192.168.12.253) 56(84) bytes of data.
64 bytes from 192.168.12.253: icmp_req=1 ttl=64 time=10.7 ms
64 bytes from 192.168.12.253: icmp_req=2 ttl=64 time=0.591 ms
^C
--- 192.168.12.253 ping statistics ---
2 packets transmitted, 2 received, 0% packet loss, time 1001ms
rtt min/avg/max/mdev = 0.591/5.649/10.707/5.058 ms
прошло 68 минут
12:36:05
#ping 192.168.12.253
PING 192.168.12.253 (192.168.12.253) 56(84) bytes of data.
64 bytes from 192.168.12.253: icmp_req=1 ttl=64 time=0.920 ms
64 bytes from 192.168.12.253: icmp_req=2 ttl=64 time=0.576 ms
^C
--- 192.168.12.253 ping statistics ---
2 packets transmitted, 2 received, 0% packet loss, time 999ms
rtt min/avg/max/mdev = 0.576/0.748/0.920/0.172 ms
12:37:59
#мшÑvim /etc/network/interfaces
iface lo inet loopback
# The primary networ.80.1erface
allow-hotplug eth0  .80
iface eth0 inet static
~
~
~
~
~
~
...
~
~
~
~
~
~
~
~
~
"/etc/network/interfaces" 10L, 188C written
прошло 19 минут
12:57:23
#vim /etc/network/interfaces
12:57:46
#шifdown eth0ц
RTNETLINK answers: No such process
прошло 26 минут
13:24:31
#вÑifconfig
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:293 errors:0 dropped:0 overruns:0 frame:0
          TX packets:293 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:20777 (20.2 KiB)  TX bytes:20777 (20.2 KiB)
13:25:27
#ifup eth0
ifup: interface eth0 already configured
13:25:27
#ifconfig
eth0      Link encap:Ethernet  HWaddr 00:22:4d:82:f9:7a
          inet addr:192.168.80.1  Bcast:192.168.80.255  Mask:255.255.255.0
          inet6 addr: fe80::222:4dff:fe82:f97a/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:38883 errors:0 dropped:4 overruns:0 frame:0
          TX packets:22659 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:21648849 (20.6 MiB)  TX bytes:2492076 (2.3 MiB)
          Interrupt:20 Memory:fe600000-fe620000
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:306 errors:0 dropped:0 overruns:0 frame:0
          TX packets:306 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:21960 (21.4 KiB)  TX bytes:21960 (21.4 KiB)
13:25:39
#ping 192.168.80.254
PING 192.168.80.254 (192.168.80.254) 56(84) bytes of data.
From 192.168.80.1 icmp_seq=2 Destination Host Unreachable
From 192.168.80.1 icmp_seq=3 Destination Host Unreachable
From 192.168.80.1 icmp_seq=4 Destination Host Unreachable
From 192.168.80.1 icmp_seq=5 Destination Host Unreachable
^C
--- 192.168.80.254 ping statistics ---
7 packets transmitted, 0 received, +4 errors, 100% packet loss, time 6032ms
pipe 3
13:25:53
#ping 192.168.70.254
PING 192.168.70.254 (192.168.70.254) 56(84) bytes of data.
From 192.168.80.1 icmp_seq=1 Destination Host Unreachable
From 192.168.80.1 icmp_seq=2 Destination Host Unreachable
From 192.168.80.1 icmp_seq=3 Destination Host Unreachable
^C
--- 192.168.70.254 ping statistics ---
5 packets transmitted, 0 received, +3 errors, 100% packet loss, time 4024ms
pipe 3
13:26:02
#route -n
Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
0.0.0.0         192.168.80.254  0.0.0.0         UG    0      0        0 eth0
192.168.80.0    0.0.0.0         255.255.255.0   U     0      0        0 eth0
13:26:55
#ping 192.168.80.254
PING 192.168.80.254 (192.168.80.254) 56(84) bytes of data.
64 bytes from 192.168.80.254: icmp_req=1 ttl=64 time=0.611 ms
64 bytes from 192.168.80.254: icmp_req=2 ttl=64 time=0.754 ms
^C
--- 192.168.80.254 ping statistics ---
2 packets transmitted, 2 received, 0% packet loss, time 999ms
rtt min/avg/max/mdev = 0.611/0.682/0.754/0.076 ms
13:27:03
#ping 192.168.70.254
PING 192.168.70.254 (192.168.70.254) 56(84) bytes of data.
64 bytes from 192.168.70.254: icmp_req=1 ttl=64 time=0.594 ms
64 bytes from 192.168.70.254: icmp_req=2 ttl=64 time=0.579 ms
^C
--- 192.168.70.254 ping statistics ---
2 packets transmitted, 2 received, 0% packet loss, time 999ms
rtt min/avg/max/mdev = 0.579/0.586/0.594/0.025 ms
13:27:06
#ping 192.168.12.254
PING 192.168.12.254 (192.168.12.254) 56(84) bytes of data.
64 bytes from 192.168.12.254: icmp_req=1 ttl=63 time=0.326 ms
^C
--- 192.168.12.254 ping statistics ---
1 packets transmitted, 1 received, 0% packet loss, time 0ms
rtt min/avg/max/mdev = 0.326/0.326/0.326/0.000 ms
13:27:26
#ping ya.ru
PING ya.ru (213.180.204.3) 56(84) bytes of data.
From 192.168.80.254 icmp_seq=1 Destination Host Unreachable
^CFrom 192.168.80.254 icmp_seq=2 Destination Host Unreachable
--- ya.ru ping statistics ---
2 packets transmitted, 0 received, +2 errors, 100% packet loss, time 5064ms
13:28:07
#ping ya.ru
PING ya.ru (93.158.134.3) 56(84) bytes of data.
^CFrom 192.168.80.254 icmp_seq=1 Destination Host Unreachable
--- ya.ru ping statistics ---
1 packets transmitted, 0 received, +1 errors, 100% packet loss, time 0ms
13:28:12
#vi /etc/resolv
13:28:12
#vi /etc/resolv.conf
13:28:45
#apt-cache search dhcp server
ahcpd - Ad-Hoc Configuration Protocol
avahi-autoipd - Avahi IPv4LL network address configuration daemon
bootp - server for the bootp protocol with DHCP support
udhcpd - Provides the busybox DHCP server implementation
dhcp-helper - A DHCP relay agent
dhcp-probe - network DHCP or BootP server discover
dhcpdump - Parse DHCP packets from tcpdump
dhcping - DHCP Daemon Ping Program
dhis-server - Dynamic Host Information System - server
dibbler-client - portable DHCPv6 client
...
ltsp-server-standalone - complete LTSP server environment
nagios-plugins-basic - Plugins for nagios compatible monitoring systems
netdiscover - active/passive network address scanner using arp requests
netdiscover-dbg - active/passive network address scanner using arp requests (debug)
pump - BOOTP and DHCP client for automatic IP configuration
pxe - free PXE daemon
python-pydhcplib - Python DHCP client/server library
python-pypureomapi - ISC DHCP OMAPI protocol implementation in Python
resolvconf - name server information handler
wide-dhcpv6-server - DHCPv6 server for automatic IPv6 hosts configuration
прошло 15 минут
13:44:05
#apt-get install isc-dhcp-server
Reading package lists... Done
Building dependency tree
Reading state information... Done
Suggested packages:
  isc-dhcp-server-ldap
The following NEW packages will be installed:
  isc-dhcp-server
0 upgraded, 1 newly installed, 0 to remove and 0 not upgraded.
Need to get 935 kB of archives.
After this operation, 2,075 kB of additional disk space will be used.
...
Preconfiguring packages ...
Selecting previously unselected package isc-dhcp-server.
(Reading database ... 128692 files and directories currently installed.)
Unpacking isc-dhcp-server (from .../isc-dhcp-server_4.2.2.dfsg.1-5+deb70u6_amd64.deb) ...
Processing triggers for man-db ...
Setting up isc-dhcp-server (4.2.2.dfsg.1-5+deb70u6) ...
Generating /etc/default/isc-dhcp-server...
Starting ISC DHCP server: dhcpdcheck syslog for diagnostics. ... failed!
 failed!
invoke-rc.d: initscript isc-dhcp-server, action "start" failed.
13:45:08
#фÑ

/dev/pts/0
13:46:18
#screen -x
/dev/pts/4
13:47:55
#cat /etc/default/isc-dhcp-server
# Defaults for isc-dhcp-server initscript
# sourced by /etc/init.d/isc-dhcp-server
# installed at /etc/default/isc-dhcp-server by the maintainer scripts
#
# This is a POSIX shell fragment
#
# Path to dhcpd's config file (default: /etc/dhcp/dhcpd.conf).
#DHCPD_CONF=/etc/dhcp/dhcpd.conf
# Path to dhcpd's PID file (default: /var/run/dhcpd.pid).
#DHCPD_PID=/var/run/dhcpd.pid
# Additional options to start dhcpd with.
#       Don't use options -cf or -pf here; use DHCPD_CONF/ DHCPD_PID instead
#OPTIONS=""
# On what interfaces should the DHCP server (dhcpd) serve DHCP requests?
#       Separate multiple interfaces with spaces, e.g. "eth0 eth1".
INTERFACES=""
13:49:01
#grep INTER /etc/default/isc-dhcp-server
INTERFACES=""
13:56:29
#Defaults for isc-dhcp-server initscript

13:57:15
#~

13:57:41
#vi /etc/default/isc-dhcp-server
13:57:59
#cat /etc/init.d/isc-dhcp-server
#!/bin/sh
#
#
### BEGIN INIT INFO
# Provides:          isc-dhcp-server
# Required-Start:    $remote_fs $network $syslog
# Required-Stop:     $remote_fs $network $syslog
# Should-Start:      $local_fs slapd $named
# Should-Stop:       $local_fs slapd
# Default-Start:     2 3 4 5
...
        status)
                echo -n "Status of $DESC: "
                check_status -v
                exit "$?"
                ;;
        *)
                echo "Usage: $0 {start|stop|restart|force-reload|status}"
                exit 1
esac
exit 0
14:01:48
#cat /etc/dhcp/dhcpd.conf
#
# Sample configuration file for ISC dhcpd for Debian
#
#
# The ddns-updates-style parameter controls whether or not the server will
# attempt to do a DNS update when a lease is confirmed. We default to the
# behavior of the version 2 packages ('none', since DHCP v2 didn't
# have support for DDNS.)
ddns-update-style none;
# option definitions common to all supported networks...
option domain-name "unix.nt";
option domain-name-servers 192.168.12.254;
default-lease-time 600;
max-lease-time 7200;
log-facility local7;
subnet 192.168.80.0 netmask 255.255.255.0 {
  range 192.168.80.100 192.168.80.200;
  option routers 192.168.80.254;
}
14:01:52
#/etc/init.d/isc-dhcp-server restart
Stopping ISC DHCP server: dhcpd failed!
Starting ISC DHCP server: dhcpd.
14:09:06
#сÑcat /var/lib/dhcp/dhclient.eth0.leases
lease {
  interface "eth0";
  fixed-address 192.168.12.103;
  filename "\\SMSBoot\\x86\\wdsnbp.com";
  option subnet-mask 255.255.255.0;
  option routers 192.168.12.1;
  option dhcp-lease-time 86400;
  option dhcp-message-type 5;
  option domain-name-servers 192.168.10.240;
  option dhcp-server-identifier 192.168.10.240;
  option dhcp-renewal-time 43200;
  option dhcp-rebinding-time 75600;
  option vendor-encapsulated-options 1:4:0:0:0:0:ff;
  option domain-name "class.msft";
  renew 0 2014/11/23 18:49:51;
  rebind 1 2014/11/24 06:29:34;
  expire 1 2014/11/24 09:29:34;
}
прошло 15 минут
14:25:01
#cat /var/lib/dhcp/dhclient.leases

14:25:08
#cat /var/lib/dhcp/dhcpd.leases
# The format of this file is documented in the dhcpd.leases(5) manual page.
# This lease file was written by isc-dhcp-4.2.2
server-duid "\000\001\000\001\034\005\352p\000\"M\202\371z";
lease 192.168.80.100 {
  starts 1 2014/11/24 13:09:20;
  ends 1 2014/11/24 13:19:20;
  cltt 1 2014/11/24 13:09:20;
  binding state active;
  next binding state free;
  rewind binding state free;
...
  starts 1 2014/11/24 13:24:22;
  ends 1 2014/11/24 13:34:22;
  cltt 1 2014/11/24 13:24:22;
  binding state active;
  next binding state free;
  rewind binding state free;
  hardware ethernet 00:19:56:fd:74:89;
  uid "\001\000\031V\375t\211";
  client-hostname "SEP001956FD7489";
}
14:25:39
#gedit
(gedit:7447): Gtk-WARNING **: Attempting to store changes into `/root/.local/share/recently-used.xbel', but failed: Failed to create file '/root/.local/share/recently-used.xbel.5UKAQX': No such file or directory
(gedit:7447): Gtk-WARNING **: Attempting to set the permissions of `/root/.local/share/recently-used.xbel', but failed: No such file or directory
(gedit:7447): Gtk-WARNING **: Attempting to store changes into `/root/.local/share/recently-used.xbel', but failed: Failed to create file '/root/.local/share/recently-used.xbel.WEX7PX': No such file or directory
(gedit:7447): Gtk-WARNING **: Attempting to set the permissions of `/root/.local/share/recently-used.xbel', but failed: No such file or directory
(gedit:7447): Gtk-WARNING **: Attempting to store changes into `/root/.local/share/recently-used.xbel', but failed: Failed to create file '/root/.local/share/recently-used.xbel.Z74QPX': No such file or directory
(gedit:7447): Gtk-WARNING **: Attempting to set the permissions of `/root/.local/share/recently-used.xbel', but failed: No such file or directory
прошло 27 минут
14:53:36
##Install Asterisk

14:56:00
## какие пакеты связаны с asterisk вообще

14:56:46
#apt-cache search asterisk | wc -l
67
15:04:11
#dpkg -l asterisk-config | grep /etc

15:04:40
#dpkg -L asterisk-config | grep /etc
/etc
/etc/asterisk
/etc/asterisk/app_mysql.conf
/etc/asterisk/ccss.conf
/etc/asterisk/iax.conf
/etc/asterisk/dundi.conf
/etc/asterisk/udptl.conf
/etc/asterisk/res_pktccops.conf
/etc/asterisk/cdr_mysql.conf
/etc/asterisk/manager.d
...
/etc/asterisk/iaxprov.conf
/etc/asterisk/alsa.conf
/etc/asterisk/manager.conf
/etc/asterisk/cdr_adaptive_odbc.conf
/etc/asterisk/extconfig.conf
/etc/asterisk/enum.conf
/etc/asterisk/extensions_minivm.conf
/etc/asterisk/cli.conf
/etc/asterisk/cel_tds.conf
/etc/asterisk/dnsmgr.conf
15:11:25
#dpkg-reconfigure asterisk-config

прошло 16 минут
15:27:42
#dpkg -L asterisk-config | sort
/.
/etc
/etc/asterisk
/etc/asterisk/adsi.conf
/etc/asterisk/agents.conf
/etc/asterisk/ais.conf
/etc/asterisk/alarmreceiver.conf
/etc/asterisk/alsa.conf
/etc/asterisk/amd.conf
/etc/asterisk/app_mysql.conf
...
/usr/share/doc/asterisk-config/examples/configs/users.conf.sample
/usr/share/doc/asterisk-config/examples/configs/voicemail.conf.sample.gz
/usr/share/doc/asterisk-config/examples/configs/vpb.conf.sample.gz
/usr/share/doc/asterisk-config/examples/eagi-sphinx-test.c.gz
/usr/share/doc/asterisk-config/examples/eagi-test.c
/usr/share/doc/asterisk-config/examples/fastagi-test
/usr/share/doc/asterisk-config/NEWS.Debian.gz
/usr/share/lintian
/usr/share/lintian/overrides
/usr/share/lintian/overrides/asterisk-config
15:29:31
#ls /usr/lib/as
aspell/   asterisk/
15:29:31
#ls /usr/lib/as
aspell/   asterisk/
15:29:31
#ls /usr/lib/as
aspell/   asterisk/
15:29:31
#ls /usr/lib/asterisk/modules/
Display all 223 possibilities? (y or n)
app_adsiprog.so             cdr_custom.so               func_extstate.so
app_alarmreceiver.so        cdr_manager.so              func_frame_trace.so
app_amd.so                  cdr_odbc.so                 func_global.so
app_authenticate.so         cdr_pgsql.so                func_groupcount.so
app_cdr.so                  cdr_radius.so               func_iconv.so
app_celgenuserevent.so      cdr_sqlite3_custom.so       func_lock.so
app_chanisavail.so          cdr_sqlite.so               func_logic.so
app_channelredirect.so      cdr_syslog.so               func_math.so
app_chanspy.so              cdr_tds.so                  func_md5.so
...
app_waitforsilence.so       func_channel.so             res_security_log.so
app_waituntil.so            func_config.so              res_smdi.so
app_while.so                func_curl.so                res_snmp.so
app_zapateller.so           func_cut.so                 res_speech.so
bridge_builtin_features.so  func_db.so                  res_srtp.so
bridge_multiplexed.so       func_devstate.so            res_stun_monitor.so
bridge_simple.so            func_dialgroup.so           res_timing_pthread.so
bridge_softmix.so           func_dialplan.so            res_timing_timerfd.so
cdr_adaptive_odbc.so        func_enum.so
cdr_csv.so                  func_env.so
15:30:47
#ls -s /usr/lib/asterisk/modules/ | sort
112 res_jabber.so
116 chan_unistim.so
128 pbx_dundi.so
 12 app_celgenuserevent.so
 12 app_chanisavail.so
 12 app_channelredirect.so
 12 app_db.so
 12 app_echo.so
 12 app_exec.so
 12 app_flash.so
...
 68 app_dial.so
 68 app_followme.so
 68 chan_agent.so
 68 res_config_pgsql.so
 72 chan_vpb.so
 76 res_fax.so
 84 app_minivm.so
 88 res_agi.so
  8 app_cdr.so
total 6644
15:30:55
#ls /usr/lib/asterisk/modules/ | sort
app_adsiprog.so
app_alarmreceiver.so
app_amd.so
app_authenticate.so
app_cdr.so
app_celgenuserevent.so
app_chanisavail.so
app_channelredirect.so
app_chanspy.so
app_confbridge.so
...
res_rtp_asterisk.so
res_rtp_multicast.so
res_security_log.so
res_smdi.so
res_snmp.so
res_speech.so
res_srtp.so
res_stun_monitor.so
res_timing_pthread.so
res_timing_timerfd.so
15:37:35
#ps aux | grep [a]ster
asterisk 12578  2.7  0.1 756256 30260 ?        Ssl  16:37   0:00 /usr/sbin/asterisk -p -U asterisk
asterisk 12579  0.0  0.0  13916   772 ?        S    16:37   0:00 astcanary /var/run/asterisk/alt.asterisk.canary.tweet.tweet.tweet 12578
15:37:45
#netstat -lnp | grep asterisk
tcp        0      0 127.0.0.1:5038          0.0.0.0:*               LISTEN      12578/asterisk
tcp        0      0 0.0.0.0:2000            0.0.0.0:*               LISTEN      12578/asterisk
udp        0      0 0.0.0.0:4520            0.0.0.0:*                           12578/asterisk
udp        0      0 0.0.0.0:4569            0.0.0.0:*                           12578/asterisk
udp        0      0 0.0.0.0:5000            0.0.0.0:*                           12578/asterisk
udp        0      0 0.0.0.0:5060            0.0.0.0:*                           12578/asterisk
unix  2      [ ACC ]     STREAM     LISTENING     105986   12578/asterisk      /var/run/asterisk/asterisk.ctl
15:38:07
#cat /var/run/asterisk/asterisk.ctl
cat: /var/run/asterisk/asterisk.ctl: No such device or address
15:38:36
##/var/run/asterisk/asterisk.ctl

15:38:45
##/var/run/asterisk/asterisk.ctl <-- socket можно подключаться к астрер

15:39:11
#ls -l /var/run/asterisk/asterisk.ctl
srwxrwx--- 1 asterisk asterisk 0 Nov 24 16:37 /var/run/asterisk/asterisk.ctl
15:39:31
#asterisk --r
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.
=========================================================================
Connected to Asterisk 1.8.13.1~dfsg1-3+deb7u3 currently running on debian8 (pid = 12578)
debian8*CLI>
!            ael          agent        agi          aoc          calendar     cc
...
System Statistics
-----------------
  System Uptime:             7 hours
  Total RAM:                 16387536 KiB
  Free RAM:                  14977292 KiB
  Buffer RAM:                69772 KiB
  Total Swap Space:          19786748 KiB
  Free Swap Space:           19786748 KiB
  Number of Processes:       371
debian8*CLI> quit
15:45:25
#mv /etc/asterisk/extensions.conf{,.bak}

15:46:17
#ls /etc/asterisk/
adsi.conf                cel_pgsql.conf           func_odbc.conf    queues.conf
agents.conf              cel_sqlite3_custom.conf  gtalk.conf        res_config_mysql.conf
ais.conf                 cel_tds.conf             h323.conf         res_config_sqlite.conf
alarmreceiver.conf       chan_dahdi.conf          http.conf         res_curl.conf
alsa.conf                chan_mobile.conf         iax.conf          res_fax.conf
amd.conf                 chan_ooh323.conf         iaxprov.conf      res_ldap.conf
app_mysql.conf           cli_aliases.conf         indications.conf  res_odbc.conf
asterisk.adsi            cli.conf                 jabber.conf       res_pgsql.conf
asterisk.conf            cli_permissions.conf     jingle.conf       res_pktccops.conf
calendar.conf            codecs.conf              logger.conf       res_snmp.conf
...
cdr_manager.conf         dundi.conf               minivm.conf       sip_notify.conf
cdr_mysql.conf           enum.conf                misdn.conf        skinny.conf
cdr_odbc.conf            extconfig.conf           modules.conf      sla.conf
cdr_pgsql.conf           extensions.ael           musiconhold.conf  smdi.conf
cdr_sqlite3_custom.conf  extensions.conf.bak      muted.conf        telcordia-1.adsi
cdr_syslog.conf          extensions.lua           osp.conf          udptl.conf
cdr_tds.conf             extensions_minivm.conf   oss.conf          unistim.conf
cel.conf                 features.conf            phone.conf        users.conf
cel_custom.conf          festival.conf            phoneprov.conf    voicemail.conf
cel_odbc.conf            followme.conf            queuerules.conf   vpb.conf
15:46:54
#scp /etc/asterisk/extensions.conf user@192.168.60.1://etc/asterisk/
The authenticity of host '192.168.60.1 (192.168.60.1)' can't be established.
ECDSA key fingerprint is e1:18:22:cd:45:86:57:45:3c:48:90:a2:fa:d8:c3:ac.
Are you sure you want to continue connecting (yes/no)? yes
Warning: Permanently added '192.168.60.1' (ECDSA) to the list of known hosts.
user@192.168.60.1's password:
/etc/asterisk/extensions.conf: No such file or directory
15:53:21
#scp /etc/asterisk/extensions.conf.bak user@192.168.60.1://etc/asterisk/
user@192.168.60.1's password:
scp: //etc/asterisk//extensions.conf.bak: Permission denied
15:53:47
#scp /etc/asterisk/sip.conf.bak root@192.168.60.1://etc/asterisk/
root@192.168.60.1's password:
sip.conf.bak                                                      100%   77KB  77.3KB/s   00:00
15:58:58
#cat /etc/asterisk/sip.conf.bak
;
; SIP Configuration example for Asterisk
;
; Note: Please read the security documentation for Asterisk in order to
;       understand the risks of installing Asterisk with the sample
;       configuration. If your Asterisk is installed on a public
;       IP address connected to the Internet, you will want to learn
;       about the various security settings BEFORE you start
;       Asterisk.
;
...
;[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.
прошло 17 минут
16:16:33
#cat /etc/asterisk/sip.conf
[general]
context=default
[1101]
type=friend
secret=1234
host=dynamic
[1102]
type=friend
secret=1234
host=dynamic
16:16:44
#exit
exit
Connection to 192.168.10.1 closed.
16:17:06
#/etc/init.d/asterisk restart
Stopping Asterisk PBX: asterisk.
Starting Asterisk PBX: asterisk.
16:17:14
#asterisk -r
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.
=========================================================================
Connected to Asterisk 1.8.13.1~dfsg1-3+deb7u3 currently running on debian8 (pid = 13121)
debian8*CLI> quit
/dev/pts/7
16:18:39
#cat /etc/asterisk/sip.conf.bak
;
; SIP Configuration example for Asterisk
;
; Note: Please read the security documentation for Asterisk in order to
;       understand the risks of installing Asterisk with the sample
;       configuration. If your Asterisk is installed on a public
;       IP address connected to the Internet, you will want to learn
;       about the various security settings BEFORE you start
;       Asterisk.
;
...
;[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.
/dev/pts/4
16:19:54
#/etc/init.d/asterisk restart
Stopping Asterisk PBX: asterisk.
Starting Asterisk PBX: asterisk.
16:20:26
#vi /etc/asterisk/sip
16:20:26
#vi /etc/asterisk/sip.conf
прошла 51 минута
17:11:47
#cat /etc/asterisk/extensions.conf
[default]
exten => _32XX,1,Dial(SIP/${EXTEN})

Вторник (11/25/14)

/dev/pts/2
08:42:28
#screen
/dev/pts/1
08:42:28
#screen
/dev/pts/6
08:46:16
#screen -x
/dev/pts/4
08:47:56
#asterisk -rvvv
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 debian8 (pid = 2372)
Verbosity is at least 3
debian8*CLI> sip reload
 Reloading SIP
debian8*CLI> quit
Executing last minute cleanups
/dev/pts/8
08:49:53
#мvi /etc/asterisk/sip
sip.conf         sip.conf.bak     sip_notify.conf
08:49:53
#vi /etc/asterisk/sip.conf
--- /tmp/l3-saved-4147.3158.28203	2014-11-25 09:52:37.917402462 +0200
+++ /etc/asterisk/sip.conf	2014-11-25 09:52:49.781402090 +0200
@@ -1,5 +1,6 @@
 [general]
 context=default
+qualify=yes                                                                                
 
 [3201]
 type=friend
прошло 47 минут
09:37:14
#фыÐcat /etc/asterisk/extensions.
extensions.ael       extensions.conf      extensions.conf.bak  extensions.lua
09:37:14
#cat /etc/asterisk/extensions.conf
[default]
exten => _32XX,1,Dial(SIP/${EXTEN})
09:37:46
#ьлÐmkdir /var/tmp/asterisk

прошло 74 минуты
10:51:48
#ls -ls /etc/asterisk/
total 792
 4 -rw-r----- 1 asterisk asterisk   140 Jan  4  2014 adsi.conf
 4 -rw-r----- 1 asterisk asterisk  2760 Jan  4  2014 agents.conf
 4 -rw-r----- 1 asterisk asterisk  2904 Jan  4  2014 ais.conf
 4 -rw-r----- 1 asterisk asterisk  2084 Jan  4  2014 alarmreceiver.conf
 4 -rw-r----- 1 asterisk asterisk  3498 Jan  4  2014 alsa.conf
 4 -rw-r----- 1 asterisk asterisk   767 Jan  4  2014 amd.conf
 4 -rw-r----- 1 asterisk asterisk  1044 Jan  4  2014 app_mysql.conf
 4 -rw-r----- 1 asterisk asterisk  3254 Jan  4  2014 asterisk.adsi
 4 -rw-r----- 1 asterisk asterisk  3465 Jan  4  2014 asterisk.conf
...
 4 -rw-r----- 1 asterisk asterisk   742 Jan  4  2014 sip_notify.conf
12 -rw-r----- 1 asterisk asterisk  9185 Jan  4  2014 skinny.conf
 8 -rw-r----- 1 asterisk asterisk  6774 Jan  4  2014 sla.conf
 4 -rw-r----- 1 asterisk asterisk  2669 Jan  4  2014 smdi.conf
 4 -rw-r----- 1 asterisk asterisk  1384 Jan  4  2014 telcordia-1.adsi
 4 -rw-r----- 1 asterisk asterisk   656 Jan  4  2014 udptl.conf
 8 -rw-r----- 1 asterisk asterisk  5096 Jan  4  2014 unistim.conf
 4 -rw-r----- 1 asterisk asterisk  2423 Jan  4  2014 users.conf
24 -rw-r----- 1 asterisk asterisk 21238 Jan  4  2014 voicemail.conf
 8 -rw-r----- 1 asterisk asterisk  5939 Jan  4  2014 vpb.conf
10:52:42
#ls /var/tmp/
asterisk
10:52:46
#ls - ls/var/tmp
ls: cannot access -: No such file or directory
ls: cannot access ls/var/tmp: No such file or directory
10:52:52
#ls -ls /var/tmp
total 4
4 drwxr-xr-x 2 asterisk asterisk 4096 Nov 25 11:51 asterisk
10:53:48
#vi /etc/asterisk/ext
10:53:48
#vi /etc/asterisk/extensions.conf
--- /tmp/l3-saved-4147.23326.18687	2014-11-25 11:54:55.433172728 +0200
+++ /etc/asterisk/extensions.conf	2014-11-25 12:01:20.129160682 +0200
@@ -2,3 +2,7 @@
 
 exten => _32XX,1,Dial(SIP/${EXTEN})
 
+exten => _81XX,1,Record(/var/tmp/asterisk/record${EXTEN:2}:gsm)
+
+exten => _82XX,1,Playback(/var/tmp/asterisk/record${EXTEN:2})
+
11:02:55
#vi /etc/asterisk/extensions.conf
11:03:52
#ls -ls /var/tmp/asterisk/
total 0
прошло 76 минут
12:20:02
#ls -ls /etc/asterisk/
total 792
 4 -rw-r----- 1 asterisk asterisk   140 Jan  4  2014 adsi.conf
 4 -rw-r----- 1 asterisk asterisk  2760 Jan  4  2014 agents.conf
 4 -rw-r----- 1 asterisk asterisk  2904 Jan  4  2014 ais.conf
 4 -rw-r----- 1 asterisk asterisk  2084 Jan  4  2014 alarmreceiver.conf
 4 -rw-r----- 1 asterisk asterisk  3498 Jan  4  2014 alsa.conf
 4 -rw-r----- 1 asterisk asterisk   767 Jan  4  2014 amd.conf
 4 -rw-r----- 1 asterisk asterisk  1044 Jan  4  2014 app_mysql.conf
 4 -rw-r----- 1 asterisk asterisk  3254 Jan  4  2014 asterisk.adsi
 4 -rw-r----- 1 asterisk asterisk  3465 Jan  4  2014 asterisk.conf
...
 4 -rw-r----- 1 asterisk asterisk   742 Jan  4  2014 sip_notify.conf
12 -rw-r----- 1 asterisk asterisk  9185 Jan  4  2014 skinny.conf
 8 -rw-r----- 1 asterisk asterisk  6774 Jan  4  2014 sla.conf
 4 -rw-r----- 1 asterisk asterisk  2669 Jan  4  2014 smdi.conf
 4 -rw-r----- 1 asterisk asterisk  1384 Jan  4  2014 telcordia-1.adsi
 4 -rw-r----- 1 asterisk asterisk   656 Jan  4  2014 udptl.conf
 8 -rw-r----- 1 asterisk asterisk  5096 Jan  4  2014 unistim.conf
 4 -rw-r----- 1 asterisk asterisk  2423 Jan  4  2014 users.conf
24 -rw-r----- 1 asterisk asterisk 21238 Jan  4  2014 voicemail.conf
 8 -rw-r----- 1 asterisk asterisk  5939 Jan  4  2014 vpb.conf
12:20:16
#vi /etc/asterisk/sip.conf
--- /tmp/l3-saved-4147.6416.29661	2014-11-25 13:21:56.725009250 +0200
+++ /etc/asterisk/sip.conf	2014-11-25 13:30:15.564993630 +0200
@@ -12,3 +12,23 @@
 secret=1234
 host=dynamic
 
+register => london:1234@192.168.12.254/liverpool
+register => london:1234@192.168.70.1/edinburgh
+
+[liverpool]
+type=friend
+host=dynamic
+secret=1234
+;чтобы все ходило через астериск
+directmedia=no
+canreinvite=no
+username=london
+
+[edinburgh]
+type=friend
+host=dynamic
+secret=1234
+;чтобы все ходило через астериск
+directmedia=no
+canreinvite=no
+username=london
прошло 16 минут
/dev/pts/4
12:36:38
#less /etc/asterisk/sip.conf
/dev/pts/8
12:36:51
#asterisk -rvvv
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 debian8 (pid = 2372)
...
debian8*CLI> sip show peers
Name/username              Host                                    Dyn Forcerport ACL Port     Status
3201/3201                  192.168.80.100                           D   N             5060     OK (13 ms)
3202/3202                  192.168.80.101                           D   N             62975    OK (1 ms)
edinburgh/london           192.168.12.254                           D   N             5060     OK (1 ms)
liverpool/london           192.168.70.1                             D   N             5060     OK (1 ms)
4 sip peers [Monitored: 4 online, 0 offline Unmonitored: 0 online, 0 offline]
debian8*CLI> sip show peers
Disconnected from Asterisk server
Executing last minute cleanups
/dev/pts/10
12:37:34
#grep register /etc/asterisk/sip.conf
register => london:1234@192.168.12.254/liverpool
register => london:1234@192.168.70.1/edinburgh
/dev/pts/8
12:38:38
#vi /etc/asterisk/sip.conf

Файлы

  • /etc/asterisk/extensions.conf
  • /etc/asterisk/sip.conf
  • /etc/asterisk/sip.conf.bak
  • /etc/default/isc-dhcp-server
  • /etc/dhcp/dhcpd.conf
  • /etc/init.d/isc-dhcp-server
  • /var/lib/dhcp/dhclient.leases
  • /var/lib/dhcp/dhcpd.leases
  • /etc/asterisk/extensions.conf
    >
    [default]
    exten => _32XX,1,Dial(SIP/${EXTEN})
    
    /etc/asterisk/sip.conf
    >
    [general]
    context=default
    [1101]
    type=friend
    secret=1234
    host=dynamic
    [1102]
    type=friend
    secret=1234
    host=dynamic
    
    /etc/asterisk/sip.conf.bak
    >
    ;
    ; SIP Configuration example for Asterisk
    ;
    ; Note: Please read the security documentation for Asterisk in order to
    ;       understand the risks of installing Asterisk with the sample
    ;       configuration. If your Asterisk is installed on a public
    ;       IP address connected to the Internet, you will want to learn
    ;       about the various security settings BEFORE you start
    ;       Asterisk.
    ;
    ;       Especially note the following settings:
    ;               - allowguest (default enabled)
    ;               - permit/deny - IP address filters
    ;               - contactpermit/contactdeny - IP address filters for registrations
    ;               - context - Which set of services you offer various users
    ;
    ; 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]
    ;        SIP/devicename/extension
    ;        SIP/devicename/extension/IPorHost
    ;        SIP/username@domain//IPorHost
    ;
    ;
    ; Devicename
    ;        devicename is defined as a peer in a section below.
    ;
    ; username@domain
    ;        Call any SIP user on the Internet
    ;        (Don't forget to enable DNS SRV records if you want to use this)
    ;
    ; devicename/extension
    ;        If you define a SIP proxy as a peer below, you may call
    ;        SIP/proxyhostname/user or SIP/user@proxyhostname
    ;        where the proxyhostname is defined in a section below
    ;        This syntax also works with ATA's with FXO ports
    ;
    ; SIP/username[:password[:md5secret[:authname]]]@host[:port]
    ;        This form allows you to specify password or md5secret and authname
    ;        without altering any authentication data in config.
    ;        Examples:
    ;
    ;        SIP/*98@mysipproxy
    ;        SIP/sales:topsecret::account02@domain.com:5062
    ;        SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
    ;
    ; IPorHost
    ;        The next server for this call regardless of domain/peer
    ;
    ; All of these dial strings specify the SIP request URI.
    ; In addition, you can specify a specific To: header by adding an
    ; exclamation mark after the dial string, like
    ;
    ;         SIP/sales@mysipproxy!sales@edvina.net
    ;
    ; A new feature for 1.8 allows one to specify a host or IP address to use
    ; when routing the call. This is typically used in tandem with func_srv if
    ; multiple methods of reaching the same domain exist. The host or IP address
    ; is specified after the third slash in the dialstring. Examples:
    ;
    ; SIP/devicename/extension/IPorHost
    ; SIP/username@domain//IPorHost
    ;
    ; CLI Commands
    ; -------------------------------------------------------------
    ; Useful CLI commands to check peers/users:
    ;   sip show peers               Show all SIP peers (including friends)
    ;   sip show registry            Show status of hosts we register with
    ;
    ;   sip set debug on             Show all SIP messages
    ;
    ;   sip reload                   Reload configuration file
    ;   sip show settings            Show the current channel configuration
    ;
    ;------- Naming devices ------------------------------------------------------
    ;
    ; When naming devices, make sure you understand how Asterisk matches calls
    ; that come in.
    ;       1. Asterisk checks the SIP From: address username and matches against
    ;          names of devices with type=user
    ;          The name is the text between square brackets [name]
    ;       2. Asterisk checks the From: addres and matches the list of devices
    ;          with a type=peer
    ;       3. Asterisk checks the IP address (and port number) that the INVITE
    ;          was sent from and matches against any devices with type=peer
    ;
    ; Don't mix extensions with the names of the devices. Devices need a unique
    ; name. The device name is *not* used as phone numbers. Phone numbers are
    ; anything you declare as an extension in the dialplan (extensions.conf).
    ;
    ; When setting up trunks, make sure there's no risk that any From: username
    ; (caller ID) will match any of your device names, because then Asterisk
    ; might match the wrong device.
    ;
    ; Note: The parameter "username" is not the username and in most cases is
    ;       not needed at all. Check below. In later releases, it's renamed
    ;       to "defaultuser" which is a better name, since it is used in
    ;       combination with the "defaultip" setting.
    ;-----------------------------------------------------------------------------
    ; ** Old configuration options **
    ; The "call-limit" configuation option is considered old is replaced
    ; by new functionality. To enable callcounters, you use the new
    ; "callcounter" setting (for extension states in queue and subscriptions)
    ; You are encouraged to use the dialplan groupcount functionality
    ; to enforce call limits instead of using this channel-specific method.
    ; You can still set limits per device in sip.conf or in a database by using
    ; "setvar" to set variables that can be used in the dialplan for various limits.
    [general]
    context=default                 ; Default context for incoming calls
    ;allowguest=no                  ; Allow or reject guest calls (default is yes)
                                    ; If your Asterisk is connected to the Internet
                                    ; and you have allowguest=yes
                                    ; you want to check which services you offer everyone
                                    ; out there, by enabling them in the default context (see below).
    ;match_auth_username=yes        ; if available, match user entry using the
                                    ; 'username' field from the authentication line
                                    ; instead of the From: field.
    allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
    ;allowoverlap=yes               ; Enable RFC3578 overlap dialing support.
                                    ; Can use the Incomplete application to collect the
                                    ; needed digits from an ambiguous dialplan match.
    ;allowoverlap=dtmf              ; Enable overlap dialing support using DTMF delivery
                                    ; methods (inband, RFC2833, SIP INFO) in the early
                                    ; media phase.  Uses the Incomplete application to
                                    ; collect the needed digits.
    ;allowtransfer=no               ; Disable all transfers (unless enabled in peers or users)
                                    ; Default is enabled. The Dial() options 't' and 'T' are not
                                    ; related as to whether SIP transfers are allowed or not.
    ;realm=mydomain.tld             ; Realm for digest authentication
                                    ; defaults to "asterisk". If you set a system name in
                                    ; asterisk.conf, it defaults to that system name
                                    ; Realms MUST be globally unique according to RFC 3261
                                    ; Set this to your host name or domain name
    ;domainsasrealm=no              ; Use domains list as realms
                                    ; You can serve multiple Realms specifying several
                                    ; 'domain=...' directives (see below).
                                    ; In this case Realm will be based on request 'From'/'To' header
                                    ; and should match one of domain names.
                                    ; Otherwise default 'realm=...' will be used.
    ; With the current situation, you can do one of four things:
    ;  a) Listen on a specific IPv4 address.      Example: bindaddr=192.0.2.1
    ;  b) Listen on a specific IPv6 address.      Example: bindaddr=2001:db8::1
    ;  c) Listen on the IPv4 wildcard.            Example: bindaddr=0.0.0.0
    ;  d) Listen on the IPv4 and IPv6 wildcards.  Example: bindaddr=::
    ; (You can choose independently for UDP, TCP, and TLS, by specifying different values for
    ; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".)
    ; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat.
    ;  IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)
    ;
    ; Using bindaddr will only enable UDP support in order to be backwards compatible with those systems
    ; that were upgraded prior to TCP support. Use udpbindaddr and tcpbindaddr to bind to UDP and TCP
    ; independently.
    ;
    ; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061
    ; for TLS).
    ;   IPv4 example: bindaddr=0.0.0.0:5062
    ;   IPv6 example: bindaddr=[::]:5062
    ;
    ; The address family of the bound UDP address is used to determine how Asterisk performs
    ; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only
    ; AAAA records are considered. In case d), both A and AAAA records are considered. Note,
    ; however, that Asterisk ignores all records except the first one. In case d), when both A
    ; and AAAA records are available, either an A or AAAA record will be first, and which one
    ; depends on the operating system. On systems using glibc, AAAA records are given
    ; priority.
    udpbindaddr=0.0.0.0             ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
                                    ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
    ; When a dialog is started with another SIP endpoint, the other endpoint
    ; should include an Allow header telling us what SIP methods the endpoint
    ; implements. However, some endpoints either do not include an Allow header
    ; or lie about what methods they implement. In the former case, Asterisk
    ; makes the assumption that the endpoint supports all known SIP methods.
    ; If you know that your SIP endpoint does not provide support for a specific
    ; method, then you may provide a comma-separated list of methods that your
    ; endpoint does not implement in the disallowed_methods option. Note that
    ; if your endpoint is truthful with its Allow header, then there is no need
    ; to set this option. This option may be set in the general section or may
    ; be set per endpoint. If this option is set both in the general section and
    ; in a peer section, then the peer setting completely overrides the general
    ; setting (i.e. the result is *not* the union of the two options).
    ;
    ; Note also that while Asterisk currently will parse an Allow header to learn
    ; what methods an endpoint supports, the only actual use for this currently
    ; is for determining if Asterisk may send connected line UPDATE requests and
    ; MESSAGE requests. Its use may be expanded in the future.
    ;
    ; disallowed_methods = UPDATE
    ;
    ; Note that the TCP and TLS support for chan_sip is currently considered
    ; experimental.  Since it is new, all of the related configuration options are
    ; subject to change in any release.  If they are changed, the changes will
    ; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
    ;
    tcpenable=no                    ; Enable server for incoming TCP connections (default is no)
    tcpbindaddr=0.0.0.0             ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
                                    ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
    ;tlsenable=no                   ; Enable server for incoming TLS (secure) connections (default is no)
    ;tlsbindaddr=0.0.0.0            ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
                                    ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
                                    ; Remember that the IP address must match the common name (hostname) in the
                                    ; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
                                    ; For details how to construct a certificate for SIP see
                                    ; http://tools.ietf.org/html/draft-ietf-sip-domain-certs
    ;tcpauthtimeout = 30            ; tcpauthtimeout specifies the maximum number
                                    ; of seconds a client has to authenticate.  If
                                    ; the client does not authenticate beofre this
                                    ; timeout expires, the client will be
                                    ; disconnected. (default: 30 seconds)
    ;tcpauthlimit = 100             ; tcpauthlimit specifies the maximum number of
                                    ; unauthenticated sessions that will be allowed
                                    ; to connect at any given time. (default: 100)
    transport=udp                   ; Set the default transports.  The order determines the primary default transport.
                                    ; If tcpenable=no and the transport set is tcp, we will fallback to UDP.
    srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                    ; Note: Asterisk only uses the first host
                                    ; in SRV records
                                    ; Disabling DNS SRV lookups disables the
                                    ; ability to place SIP calls based on domain
                                    ; names to some other SIP users on the Internet
                                    ; Specifying a port in a SIP peer definition or
                                    ; when dialing outbound calls will supress SRV
                                    ; lookups for that peer or call.
    ;pedantic=yes                   ; Enable checking of tags in headers,
                                    ; international character conversions in URIs
                                    ; and multiline formatted headers for strict
                                    ; SIP compatibility (defaults to "yes")
    ; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
    ;tos_sip=cs3                    ; Sets TOS for SIP packets.
    ;tos_audio=ef                   ; Sets TOS for RTP audio packets.
    ;tos_video=af41                 ; Sets TOS for RTP video packets.
    ;tos_text=af41                  ; Sets TOS for RTP text packets.
    ;cos_sip=3                      ; Sets 802.1p priority for SIP packets.
    ;cos_audio=5                    ; Sets 802.1p priority for RTP audio packets.
    ;cos_video=4                    ; Sets 802.1p priority for RTP video packets.
    ;cos_text=3                     ; Sets 802.1p priority for RTP text packets.
    ;maxexpiry=3600                 ; Maximum allowed time of incoming registrations
                                    ; and subscriptions (seconds)
    ;minexpiry=60                   ; Minimum length of registrations/subscriptions (default 60)
    ;defaultexpiry=120              ; Default length of incoming/outgoing registration
    ;mwiexpiry=3600                 ; Expiry time for outgoing MWI subscriptions
    ;maxforwards=70                 ; Setting for the SIP Max-Forwards: header (loop prevention)
                                    ; Default value is 70
    ;qualifyfreq=60                 ; Qualification: How often to check for the host to be up in seconds
                                    ; and reported in milliseconds with sip show settings.
                                    ; Set to low value if you use low timeout for NAT of UDP sessions
                                    ; Default: 60
    ;qualifygap=100                 ; Number of milliseconds between each group of peers being qualified
                                    ; Default: 100
    ;qualifypeers=1                 ; Number of peers in a group to be qualified at the same time
                                    ; Default: 1
    ;notifymimetype=text/plain      ; Allow overriding of mime type in MWI NOTIFY
    ;buggymwi=no                    ; Cisco SIP firmware doesn't support the MWI RFC
                                    ; fully. Enable this option to not get error messages
                                    ; when sending MWI to phones with this bug.
    ;mwi_from=asterisk              ; When sending MWI NOTIFY requests, use this setting in
                                    ; the From: header as the "name" portion. Also fill the
                                    ; "user" portion of the URI in the From: header with this
                                    ; value if no fromuser is set
                                    ; Default: empty
    ;vmexten=voicemail              ; dialplan extension to reach mailbox sets the
                                    ; Message-Account in the MWI notify message
                                    ; defaults to "asterisk"
    ; Codec negotiation
    ;
    ; When Asterisk is receiving a call, the codec will initially be set to the
    ; first codec in the allowed codecs defined for the user receiving the call
    ; that the caller also indicates that it supports. But, after the caller
    ; starts sending RTP, Asterisk will switch to using whatever codec the caller
    ; is sending.
    ;
    ; When Asterisk is placing a call, the codec used will be the first codec in
    ; the allowed codecs that the callee indicates that it supports. Asterisk will
    ; *not* switch to whatever codec the callee is sending.
    ;
    ;preferred_codec_only=yes       ; Respond to a SIP invite with the single most preferred codec
                                    ; rather than advertising all joint codec capabilities. This
                                    ; limits the other side's codec choice to exactly what we prefer.
    ;disallow=all                   ; First disallow all codecs
    ;allow=ulaw                     ; Allow codecs in order of preference
    ;allow=ilbc                     ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
                                    ; for framing options
    ;
    ; This option specifies a preference for which music on hold class this channel
    ; should listen to when put on hold if the music class has not been set on the
    ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
    ; channel putting this one on hold did not suggest a music class.
    ;
    ; This option may be specified globally, or on a per-user or per-peer basis.
    ;
    ;mohinterpret=default
    ;
    ; This option specifies which music on hold class to suggest to the peer channel
    ; when this channel places the peer on hold. It may be specified globally or on
    ; a per-user or per-peer basis.
    ;
    ;mohsuggest=default
    ;
    ;parkinglot=plaza               ; Sets the default parking lot for call parking
                                    ; This may also be set for individual users/peers
                                    ; Parkinglots are configured in features.conf
    ;language=en                    ; Default language setting for all users/peers
                                    ; This may also be set for individual users/peers
    ;relaxdtmf=yes                  ; Relax dtmf handling
    ;trustrpid = no                 ; If Remote-Party-ID should be trusted
    ;sendrpid = yes                 ; If Remote-Party-ID should be sent (defaults to no)
    ;sendrpid = rpid                ; Use the "Remote-Party-ID" header
                                    ; to send the identity of the remote party
                                    ; This is identical to sendrpid=yes
    ;sendrpid = pai                 ; Use the "P-Asserted-Identity" header
                                    ; to send the identity of the remote party
    ;rpid_update = no               ; In certain cases, the only method by which a connected line
                                    ; change may be immediately transmitted is with a SIP UPDATE request.
                                    ; If communicating with another Asterisk server, and you wish to be able
                                    ; transmit such UPDATE messages to it, then you must enable this option.
                                    ; Otherwise, we will have to wait until we can send a reinvite to
                                    ; transmit the information.
    ;prematuremedia=no              ; Some ISDN links send empty media frames before
                                    ; the call is in ringing or progress state. The SIP
                                    ; channel will then send 183 indicating early media
                                    ; which will be empty - thus users get no ring signal.
                                    ; Setting this to "yes" will stop any media before we have
                                    ; call progress (meaning the SIP channel will not send 183 Session
                                    ; Progress for early media). Default is "yes". Also make sure that
                                    ; the SIP peer is configured with progressinband=never.
                                    ;
                                    ; In order for "noanswer" applications to work, you need to run
                                    ; the progress() application in the priority before the app.
    ;progressinband=never           ; If we should generate in-band ringing always
                                    ; use 'never' to never use in-band signalling, even in cases
                                    ; where some buggy devices might not render it
                                    ; Valid values: yes, no, never Default: never
    ;useragent=Asterisk PBX         ; Allows you to change the user agent string
                                    ; The default user agent string also contains the Asterisk
                                    ; version. If you don't want to expose this, change the
                                    ; useragent string.
    ;promiscredir = no              ; If yes, allows 302 or REDIR to non-local SIP address
                                    ; Note that promiscredir when redirects are made to the
                                    ; local system will cause loops since Asterisk is incapable
                                    ; of performing a "hairpin" call.
    ;usereqphone = no               ; If yes, ";user=phone" is added to uri that contains
                                    ; a valid phone number
    ;dtmfmode = rfc2833             ; Set default dtmfmode for sending DTMF. Default: rfc2833
                                    ; Other options:
                                    ; info : SIP INFO messages (application/dtmf-relay)
                                    ; shortinfo : SIP INFO messages (application/dtmf)
                                    ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
                                    ; auto : Use rfc2833 if offered, inband otherwise
    ;compactheaders = yes           ; send compact sip headers.
    ;
    ;videosupport=yes               ; Turn on support for SIP video. You need to turn this
                                    ; on in this section to get any video support at all.
                                    ; You can turn it off on a per peer basis if the general
                                    ; video support is enabled, but you can't enable it for
                                    ; one peer only without enabling in the general section.
                                    ; If you set videosupport to "always", then RTP ports will
                                    ; always be set up for video, even on clients that don't
                                    ; support it.  This assists callfile-derived calls and
                                    ; certain transferred calls to use always use video when
                                    ; available. [yes|NO|always]
    ;maxcallbitrate=384             ; Maximum bitrate for video calls (default 384 kb/s)
                                    ; Videosupport and maxcallbitrate is settable
                                    ; for peers and users as well
    ;callevents=no                  ; generate manager events when sip ua
                                    ; performs events (e.g. hold)
    ;authfailureevents=no           ; generate manager "peerstatus" events when peer can't
                                    ; authenticate with Asterisk. Peerstatus will be "rejected".
    ;alwaysauthreject = yes         ; When an incoming INVITE or REGISTER is to be rejected,
                                    ; for any reason, always reject with an identical response
                                    ; equivalent to valid username and invalid password/hash
                                    ; instead of letting the requester know whether there was
                                    ; a matching user or peer for their request.  This reduces
                                    ; the ability of an attacker to scan for valid SIP usernames.
                                    ; This option is set to "yes" by default.
    ;auth_options_requests = yes    ; Enabling this option will authenticate OPTIONS requests just like
                                    ; INVITE requests are.  By default this option is disabled.
    ;g726nonstandard = yes          ; If the peer negotiates G726-32 audio, use AAL2 packing
                                    ; order instead of RFC3551 packing order (this is required
                                    ; for Sipura and Grandstream ATAs, among others). This is
                                    ; contrary to the RFC3551 specification, the peer _should_
                                    ; be negotiating AAL2-G726-32 instead :-(
    ;outboundproxy=proxy.provider.domain            ; send outbound signaling to this proxy, not directly to the devices
    ;outboundproxy=proxy.provider.domain:8080       ; send outbound signaling to this proxy, not directly to the devices
    ;outboundproxy=proxy.provider.domain,force      ; Send ALL outbound signalling to proxy, ignoring route: headers
    ;outboundproxy=tls://proxy.provider.domain      ; same as '=proxy.provider.domain' except we try to connect with tls
    ;outboundproxy=192.0.2.1                        ; IPv4 address literal (default port is 5060)
    ;outboundproxy=2001:db8::1                      ; IPv6 address literal (default port is 5060)
    ;outboundproxy=192.168.0.2.1:5062               ; IPv4 address literal with explicit port
    ;outboundproxy=[2001:db8::1]:5062               ; IPv6 address literal with explicit port
    ;                                               ; (could also be tcp,udp) - defining transports on the proxy line only
    ;                                               ; applies for the global proxy, otherwise use the transport= option
    ;matchexternaddrlocally = yes     ; Only substitute the externaddr or externhost setting if it matches
                                    ; your localnet setting. Unless you have some sort of strange network
                                    ; setup you will not need to enable this.
    ;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from registering
                                    ; as any IP address used for staticly defined
                                    ; hosts.  This helps avoid the configuration
                                    ; error of allowing your users to register at
                                    ; the same address as a SIP provider.
    ;contactdeny=0.0.0.0/0.0.0.0           ; Use contactpermit and contactdeny to
    ;contactpermit=172.16.0.0/255.255.0.0  ; restrict at what IPs your users may
                                           ; register their phones.
    ;engine=asterisk                ; RTP engine to use when communicating with the device
    ;
    ; If regcontext is specified, Asterisk will dynamically create and destroy a
    ; NoOp priority 1 extension for a given peer who registers or unregisters with
    ; us and have a "regexten=" configuration item.
    ; Multiple contexts may be specified by separating them with '&'. The
    ; actual extension is the 'regexten' parameter of the registering peer or its
    ; name if 'regexten' is not provided.  If more than one context is provided,
    ; the context must be specified within regexten by appending the desired
    ; context after '@'.  More than one regexten may be supplied if they are
    ; separated by '&'.  Patterns may be used in regexten.
    ;
    ;regcontext=sipregistrations
    ;regextenonqualify=yes          ; Default "no"
                                    ; If you have qualify on and the peer becomes unreachable
                                    ; this setting will enforce inactivation of the regexten
                                    ; extension for the peer
    ;legacy_useroption_parsing=yes  ; Default "no"      ; If you have this option enabled and there are semicolons
                                                        ; in the user field of a sip URI, the field be truncated
                                                        ; at the first semicolon seen. This effectively makes
                                                        ; semicolon a non-usable character for peer names, extensions,
                                                        ; and maybe other, less tested things.  This can be useful
                                                        ; for improving compatability with devices that like to use
                                                        ; user options for whatever reason.  The behavior is similar to
                                                        ; how SIP URI's were typically handled in 1.6.2, hence the name.
    ; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
    ; in square brackets.  For example, the caller id value 555.5555 becomes 5555555
    ; when this option is enabled.  Disabling this option results in no modification
    ; of the caller id value, which is necessary when the caller id represents something
    ; that must be preserved.  This option can only be used in the [general] section.
    ; By default this option is on.
    ;
    ;shrinkcallerid=yes     ; on by default
    ;use_q850_reason = no ; Default "no"
                          ; Set to yes add Reason header and use Reason header if it is available.
    ;
    ;------------------------ TLS settings ------------------------------------------------------------
    ;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem format only) to use for TLS connections
                                            ; default is to look for "asterisk.pem" in current directory
    ;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem format only) for TLS connections.
                                          ; If no tlsprivatekey is specified, tlscertfile is searched for
                                          ; for both public and private key.
    ;tlscafile=</path/to/certificate>
    ;        If the server your connecting to uses a self signed certificate
    ;        you should have their certificate installed here so the code can
    ;        verify the authenticity of their certificate.
    ;tlscapath=</path/to/ca/dir>
    ;        A directory full of CA certificates.  The files must be named with
    ;        the CA subject name hash value.
    ;        (see man SSL_CTX_load_verify_locations for more info)
    ;tlsdontverifyserver=[yes|no]
    ;        If set to yes, don't verify the servers certificate when acting as
    ;        a client.  If you don't have the server's CA certificate you can
    ;        set this and it will connect without requiring tlscafile to be set.
    ;        Default is no.
    ;tlscipher=<SSL cipher string>
    ;        A string specifying which SSL ciphers to use or not use
    ;        A list of valid SSL cipher strings can be found at:
    ;                http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
    ;
    ;tlsclientmethod=tlsv1     ; values include tlsv1, sslv3, sslv2.
                               ; Specify protocol for outbound client connections.
                               ; If left unspecified, the default is sslv2.
    ;
    ;--------------------------- SIP timers ----------------------------------------------------
    ; These timers are used primarily in INVITE transactions.
    ; The default for Timer T1 is 500 ms or the measured run-trip time between
    ; Asterisk and the device if you have qualify=yes for the device.
    ;
    ;t1min=100                      ; Minimum roundtrip time for messages to monitored hosts
                                    ; Defaults to 100 ms
    ;timert1=500                    ; Default T1 timer
                                    ; Defaults to 500 ms or the measured round-trip
                                    ; time to a peer (qualify=yes).
    ;timerb=32000                   ; Call setup timer. If a provisional response is not received
                                    ; in this amount of time, the call will autocongest
                                    ; Defaults to 64*timert1
    ;--------------------------- RTP timers ----------------------------------------------------
    ; These timers are currently used for both audio and video streams. The RTP timeouts
    ; are only applied to the audio channel.
    ; The settings are settable in the global section as well as per device
    ;
    ;rtptimeout=60                  ; Terminate call if 60 seconds of no RTP or RTCP activity
                                    ; on the audio channel
                                    ; when we're not on hold. This is to be able to hangup
                                    ; a call in the case of a phone disappearing from the net,
                                    ; like a powerloss or grandma tripping over a cable.
    ;rtpholdtimeout=300             ; Terminate call if 300 seconds of no RTP or RTCP activity
                                    ; on the audio channel
                                    ; when we're on hold (must be > rtptimeout)
    ;rtpkeepalive=<secs>            ; Send keepalives in the RTP stream to keep NAT open
                                    ; (default is off - zero)
    ;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
    ; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
    ; This mechanism can detect and reclaim SIP channels that do not terminate through normal
    ; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
    ; The operation of Session-Timers is driven by the following configuration parameters:
    ;
    ; * session-timers    - Session-Timers feature operates in the following three modes:
    ;                            originate : Request and run session-timers always
    ;                            accept    : Run session-timers only when requested by other UA
    ;                            refuse    : Do not run session timers in any case
    ;                       The default mode of operation is 'accept'.
    ; * session-expires   - Maximum session refresh interval in seconds. Defaults to 1800 secs.
    ; * session-minse     - Minimum session refresh interval in seconds. Defualts to 90 secs.
    ; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
    ;
    ;session-timers=originate
    ;session-expires=600
    ;session-minse=90
    ;session-refresher=uas
    ;
    ;--------------------------- SIP DEBUGGING ---------------------------------------------------
    ;sipdebug = yes                 ; Turn on SIP debugging by default, from
                                    ; the moment the channel loads this configuration
    ;recordhistory=yes              ; Record SIP history by default
                                    ; (see sip history / sip no history)
    ;dumphistory=yes                ; Dump SIP history at end of SIP dialogue
                                    ; SIP history is output to the DEBUG logging channel
    ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
    ; You can subscribe to the status of extensions with a "hint" priority
    ; (See extensions.conf.sample for examples)
    ; chan_sip support two major formats for notifications: dialog-info and SIMPLE
    ;
    ; You will get more detailed reports (busy etc) if you have a call counter enabled
    ; for a device.
    ;
    ; If you set the busylevel, we will indicate busy when we have a number of calls that
    ; matches the busylevel treshold.
    ;
    ; For queues, you will need this level of detail in status reporting, regardless
    ; if you use SIP subscriptions. Queues and manager use the same internal interface
    ; for reading status information.
    ;
    ; Note: Subscriptions does not work if you have a realtime dialplan and use the
    ; realtime switch.
    ;
    ;allowsubscribe=no              ; Disable support for subscriptions. (Default is yes)
    ;subscribecontext = default     ; Set a specific context for SUBSCRIBE requests
                                    ; Useful to limit subscriptions to local extensions
                                    ; Settable per peer/user also
    ;notifyringing = no             ; Control whether subscriptions already INUSE get sent
                                    ; RINGING when another call is sent (default: yes)
    ;notifyhold = yes               ; Notify subscriptions on HOLD state (default: no)
                                    ; Turning on notifyringing and notifyhold will add a lot
                                    ; more database transactions if you are using realtime.
    ;notifycid = yes                ; Control whether caller ID information is sent along with
                                    ; dialog-info+xml notifications (supported by snom phones).
                                    ; Note that this feature will only work properly when the
                                    ; incoming call is using the same extension and context that
                                    ; is being used as the hint for the called extension.  This means
                                    ; that it won't work when using subscribecontext for your sip
                                    ; user or peer (if subscribecontext is different than context).
                                    ; This is also limited to a single caller, meaning that if an
                                    ; extension is ringing because multiple calls are incoming,
                                    ; only one will be used as the source of caller ID.  Specify
                                    ; 'ignore-context' to ignore the called context when looking
                                    ; for the caller's channel.  The default value is 'no.' Setting
                                    ; notifycid to 'ignore-context' also causes call-pickups attempted
                                    ; via SNOM's NOTIFY mechanism to set the context for the call pickup
                                    ; to PICKUPMARK.
    ;callcounter = yes              ; Enable call counters on devices. This can be set per
                                    ; device too.
    ;----------------------------------------- T.38 FAX SUPPORT ----------------------------------
    ;
    ; This setting is available in the [general] section as well as in device configurations.
    ; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off.
    ;
    ; t38pt_udptl = yes            ; Enables T.38 with FEC error correction.
    ; t38pt_udptl = yes,fec        ; Enables T.38 with FEC error correction.
    ; t38pt_udptl = yes,redundancy ; Enables T.38 with redundancy error correction.
    ; t38pt_udptl = yes,none       ; Enables T.38 with no error correction.
    ;
    ; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that
    ; is based on an incorrect interpretation of the T.38 recommendation, and results in failures
    ; because Asterisk does not believe it can send T.38 packets of a reasonable size to that
    ; endpoint (Cisco media gateways are one example of this situation). In these cases, during a
    ; T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL
    ; stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you
    ; can set an override (globally, or on a per-device basis) to make Asterisk ignore the
    ; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead.
    ; This can be done by appending 'maxdatagram=<value>' to the t38pt_udptl configuration option,
    ; like this:
    ;
    ; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error correction and overrides
    ;                                       ; the other endpoint's provided value to assume we can
    ;                                       ; send 400 byte T.38 FAX packets to it.
    ;
    ; FAX detection will cause the SIP channel to jump to the 'fax' extension (if it exists)
    ; based one or more events being detected. The events that can be detected are an incoming
    ; CNG tone or an incoming T.38 re-INVITE request.
    ;
    ; faxdetect = yes               ; Default 'no', 'yes' enables both CNG and T.38 detection
    ; faxdetect = cng               ; Enables only CNG detection
    ; faxdetect = t38               ; Enables only T.38 detection
    ;
    ;----------------------------------------- OUTBOUND SIP REGISTRATIONS  ------------------------
    ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
    ; Format for the register statement is:
    ;       register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
    ;
    ;
    ;
    ; domain is either
    ;       - domain in DNS
    ;       - host name in DNS
    ;       - the name of a peer defined below or in realtime
    ; The domain is where you register your username, so your SIP uri you are registering to
    ; is username@domain
    ;
    ; If no extension is given, the 's' extension is used. The extension needs to
    ; be defined in extensions.conf to be able to accept calls from this SIP proxy
    ; (provider).
    ;
    ; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
    ; this is equivalent to having the following line in the general section:
    ;
    ;        register => username:secret@host/callbackextension
    ;
    ; and more readable because you don't have to write the parameters in two places
    ; (note that the "port" is ignored - this is a bug that should be fixed).
    ;
    ; Note that a register= line doesn't mean that we will match the incoming call in any
    ; other way than described above. If you want to control where the call enters your
    ; dialplan, which context, you want to define a peer with the hostname of the provider's
    ; server. If the provider has multiple servers to place calls to your system, you need
    ; a peer for each server.
    ;
    ; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may
    ; contain a port number. Since the logical separator between a host and port number is a
    ; ':' character, and this character is already used to separate between the optional "secret"
    ; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish
    ; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
    ; they are blank. See the third example below for an illustration.
    ;
    ;
    ; Examples:
    ;
    ;register => 1234:password@mysipprovider.com
    ;
    ;     This will pass incoming calls to the 's' extension
    ;
    ;
    ;register => 2345:password@sip_proxy/1234
    ;
    ;    Register 2345 at sip provider 'sip_proxy'.  Calls from this provider
    ;    connect to local extension 1234 in extensions.conf, default context,
    ;    unless you configure a [sip_proxy] section below, and configure a
    ;    context.
    ;    Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
    ;    Tip 2: Use separate inbound and outbound sections for SIP providers
    ;           (instead of type=friend) if you have calls in both directions
    ;
    ;register => 3456@mydomain:5082::@mysipprovider.com
    ;
    ;    Note that in this example, the optional authuser and secret portions have
    ;    been left blank because we have specified a port in the user section
    ;
    ;register => tls://username:xxxxxx@sip-tls-proxy.example.org
    ;
    ;    The 'transport' part defaults to 'udp' but may also be 'tcp' or 'tls'.
    ;    Using 'udp://' explicitly is also useful in case the username part
    ;    contains a '/' ('user/name').
    ;registertimeout=20             ; retry registration calls every 20 seconds (default)
    ;registerattempts=10            ; Number of registration attempts before we give up
                                    ; 0 = continue forever, hammering the other server
                                    ; until it accepts the registration
                                    ; Default is 0 tries, continue forever
    ;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
    ; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
    ; by other phones. At this time, you can only subscribe using UDP as the transport.
    ; Format for the mwi register statement is:
    ;       mwi => user[:secret[:authuser]]@host[:port]/mailbox
    ;
    ; Examples:
    ;mwi => 1234:password@mysipprovider.com/1234
    ;mwi => 1234:password@myportprovider.com:6969/1234
    ;mwi => 1234:password:authuser@myauthprovider.com/1234
    ;mwi => 1234:password:authuser@myauthportprovider.com:6969/1234
    ;
    ; MWI received will be stored in the 1234 mailbox of the SIP_Remote context. It can be used by other phones by following the below:
    ; mailbox=1234@SIP_Remote
    ;----------------------------------------- NAT SUPPORT ------------------------
    ;
    ; WARNING: SIP operation behind a NAT is tricky and you really need
    ; to read and understand well the following section.
    ;
    ; When Asterisk is behind a NAT device, the "local" address (and port) that
    ; a socket is bound to has different values when seen from the inside or
    ; from the outside of the NATted network. Unfortunately this address must
    ; be communicated to the outside (e.g. in SIP and SDP messages), and in
    ; order to determine the correct value Asterisk needs to know:
    ;
    ; + whether it is talking to someone "inside" or "outside" of the NATted network.
    ;   This is configured by assigning the "localnet" parameter with a list
    ;   of network addresses that are considered "inside" of the NATted network.
    ;   IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
    ;   Multiple entries are allowed, e.g. a reasonable set is the following:
    ;
    ;      localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
    ;      localnet=10.0.0.0/255.0.0.0      ; Also RFC1918
    ;      localnet=172.16.0.0/12           ; Another RFC1918 with CIDR notation
    ;      localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
    ;
    ; + the "externally visible" address and port number to be used when talking
    ;   to a host outside the NAT. This information is derived by one of the
    ;   following (mutually exclusive) config file parameters:
    ;
    ;   a. "externaddr = hostname[:port]" specifies a static address[:port] to
    ;      be used in SIP and SDP messages.
    ;      The hostname is looked up only once, when [re]loading sip.conf .
    ;      If a port number is not present, use the port specified in the "udpbindaddr"
    ;      (which is not guaranteed to work correctly, because a NAT box might remap the
    ;      port number as well as the address).
    ;      This approach can be useful if you have a NAT device where you can
    ;      configure the mapping statically. Examples:
    ;
    ;        externaddr = 12.34.56.78          ; use this address.
    ;        externaddr = 12.34.56.78:9900     ; use this address and port.
    ;        externaddr = mynat.my.org:12600   ; Public address of my nat box.
    ;        externtcpport = 9900   ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT.
    ;                               ; externtcpport will default to the externaddr or externhost port if either one is set.
    ;        externtlsport = 12600  ; The externally mapped tls port, when Asterisk is behind a static NAT or PAT.
    ;                               ; externtlsport port will default to the RFC designated port of 5061.
    ;
    ;   b. "externhost = hostname[:port]" is similar to "externaddr" except
    ;      that the hostname is looked up every "externrefresh" seconds
    ;      (default 10s). This can be useful when your NAT device lets you choose
    ;      the port mapping, but the IP address is dynamic.
    ;      Beware, you might suffer from service disruption when the name server
    ;      resolution fails. Examples:
    ;
    ;        externhost=foo.dyndns.net       ; refreshed periodically
    ;        externrefresh=180               ; change the refresh interval
    ;
    ;   Note that at the moment all these mechanism work only for the SIP socket.
    ;   The IP address discovered with externaddr/externhost is reused for
    ;   media sessions as well, but the port numbers are not remapped so you
    ;   may still experience problems.
    ;
    ; NOTE 1: in some cases, NAT boxes will use different port numbers in
    ; the internal<->external mapping. In these cases, the "externaddr" and
    ; "externhost" might not help you configure addresses properly.
    ;
    ; NOTE 2: when using "externaddr" or "externhost", the address part is
    ; also used as the external address for media sessions. Thus, the port
    ; information in the SDP may be wrong!
    ;
    ; In addition to the above, Asterisk has an additional "nat" parameter to
    ; address NAT-related issues in incoming SIP or media sessions.
    ; In particular, depending on the 'nat= ' settings described below, Asterisk
    ; may override the address/port information specified in the SIP/SDP messages,
    ; and use the information (sender address) supplied by the network stack instead.
    ; However, this is only useful if the external traffic can reach us.
    ; The following settings are allowed (both globally and in individual sections):
    ;
    ;        nat = no                ; Use rport if the remote side says to use it.
    ;        nat = force_rport       ; Force rport to always be on. (default)
    ;        nat = yes               ; Force rport to always be on and perform comedia RTP handling.
    ;        nat = comedia           ; Use rport if the remote side says to use it and perform comedia RTP handling.
    ;
    ; 'comedia RTP handling' refers to the technique of sending RTP to the port that the
    ; the other endpoint's RTP arrived from, and means 'connection-oriented media'. This is
    ; only partially related to RFC 4145 which was referred to as COMEDIA while it was in
    ; draft form. This method is used to accomodate endpoints that may be located behind
    ; NAT devices, and as such the port number they tell Asterisk to send RTP packets to
    ; for their media streams is not actual port number that will be used on the nearer
    ; side of the NAT.
    ;
    ; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
    ; the nat setting in a peer definition, then the peer username will be discoverable
    ; by outside parties as Asterisk will respond to different ports for defined and
    ; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
    ; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the
    ; other, then valid peers with settings differing from those in the general section will
    ; be discoverable.
    ;
    ; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by
    ; RFC 4961; Asterisk will always send RTP packets from the same port number it expects
    ; to receive them on.
    ;
    ; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using
    ; the media_address configuration option. This is only applicable to the general section and
    ; can not be set per-user or per-peer.
    ;
    ; media_address = 172.16.42.1
    ;
    ; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the
    ; perceived external network address has changed.  When the stun_monitor is installed and
    ; configured, chan_sip will renew all outbound registrations when the monitor detects any sort
    ; of network change has occurred. By default this option is enabled, but only takes effect once
    ; res_stun_monitor is configured.  If res_stun_monitor is enabled and you wish to not
    ; generate all outbound registrations on a network change, use the option below to disable
    ; this feature.
    ;
    ; subscribe_network_change_event = yes ; on by default
    ;----------------------------------- MEDIA HANDLING --------------------------------
    ; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
    ; no reason for Asterisk to stay in the media path, the media will be redirected.
    ; This does not really work well in the case where Asterisk is outside and the
    ; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
    ;
    ;directmedia=yes                ; Asterisk by default tries to redirect the
                                    ; RTP media stream to go directly from
                                    ; the caller to the callee.  Some devices do not
                                    ; support this (especially if one of them is behind a NAT).
                                    ; The default setting is YES. If you have all clients
                                    ; behind a NAT, or for some other reason want Asterisk to
                                    ; stay in the audio path, you may want to turn this off.
                                    ; This setting also affect direct RTP
                                    ; at call setup (a new feature in 1.4 - setting up the
                                    ; call directly between the endpoints instead of sending
                                    ; a re-INVITE).
                                    ; Additionally this option does not disable all reINVITE operations.
                                    ; It only controls Asterisk generating reINVITEs for the specific
                                    ; purpose of setting up a direct media path. If a reINVITE is
                                    ; needed to switch a media stream to inactive (when placed on
                                    ; hold) or to T.38, it will still be done, regardless of this
                                    ; setting. Note that direct T.38 is not supported.
    ;directmedia=nonat              ; An additional option is to allow media path redirection
                                    ; (reinvite) but only when the peer where the media is being
                                    ; sent is known to not be behind a NAT (as the RTP core can
                                    ; determine it based on the apparent IP address the media
                                    ; arrives from).
    ;directmedia=update             ; Yet a third option... use UPDATE for media path redirection,
                                    ; instead of INVITE. This can be combined with 'nonat', as
                                    ; 'directmedia=update,nonat'. It implies 'yes'.
    ;directrtpsetup=yes             ; Enable the new experimental direct RTP setup. This sets up
                                    ; the call directly with media peer-2-peer without re-invites.
                                    ; Will not work for video and cases where the callee sends
                                    ; RTP payloads and fmtp headers in the 200 OK that does not match the
                                    ; callers INVITE. This will also fail if directmedia is enabled when
                                    ; the device is actually behind NAT.
    ;directmediadeny=0.0.0.0/0      ; Use directmediapermit and directmediadeny to restrict
    ;directmediapermit=172.16.0.0/16; which peers should be able to pass directmedia to each other
                                    ; (There is no default setting, this is just an example)
                                    ; Use this if some of your phones are on IP addresses that
                                    ; can not reach each other directly. This way you can force
                                    ; RTP to always flow through asterisk in such cases.
    ;ignoresdpversion=yes           ; By default, Asterisk will honor the session version
                                    ; number in SDP packets and will only modify the SDP
                                    ; session if the version number changes. This option will
                                    ; force asterisk to ignore the SDP session version number
                                    ; and treat all SDP data as new data.  This is required
                                    ; for devices that send us non standard SDP packets
                                    ; (observed with Microsoft OCS). By default this option is
                                    ; off.
    ;sdpsession=Asterisk PBX        ; Allows you to change the SDP session name string, (s=)
                                    ; Like the useragent parameter, the default user agent string
                                    ; also contains the Asterisk version.
    ;sdpowner=root                  ; Allows you to change the username field in the SDP owner string, (o=)
                                    ; This field MUST NOT contain spaces
    ;encryption=no                  ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
                                    ; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
                                    ; the peer does not support SRTP. Defaults to no.
    ;----------------------------------------- REALTIME SUPPORT ------------------------
    ; For additional information on ARA, the Asterisk Realtime Architecture,
    ; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
    ;
    ;rtcachefriends=yes             ; Cache realtime friends by adding them to the internal list
                                    ; just like friends added from the config file only on a
                                    ; as-needed basis? (yes|no)
    ;rtsavesysname=yes              ; Save systemname in realtime database at registration
                                    ; Default= no
    ;rtupdate=yes                   ; Send registry updates to database using realtime? (yes|no)
                                    ; If set to yes, when a SIP UA registers successfully, the ip address,
                                    ; the origination port, the registration period, and the username of
                                    ; the UA will be set to database via realtime.
                                    ; If not present, defaults to 'yes'. Note: realtime peers will
                                    ; probably not function across reloads in the way that you expect, if
                                    ; you turn this option off.
    ;rtautoclear=yes                ; Auto-Expire friends created on the fly on the same schedule
                                    ; as if it had just registered? (yes|no|<seconds>)
                                    ; If set to yes, when the registration expires, the friend will
                                    ; vanish from the configuration until requested again. If set
                                    ; to an integer, friends expire within this number of seconds
                                    ; instead of the registration interval.
    ;ignoreregexpire=yes            ; Enabling this setting has two functions:
                                    ;
                                    ; For non-realtime peers, when their registration expires, the
                                    ; information will _not_ be removed from memory or the Asterisk database
                                    ; if you attempt to place a call to the peer, the existing information
                                    ; will be used in spite of it having expired
                                    ;
                                    ; For realtime peers, when the peer is retrieved from realtime storage,
                                    ; the registration information will be used regardless of whether
                                    ; it has expired or not; if it expires while the realtime peer
                                    ; is still in memory (due to caching or other reasons), the
                                    ; information will not be removed from realtime storage
    ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
    ; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
    ; domains, each of which can direct the call to a specific context if desired.
    ; By default, all domains are accepted and sent to the default context or the
    ; context associated with the user/peer placing the call.
    ; REGISTER to non-local domains will be automatically denied if a domain
    ; list is configured.
    ;
    ; Domains can be specified using:
    ; domain=<domain>[,<context>]
    ; Examples:
    ; domain=myasterisk.dom
    ; domain=customer.com,customer-context
    ;
    ; In addition, all the 'default' domains associated with a server should be
    ; added if incoming request filtering is desired.
    ; autodomain=yes
    ;
    ; To disallow requests for domains not serviced by this server:
    ; allowexternaldomains=no
    ;domain=mydomain.tld,mydomain-incoming
                                    ; Add domain and configure incoming context
                                    ; for external calls to this domain
    ;domain=1.2.3.4                 ; Add IP address as local domain
                                    ; You can have several "domain" settings
    ;allowexternaldomains=no        ; Disable INVITE and REFER to non-local domains
                                    ; Default is yes
    ;autodomain=yes                 ; Turn this on to have Asterisk add local host
                                    ; name and local IP to domain list.
    ; fromdomain=mydomain.tld       ; When making outbound SIP INVITEs to
                                    ; non-peers, use your primary domain "identity"
                                    ; for From: headers instead of just your IP
                                    ; address. This is to be polite and
                                    ; it may be a mandatory requirement for some
                                    ; destinations which do not have a prior
                                    ; account relationship with your server.
    ;------------------------------ Advice of Charge CONFIGURATION --------------------------
    ; snom_aoc_enabled = yes;     ; This options turns on and off support for sending AOC-D and
                                  ; AOC-E to snom endpoints.  This option can be used both in the
                                  ; peer and global scope.  The default for this option is off.
    ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
    ; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
                                  ; SIP channel. Defaults to "no". An enabled jitterbuffer will
                                  ; be used only if the sending side can create and the receiving
                                  ; side can not accept jitter. The SIP channel can accept jitter,
                                  ; thus a jitterbuffer on the receive SIP side will be used only
                                  ; if it is forced and enabled.
    ; jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a SIP
                                  ; channel. Defaults to "no".
    ; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.
    ; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
                                  ; resynchronized. Useful to improve the quality of the voice, with
                                  ; big jumps in/broken timestamps, usually sent from exotic devices
                                  ; and programs. Defaults to 1000.
    ; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a SIP
                                  ; channel. Two implementations are currently available - "fixed"
                                  ; (with size always equals to jbmaxsize) and "adaptive" (with
                                  ; variable size, actually the new jb of IAX2). Defaults to fixed.
    ; jbtargetextra = 40          ; This option only affects the jb when 'jbimpl = adaptive' is set.
                                  ; The option represents the number of milliseconds by which the new jitter buffer
                                  ; will pad its size. the default is 40, so without modification, the new
                                  ; jitter buffer will set its size to the jitter value plus 40 milliseconds.
                                  ; increasing this value may help if your network normally has low jitter,
                                  ; but occasionally has spikes.
    ; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
    ;----------------------------- SIP_CAUSE reporting ---------------------------------
    ; storesipcause = no          ; This option causes chan_sip to set the
                                  ; HASH(SIP_CAUSE,<channel name>) channel variable
                                  ; to the value of the last sip response.
                                  ; WARNING: enabling this option carries a
                                  ; significant performance burden. It should only
                                  ; be used in low call volume situations. This
                                  ; option defaults to "no".
    ;-----------------------------------------------------------------------------------
    [authentication]
    ; Global credentials for outbound calls, i.e. when a proxy challenges your
    ; Asterisk server for authentication. These credentials override
    ; any credentials in peer/register definition if realm is matched.
    ;
    ; This way, Asterisk can authenticate for outbound calls to other
    ; realms. We match realm on the proxy challenge and pick an set of
    ; credentials from this list
    ; Syntax:
    ;        auth = <user>:<secret>@<realm>
    ;        auth = <user>#<md5secret>@<realm>
    ; Example:
    ;auth=mark:topsecret@digium.com
    ;
    ; You may also add auth= statements to [peer] definitions
    ; Peer auth= override all other authentication settings if we match on realm
    ;------------------------------------------------------------------------------
    ; DEVICE CONFIGURATION
    ;
    ; SIP entities have a 'type' which determines their roles within Asterisk.
    ; * For entities with 'type=peer':
    ;   Peers handle both inbound and outbound calls and are matched by ip/port, so for
    ;   The case of incoming calls from the peer, the IP address must match in order for
    ;   The invitation to work. This means calls made from either direction won't work if
    ;   The peer is unregistered while host=dynamic or if the host is otherise not set to
    ;   the correct IP of the sender.
    ; * For entities with 'type=user':
    ;   Asterisk users handle inbound calls only (meaning they call Asterisk, Asterisk can't
    ;   call them) and are matched by their authorization information (authname and secret).
    ;   Asterisk doesn't rely on their IP and will accept calls regardless of the host setting
    ;   as long as the incoming SIP invite authorizes successfully.
    ; * For entities with 'type=friend':
    ;   Asterisk will create the entity as both a friend and a peer. Asterisk will accept
    ;   calls from friends like it would for users, requiring only that the authorization
    ;   matches rather than the IP address. Since it is also a peer, a friend entity can
    ;   be called as long as its IP is known to Asterisk. In the case of host=dynamic,
    ;   this means it is necessary for the entity to register before Asterisk can call it.
    ;
    ; Use remotesecret for outbound authentication, and secret for authenticating
    ; inbound requests. For historical reasons, if no remotesecret is supplied for an
    ; outbound registration or call, the secret will be used.
    ;
    ; For device names, we recommend using only a-z, numerics (0-9) and underscore
    ;
    ; For local phones, type=friend works most of the time
    ;
    ; If you have one-way audio, you probably have NAT problems.
    ; If Asterisk is on a public IP, and the phone is inside of a NAT device
    ; you will need to configure nat option for those phones.
    ; Also, turn on qualify=yes to keep the nat session open
    ;
    ; Configuration options available
    ; --------------------
    ; context
    ; callingpres
    ; permit
    ; deny
    ; secret
    ; md5secret
    ; remotesecret
    ; transport
    ; dtmfmode
    ; directmedia
    ; nat
    ; callgroup
    ; pickupgroup
    ; language
    ; allow
    ; disallow
    ; insecure
    ; trustrpid
    ; progressinband
    ; promiscredir
    ; useclientcode
    ; accountcode
    ; setvar
    ; callerid
    ; amaflags
    ; callcounter
    ; busylevel
    ; allowoverlap
    ; allowsubscribe
    ; allowtransfer
    ; ignoresdpversion
    ; subscribecontext
    ; template
    ; videosupport
    ; maxcallbitrate
    ; rfc2833compensate
    ; mailbox
    ; session-timers
    ; session-expires
    ; session-minse
    ; session-refresher
    ; t38pt_usertpsource
    ; regexten
    ; fromdomain
    ; fromuser
    ; host
    ; port
    ; qualify
    ; defaultip
    ; defaultuser
    ; rtptimeout
    ; rtpholdtimeout
    ; sendrpid
    ; outboundproxy
    ; rfc2833compensate
    ; callbackextension
    ; registertrying
    ; timert1
    ; timerb
    ; qualifyfreq
    ; t38pt_usertpsource
    ; contactpermit         ; Limit what a host may register as (a neat trick
    ; contactdeny           ; is to register at the same IP as a SIP provider,
    ;                       ; then call oneself, and get redirected to that
    ;                       ; same location).
    ; directmediapermit
    ; directmediadeny
    ; unsolicited_mailbox
    ; use_q850_reason
    ; maxforwards
    ; encryption
    ;[sip_proxy]
    ; For incoming calls only. Example: FWD (Free World Dialup)
    ; We match on IP address of the proxy for incoming calls
    ; since we can not match on username (caller id)
    ;type=peer
    ;context=from-fwd
    ;host=fwd.pulver.com
    ;[sip_proxy-out]
    ;type=peer                        ; we only want to call out, not be called
    ;remotesecret=guessit             ; Our password to their service
    ;defaultuser=yourusername         ; Authentication user for outbound proxies
    ;fromuser=yourusername            ; Many SIP providers require this!
    ;fromdomain=provider.sip.domain
    ;host=box.provider.com
    ;transport=udp,tcp                ; This sets the default transport type to udp for outgoing, and will
    ;                                 ; accept both tcp and udp. The default transport type is only used for
    ;                                 ; outbound messages until a Registration takes place.  During the
    ;                                 ; peer Registration the transport type may change to another supported
    ;                                 ; type if the peer requests so.
    ;usereqphone=yes                  ; This provider requires ";user=phone" on URI
    ;callcounter=yes                  ; Enable call counter
    ;busylevel=2                      ; Signal busy at 2 or more calls
    ;outboundproxy=proxy.provider.domain  ; send outbound signaling to this proxy, not directly to the peer
    ;port=80                          ; The port number we want to connect to on the remote side
                                      ; Also used as "defaultport" in combination with "defaultip" settings
    ;--- sample definition for a provider
    ;[provider1]
    ;type=peer
    ;host=sip.provider1.com
    ;fromuser=4015552299              ; how your provider knows you
    ;remotesecret=youwillneverguessit ; The password we use to authenticate to them
    ;secret=gissadetdu                ; The password they use to contact us
    ;callbackextension=123            ; Register with this server and require calls coming back to this extension
    ;transport=udp,tcp                ; This sets the transport type to udp for outgoing, and will
    ;                                 ;   accept both tcp and udp. Default is udp. The first transport
    ;                                 ;   listed will always be used for outgoing connections.
    ;unsolicited_mailbox=4015552299   ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old
    ;                                 ;   message count will be stored in the configured virtual mailbox. It can be used
    ;                                 ;   by any device supporting MWI by specifying <configured value>@SIP_Remote as the
    ;                                 ;   mailbox.
    ;
    ; Because you might have a large number of similar sections, it is generally
    ; convenient to use templates for the common parameters, and add them
    ; the the various sections. Examples are below, and we can even leave
    ; the templates uncommented as they will not harm:
    [basic-options](!)                ; a template
            dtmfmode=rfc2833
            context=from-office
            type=friend
    [natted-phone](!,basic-options)   ; another template inheriting basic-options
            directmedia=no
            host=dynamic
    [public-phone](!,basic-options)   ; another template inheriting basic-options
            directmedia=yes
    [my-codecs](!)                    ; a template for my preferred codecs
            disallow=all
            allow=ilbc
            allow=g729
            allow=gsm
            allow=g723
            allow=ulaw
    [ulaw-phone](!)                   ; and another one for ulaw-only
            disallow=all
            allow=ulaw
    ; and finally instantiate a few phones
    ;
    ; [2133](natted-phone,my-codecs)
    ;        secret = peekaboo
    ; [2134](natted-phone,ulaw-phone)
    ;        secret = not_very_secret
    ; [2136](public-phone,ulaw-phone)
    ;        secret = not_very_secret_either
    ; ...
    ;
    ; Standard configurations not using templates look like this:
    ;
    ;[grandstream1]
    ;type=friend
    ;context=from-sip                ; Where to start in the dialplan when this phone calls
    ;callerid=John Doe <1234>        ; Full caller ID, to override the phones config
                                     ; on incoming calls to Asterisk
    ;host=192.168.0.23               ; we have a static but private IP address
                                     ; No registration allowed
    ;directmedia=yes                 ; allow RTP voice traffic to bypass Asterisk
    ;dtmfmode=info                   ; either RFC2833 or INFO for the BudgeTone
    ;call-limit=1                    ; permit only 1 outgoing call and 1 incoming call at a time
                                     ; from the phone to asterisk (deprecated)
                                     ; 1 for the explicit peer, 1 for the explicit user,
                                     ; remember that a friend equals 1 peer and 1 user in
                                     ; memory
                                     ; There is no combined call counter for a "friend"
                                     ; so there's currently no way in sip.conf to limit
                                     ; to one inbound or outbound call per phone. Use
                                     ; the group counters in the dial plan for that.
                                     ;
    ;mailbox=1234@default            ; mailbox 1234 in voicemail context "default"
    ;disallow=all                    ; need to disallow=all before we can use allow=
    ;allow=ulaw                      ; Note: In user sections the order of codecs
                                     ; listed with allow= does NOT matter!
    ;allow=alaw
    ;allow=g723.1                    ; Asterisk only supports g723.1 pass-thru!
    ;allow=g729                      ; Pass-thru only unless g729 license obtained
    ;callingpres=allowed_passed_screen ; Set caller ID presentation
                                     ; See README.callingpres for more information
    ;[xlite1]
    ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
    ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
    ;type=friend
    ;regexten=1234                   ; When they register, create extension 1234
    ;callerid="Jane Smith" <5678>
    ;host=dynamic                    ; This device needs to register
    ;directmedia=no                  ; Typically set to NO if behind NAT
    ;disallow=all
    ;allow=gsm                       ; GSM consumes far less bandwidth than ulaw
    ;allow=ulaw
    ;allow=alaw
    ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
    ;registertrying=yes              ; Send a 100 Trying when the device registers.
    ;[snom]
    ;type=friend                     ; Friends place calls and receive calls
    ;context=from-sip                ; Context for incoming calls from this user
    ;secret=blah
    ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
    ;language=de                     ; Use German prompts for this user
    ;host=dynamic                    ; This peer register with us
    ;dtmfmode=inband                 ; Choices are inband, rfc2833, or info
    ;defaultip=192.168.0.59          ; IP used until peer registers
    ;mailbox=1234@context,2345       ; Mailbox(-es) for message waiting indicator
    ;subscribemwi=yes                ; Only send notifications if this phone
                                     ; subscribes for mailbox notification
    ;vmexten=voicemail               ; dialplan extension to reach mailbox
                                     ; sets the Message-Account in the MWI notify message
                                     ; defaults to global vmexten which defaults to "asterisk"
    ;disallow=all
    ;allow=ulaw                      ; dtmfmode=inband only works with ulaw or alaw!
    ;[polycom]
    ;type=friend                     ; Friends place calls and receive calls
    ;context=from-sip                ; Context for incoming calls from this user
    ;secret=blahpoly
    ;host=dynamic                    ; This peer register with us
    ;dtmfmode=rfc2833                ; Choices are inband, rfc2833, or info
    ;defaultuser=polly               ; Username to use in INVITE until peer registers
    ;defaultip=192.168.40.123
                                     ; Normally you do NOT need to set this parameter
    ;disallow=all
    ;allow=ulaw                      ; dtmfmode=inband only works with ulaw or alaw!
    ;progressinband=no               ; Polycom phones don't work properly with "never"
    ;[pingtel]
    ;type=friend
    ;secret=blah
    ;host=dynamic
    ;insecure=port                   ; Allow matching of peer by IP address without
                                     ; matching port number
    ;insecure=invite                 ; Do not require authentication of incoming INVITEs
    ;insecure=port,invite            ; (both)
    ;qualify=1000                    ; Consider it down if it's 1 second to reply
                                     ; Helps with NAT session
                                     ; qualify=yes uses default value
    ;qualifyfreq=60                  ; Qualification: How often to check for the
                                     ; host to be up in seconds
                                     ; Set to low value if you use low timeout for
                                     ; NAT of UDP sessions
    ;
    ; Call group and Pickup group should be in the range from 0 to 63
    ;
    ;callgroup=1,3-4                 ; We are in caller groups 1,3,4
    ;pickupgroup=1,3-5               ; We can do call pick-p for call group 1,3,4,5
    ;defaultip=192.168.0.60          ; IP address to use if peer has not registered
    ;deny=0.0.0.0/0.0.0.0            ; ACL: Control access to this account based on IP address
    ;permit=192.168.0.60/255.255.255.0
    ;permit=192.168.0.60/24          ; we can also use CIDR notation for subnet masks
    ;permit=2001:db8::/32            ; IPv6 ACLs can be specified if desired. IPv6 ACLs
                                     ; apply only to IPv6 addresses, and IPv4 ACLs apply
                                     ; only to IPv4 addresses.
    ;[cisco1]
    ;type=friend
    ;secret=blah
    ;qualify=200                     ; Qualify peer is no more than 200ms away
    ;host=dynamic                    ; This device registers with us
    ;directmedia=no                  ; Asterisk by default tries to redirect the
                                     ; RTP media stream (audio) to go directly from
                                     ; the caller to the callee.  Some devices do not
                                     ; support this (especially if one of them is
                                     ; behind a NAT).
    ;defaultip=192.168.0.4           ; IP address to use until registration
    ;defaultuser=goran               ; Username to use when calling this device before registration
                                     ; Normally you do NOT need to set this parameter
    ;setvar=CUSTID=5678              ; Channel variable to be set for all calls from or to this device
    ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep   ; This channel variable will
                                                    ; cause the given audio file to
                                                    ; be played upon completion of
                                                    ; an attended transfer.
    ;[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.
    
    /etc/default/isc-dhcp-server
    >
    # Defaults for isc-dhcp-server initscript
    # sourced by /etc/init.d/isc-dhcp-server
    # installed at /etc/default/isc-dhcp-server by the maintainer scripts
    #
    # This is a POSIX shell fragment
    #
    # Path to dhcpd's config file (default: /etc/dhcp/dhcpd.conf).
    #DHCPD_CONF=/etc/dhcp/dhcpd.conf
    # Path to dhcpd's PID file (default: /var/run/dhcpd.pid).
    #DHCPD_PID=/var/run/dhcpd.pid
    # Additional options to start dhcpd with.
    #       Don't use options -cf or -pf here; use DHCPD_CONF/ DHCPD_PID instead
    #OPTIONS=""
    # On what interfaces should the DHCP server (dhcpd) serve DHCP requests?
    #       Separate multiple interfaces with spaces, e.g. "eth0 eth1".
    INTERFACES=""
    
    /etc/dhcp/dhcpd.conf
    >
    #
    # Sample configuration file for ISC dhcpd for Debian
    #
    #
    # The ddns-updates-style parameter controls whether or not the server will
    # attempt to do a DNS update when a lease is confirmed. We default to the
    # behavior of the version 2 packages ('none', since DHCP v2 didn't
    # have support for DDNS.)
    ddns-update-style none;
    # option definitions common to all supported networks...
    option domain-name "unix.nt";
    option domain-name-servers 192.168.12.254;
    default-lease-time 600;
    max-lease-time 7200;
    log-facility local7;
    subnet 192.168.80.0 netmask 255.255.255.0 {
      range 192.168.80.100 192.168.80.200;
      option routers 192.168.80.254;
    }
    
    /etc/init.d/isc-dhcp-server
    >
    #!/bin/sh
    #
    #
    ### BEGIN INIT INFO
    # Provides:          isc-dhcp-server
    # Required-Start:    $remote_fs $network $syslog
    # Required-Stop:     $remote_fs $network $syslog
    # Should-Start:      $local_fs slapd $named
    # Should-Stop:       $local_fs slapd
    # Default-Start:     2 3 4 5
    # Default-Stop:      0 1 6
    # Short-Description: DHCP server
    # Description:       Dynamic Host Configuration Protocol Server
    ### END INIT INFO
    PATH=/sbin:/bin:/usr/sbin:/usr/bin
    test -f /usr/sbin/dhcpd || exit 0
    DHCPD_DEFAULT="${DHCPD_DEFAULT:-/etc/default/isc-dhcp-server}"
    # It is not safe to start if we don't have a default configuration...
    if [ ! -f "$DHCPD_DEFAULT" ]; then
            echo "$DHCPD_DEFAULT does not exist! - Aborting..."
            if [ "$DHCPD_DEFAULT" = "/etc/default/isc-dhcp-server" ]; then
                    echo "Run 'dpkg-reconfigure isc-dhcp-server' to fix the problem."
            fi
            exit 0
    fi
    . /lib/lsb/init-functions
    # Read init script configuration
    [ -f "$DHCPD_DEFAULT" ] && . "$DHCPD_DEFAULT"
    NAME=dhcpd
    DESC="ISC DHCP server"
    # fallback to default config file
    DHCPD_CONF=${DHCPD_CONF:-/etc/dhcp/dhcpd.conf}
    # try to read pid file name from config file, with fallback to /var/run/dhcpd.pid
    if [ -z "$DHCPD_PID" ]; then
            DHCPD_PID=$(sed -n -e 's/^[ \t]*pid-file-name[ \t]*"(.*)"[ \t]*;.*$/\1/p' < "$DHCPD_CONF" 2>/dev/null | head -n 1)
    fi
    DHCPD_PID="${DHCPD_PID:-/var/run/dhcpd.pid}"
    test_config()
    {
            if ! /usr/sbin/dhcpd -t $OPTIONS -q -cf "$DHCPD_CONF" > /dev/null 2>&1; then
                    echo "dhcpd self-test failed. Please fix $DHCPD_CONF."
                    echo "The error was: "
                    /usr/sbin/dhcpd -t $OPTIONS -cf "$DHCPD_CONF"
                    exit 1
            fi
    }
    # single arg is -v for messages, -q for none
    check_status()
    {
        if [ ! -r "$DHCPD_PID" ]; then
            test "$1" != -v || echo "$NAME is not running."
            return 3
        fi
        if read pid < "$DHCPD_PID" && ps -p "$pid" > /dev/null 2>&1; then
            test "$1" != -v || echo "$NAME is running."
            return 0
        else
            test "$1" != -v || echo "$NAME is not running but $DHCPD_PID exists."
            return 1
        fi
    }
    case "$1" in
            start)
                    test_config
                    log_daemon_msg "Starting $DESC" "$NAME"
                    start-stop-daemon --start --quiet --pidfile "$DHCPD_PID" \
                            --exec /usr/sbin/dhcpd -- \
                            -q $OPTIONS -cf "$DHCPD_CONF" -pf "$DHCPD_PID" $INTERFACES
                    sleep 2
                    if check_status -q; then
                            log_end_msg 0
                    else
                            log_failure_msg "check syslog for diagnostics."
                            log_end_msg 1
                            exit 1
                    fi
                    ;;
            stop)
                    log_daemon_msg "Stopping $DESC" "$NAME"
                    start-stop-daemon --stop --quiet --pidfile "$DHCPD_PID"
                    log_end_msg $?
                    rm -f "$DHCPD_PID"
                    ;;
            restart | force-reload)
                    test_config
                    $0 stop
                    sleep 2
                    $0 start
                    if [ "$?" != "0" ]; then
                            exit 1
                    fi
                    ;;
            status)
                    echo -n "Status of $DESC: "
                    check_status -v
                    exit "$?"
                    ;;
            *)
                    echo "Usage: $0 {start|stop|restart|force-reload|status}"
                    exit 1
    esac
    exit 0
    
    /var/lib/dhcp/dhclient.leases
    >
    /var/lib/dhcp/dhcpd.leases
    >
    # The format of this file is documented in the dhcpd.leases(5) manual page.
    # This lease file was written by isc-dhcp-4.2.2
    server-duid "\000\001\000\001\034\005\352p\000\"M\202\371z";
    lease 192.168.80.100 {
      starts 1 2014/11/24 13:09:20;
      ends 1 2014/11/24 13:19:20;
      cltt 1 2014/11/24 13:09:20;
      binding state active;
      next binding state free;
      rewind binding state free;
      hardware ethernet 00:19:56:fd:74:89;
      uid "\001\000\031V\375t\211";
      client-hostname "SEP001956FD7489";
    }
    lease 192.168.80.101 {
      starts 1 2014/11/24 13:10:46;
      ends 1 2014/11/24 13:20:46;
      cltt 1 2014/11/24 13:10:46;
      binding state active;
      next binding state free;
      rewind binding state free;
      hardware ethernet 00:19:e3:37:46:a0;
      uid "\001\000\031\3437F\240";
      client-hostname "natnet";
    }
    lease 192.168.80.100 {
      starts 1 2014/11/24 13:14:21;
      ends 1 2014/11/24 13:24:21;
      cltt 1 2014/11/24 13:14:21;
      binding state active;
      next binding state free;
      rewind binding state free;
      hardware ethernet 00:19:56:fd:74:89;
      uid "\001\000\031V\375t\211";
      client-hostname "SEP001956FD7489";
    }
    lease 192.168.80.101 {
      starts 1 2014/11/24 13:15:47;
      ends 1 2014/11/24 15:15:47;
      cltt 1 2014/11/24 13:15:47;
      binding state active;
      next binding state free;
      rewind binding state free;
      hardware ethernet 00:19:e3:37:46:a0;
      uid "\001\000\031\3437F\240";
      client-hostname "natnet";
    }
    lease 192.168.80.100 {
      starts 1 2014/11/24 13:19:21;
      ends 1 2014/11/24 13:29:21;
      cltt 1 2014/11/24 13:19:21;
      binding state active;
      next binding state free;
      rewind binding state free;
      hardware ethernet 00:19:56:fd:74:89;
      uid "\001\000\031V\375t\211";
      client-hostname "SEP001956FD7489";
    }
    lease 192.168.80.100 {
      starts 1 2014/11/24 13:24:22;
      ends 1 2014/11/24 13:34:22;
      cltt 1 2014/11/24 13:24:22;
      binding state active;
      next binding state free;
      rewind binding state free;
      hardware ethernet 00:19:56:fd:74:89;
      uid "\001\000\031V\375t\211";
      client-hostname "SEP001956FD7489";
    }
    

    Статистика

    Время первой команды журнала09:08:50 2014-11-24
    Время последней команды журнала12:38:38 2014-11-25
    Количество командных строк в журнале100
    Процент команд с ненулевым кодом завершения, %12.00
    Процент синтаксически неверно набранных команд, % 0.00
    Суммарное время работы с терминалом *, час 4.76
    Количество командных строк в единицу времени, команда/мин 0.35
    Частота использования команд
    ls15|=============| 13.76%
    cat11|==========| 10.09%
    vi11|==========| 10.09%
    screen8|=======| 7.34%
    ping8|=======| 7.34%
    grep6|=====| 5.50%
    asterisk4|===| 3.67%
    dpkg3|==| 2.75%
    sort3|==| 2.75%
    scp3|==| 2.75%
    /etc/init.d/asterisk2|=| 1.83%
    asterisk.ctl2|=| 1.83%
    apt-cache2|=| 1.83%
    ifconfig1|| 0.92%
    мшÑvim1|| 0.92%
    #1|| 0.92%
    мvi1|| 0.92%
    dpkg-reconfigure1|| 0.92%
    wc1|| 0.92%
    #Install1|| 0.92%
    netstat1|| 0.92%
    вÑifconfig1|| 0.92%
    apt-get1|| 0.92%
    --1|| 0.92%
    ~1|| 0.92%
    Defaults1|| 0.92%
    mv1|| 0.92%
    шifdown1|| 0.92%
    ьлÐmkdir1|| 0.92%
    ps1|| 0.92%
    alias1|| 0.92%
    фÑ1|| 0.92%
    зшping1|| 0.92%
    сÑcat1|| 0.92%
    фыÐcat1|| 0.92%
    ifup1|| 0.92%
    route1|| 0.92%
    vim1|| 0.92%
    шifconfig1|| 0.92%
    gedit1|| 0.92%
    less1|| 0.92%
    /etc/init.d/isc-dhcp-server1|| 0.92%
    exit1|| 0.92%
    #####1|| 0.92%
    ____
    *) 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

    $Id$