Журнал лабораторных работ

Содержание

Журнал

Понедельник (10/17/11)

/dev/pts/1
14:09:15
#ifconfig
eth0      Link encap:Ethernet  HWaddr 2c:27:d7:46:19:8f
          inet addr:192.168.15.21  Bcast:192.168.15.255  Mask:255.255.255.0
          inet6 addr: fe80::2e27:d7ff:fe46:198f/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:51396 errors:0 dropped:0 overruns:0 frame:0
          TX packets:19304 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:38767463 (36.9 MiB)  TX bytes:1983309 (1.8 MiB)
          Interrupt:20 Memory:fe400000-fe420000
lo        Link encap:Local Loopback
          inet addr:127.0.0.1  Mask:255.0.0.0
          inet6 addr: ::1/128 Scope:Host
          UP LOOPBACK RUNNING  MTU:16436  Metric:1
          RX packets:13 errors:0 dropped:0 overruns:0 frame:0
          TX packets:13 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:879 (879.0 B)  TX bytes:879 (879.0 B)
/dev/pts/7
14:09:15
#ifconfig
eth0      Link encap:Ethernet  HWaddr 2c:27:d7:46:19:8f
          inet addr:192.168.15.21  Bcast:192.168.15.255  Mask:255.255.255.0
          inet6 addr: fe80::2e27:d7ff:fe46:198f/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:51396 errors:0 dropped:0 overruns:0 frame:0
          TX packets:19304 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:38767463 (36.9 MiB)  TX bytes:1983309 (1.8 MiB)
          Interrupt:20 Memory:fe400000-fe420000
lo        Link encap:Local Loopback
          inet addr:127.0.0.1  Mask:255.0.0.0
          inet6 addr: ::1/128 Scope:Host
          UP LOOPBACK RUNNING  MTU:16436  Metric:1
          RX packets:13 errors:0 dropped:0 overruns:0 frame:0
          TX packets:13 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:879 (879.0 B)  TX bytes:879 (879.0 B)
/dev/pts/1
14:09:18
#ifup eth0

/dev/pts/7
14:09:18
#ifup eth0

/dev/pts/1
14:09:37
#ifconfig
eth0      Link encap:Ethernet  HWaddr 2c:27:d7:46:19:8f
          inet addr:192.168.10.1  Bcast:192.168.10.255  Mask:255.255.255.0
          inet6 addr: fe80::2e27:d7ff:fe46:198f/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:51446 errors:0 dropped:0 overruns:0 frame:0
          TX packets:19339 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:38773157 (36.9 MiB)  TX bytes:1995270 (1.9 MiB)
          Interrupt:20 Memory:fe400000-fe420000
lo        Link encap:Local Loopback
          inet addr:127.0.0.1  Mask:255.0.0.0
          inet6 addr: ::1/128 Scope:Host
          UP LOOPBACK RUNNING  MTU:16436  Metric:1
          RX packets:15 errors:0 dropped:0 overruns:0 frame:0
          TX packets:15 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:1037 (1.0 KiB)  TX bytes:1037 (1.0 KiB)
/dev/pts/7
14:09:37
#ifconfig
eth0      Link encap:Ethernet  HWaddr 2c:27:d7:46:19:8f
          inet addr:192.168.10.1  Bcast:192.168.10.255  Mask:255.255.255.0
          inet6 addr: fe80::2e27:d7ff:fe46:198f/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:51446 errors:0 dropped:0 overruns:0 frame:0
          TX packets:19339 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:38773157 (36.9 MiB)  TX bytes:1995270 (1.9 MiB)
          Interrupt:20 Memory:fe400000-fe420000
lo        Link encap:Local Loopback
          inet addr:127.0.0.1  Mask:255.0.0.0
          inet6 addr: ::1/128 Scope:Host
          UP LOOPBACK RUNNING  MTU:16436  Metric:1
          RX packets:15 errors:0 dropped:0 overruns:0 frame:0
          TX packets:15 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:1037 (1.0 KiB)  TX bytes:1037 (1.0 KiB)
/dev/pts/1
14:09:39
#ping 192.168.10.254
PING 192.168.10.254 (192.168.10.254) 56(84) bytes of data.
From 192.168.10.1 icmp_seq=1 Destination Host Unreachable
From 192.168.10.1 icmp_seq=2 Destination Host Unreachable
From 192.168.10.1 icmp_seq=5 Destination Host Unreachable
From 192.168.10.1 icmp_seq=7 Destination Host Unreachable
From 192.168.10.1 icmp_seq=8 Destination Host Unreachable
From 192.168.10.1 icmp_seq=10 Destination Host Unreachable
From 192.168.10.1 icmp_seq=11 Destination Host Unreachable
From 192.168.10.1 icmp_seq=14 Destination Host Unreachable
From 192.168.10.1 icmp_seq=15 Destination Host Unreachable
...
From 192.168.10.1 icmp_seq=105 Destination Host Unreachable
From 192.168.10.1 icmp_seq=106 Destination Host Unreachable
From 192.168.10.1 icmp_seq=107 Destination Host Unreachable
From 192.168.10.1 icmp_seq=108 Destination Host Unreachable
From 192.168.10.1 icmp_seq=109 Destination Host Unreachable
From 192.168.10.1 icmp_seq=110 Destination Host Unreachable
^C
--- 192.168.10.254 ping statistics ---
112 packets transmitted, 0 received, +75 errors, 100% packet loss, time 111592ms
pipe 3
/dev/pts/7
14:09:39
#ping 192.168.10.254
PING 192.168.10.254 (192.168.10.254) 56(84) bytes of data.
From 192.168.10.1 icmp_seq=1 Destination Host Unreachable
From 192.168.10.1 icmp_seq=2 Destination Host Unreachable
From 192.168.10.1 icmp_seq=5 Destination Host Unreachable
From 192.168.10.1 icmp_seq=7 Destination Host Unreachable
From 192.168.10.1 icmp_seq=8 Destination Host Unreachable
From 192.168.10.1 icmp_seq=10 Destination Host Unreachable
From 192.168.10.1 icmp_seq=11 Destination Host Unreachable
From 192.168.10.1 icmp_seq=14 Destination Host Unreachable
From 192.168.10.1 icmp_seq=15 Destination Host Unreachable
...
From 192.168.10.1 icmp_seq=105 Destination Host Unreachable
From 192.168.10.1 icmp_seq=106 Destination Host Unreachable
From 192.168.10.1 icmp_seq=107 Destination Host Unreachable
From 192.168.10.1 icmp_seq=108 Destination Host Unreachable
From 192.168.10.1 icmp_seq=109 Destination Host Unreachable
From 192.168.10.1 icmp_seq=110 Destination Host Unreachable
^C
--- 192.168.10.254 ping statistics ---
112 packets transmitted, 0 received, +75 errors, 100% packet loss, time 111592ms
pipe 3
/dev/pts/1
14:11:42
#ping 192.168.10.254
PING 192.168.10.254 (192.168.10.254) 56(84) bytes of data.
From 192.168.10.1 icmp_seq=3 Destination Host Unreachable
From 192.168.10.1 icmp_seq=4 Destination Host Unreachable
From 192.168.10.1 icmp_seq=5 Destination Host Unreachable
From 192.168.10.1 icmp_seq=6 Destination Host Unreachable
From 192.168.10.1 icmp_seq=7 Destination Host Unreachable
From 192.168.10.1 icmp_seq=8 Destination Host Unreachable
From 192.168.10.1 icmp_seq=9 Destination Host Unreachable
From 192.168.10.1 icmp_seq=10 Destination Host Unreachable
From 192.168.10.1 icmp_seq=11 Destination Host Unreachable
...
64 bytes from 192.168.10.254: icmp_req=156 ttl=64 time=0.573 ms
64 bytes from 192.168.10.254: icmp_req=157 ttl=64 time=0.610 ms
64 bytes from 192.168.10.254: icmp_req=158 ttl=64 time=0.560 ms
64 bytes from 192.168.10.254: icmp_req=159 ttl=64 time=0.556 ms
64 bytes from 192.168.10.254: icmp_req=160 ttl=64 time=545 ms
64 bytes from 192.168.10.254: icmp_req=161 ttl=64 time=0.604 ms
^C
--- 192.168.10.254 ping statistics ---
161 packets transmitted, 44 received, +92 errors, 72% packet loss, time 160642ms
rtt min/avg/max/mdev = 0.551/34.220/545.284/94.095 ms, pipe 3
/dev/pts/7
14:11:42
#ping 192.168.10.254
PING 192.168.10.254 (192.168.10.254) 56(84) bytes of data.
From 192.168.10.1 icmp_seq=3 Destination Host Unreachable
From 192.168.10.1 icmp_seq=4 Destination Host Unreachable
From 192.168.10.1 icmp_seq=5 Destination Host Unreachable
From 192.168.10.1 icmp_seq=6 Destination Host Unreachable
From 192.168.10.1 icmp_seq=7 Destination Host Unreachable
From 192.168.10.1 icmp_seq=8 Destination Host Unreachable
From 192.168.10.1 icmp_seq=9 Destination Host Unreachable
From 192.168.10.1 icmp_seq=10 Destination Host Unreachable
From 192.168.10.1 icmp_seq=11 Destination Host Unreachable
...
64 bytes from 192.168.10.254: icmp_req=156 ttl=64 time=0.573 ms
64 bytes from 192.168.10.254: icmp_req=157 ttl=64 time=0.610 ms
64 bytes from 192.168.10.254: icmp_req=158 ttl=64 time=0.560 ms
64 bytes from 192.168.10.254: icmp_req=159 ttl=64 time=0.556 ms
64 bytes from 192.168.10.254: icmp_req=160 ttl=64 time=545 ms
64 bytes from 192.168.10.254: icmp_req=161 ttl=64 time=0.604 ms
^C
--- 192.168.10.254 ping statistics ---
161 packets transmitted, 44 received, +92 errors, 72% packet loss, time 160642ms
rtt min/avg/max/mdev = 0.551/34.220/545.284/94.095 ms, pipe 3
/dev/pts/1
14:14:35
#ping 10.0.35.1
PING 10.0.35.1 (10.0.35.1) 56(84) bytes of data.
^O^O^C
--- 10.0.35.1 ping statistics ---
19 packets transmitted, 0 received, 100% packet loss, time 18144ms
/dev/pts/7
14:14:35
#ping 10.0.35.1
PING 10.0.35.1 (10.0.35.1) 56(84) bytes of data.
^O^O^C
--- 10.0.35.1 ping statistics ---
19 packets transmitted, 0 received, 100% packet loss, time 18144ms
/dev/pts/1
14:15:01
#ping 10.0.35.1
PING 10.0.35.1 (10.0.35.1) 56(84) bytes of data.
^C
--- 10.0.35.1 ping statistics ---
1 packets transmitted, 0 received, 100% packet loss, time 0ms
/dev/pts/7
14:15:01
#ping 10.0.35.1
PING 10.0.35.1 (10.0.35.1) 56(84) bytes of data.
^C
--- 10.0.35.1 ping statistics ---
1 packets transmitted, 0 received, 100% packet loss, time 0ms
/dev/pts/1
14:15:04
#route
Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
^C
/dev/pts/7
14:15:04
#route
Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
^C
/dev/pts/1
14:15:10
#netstat -rn
Kernel IP routing table
Destination     Gateway         Genmask         Flags   MSS Window  irtt Iface
0.0.0.0         192.168.10.254  0.0.0.0         UG        0 0          0 eth0
192.168.10.0    0.0.0.0         255.255.255.0   U         0 0          0 eth0
/dev/pts/7
14:15:10
#netstat -rn
Kernel IP routing table
Destination     Gateway         Genmask         Flags   MSS Window  irtt Iface
0.0.0.0         192.168.10.254  0.0.0.0         UG        0 0          0 eth0
192.168.10.0    0.0.0.0         255.255.255.0   U         0 0          0 eth0
/dev/pts/1
14:15:16
#ping 10.0.35.1
PING 10.0.35.1 (10.0.35.1) 56(84) bytes of data.
^E64 bytes from 10.0.35.1: icmp_req=48 ttl=63 time=2.52 ms
64 bytes from 10.0.35.1: icmp_req=49 ttl=63 time=0.253 ms
64 bytes from 10.0.35.1: icmp_req=50 ttl=63 time=0.248 ms
64 bytes from 10.0.35.1: icmp_req=51 ttl=63 time=0.280 ms
64 bytes from 10.0.35.1: icmp_req=52 ttl=63 time=0.235 ms
64 bytes from 10.0.35.1: icmp_req=53 ttl=63 time=0.256 ms
64 bytes from 10.0.35.1: icmp_req=54 ttl=63 time=0.210 ms
64 bytes from 10.0.35.1: icmp_req=55 ttl=63 time=0.253 ms
64 bytes from 10.0.35.1: icmp_req=56 ttl=63 time=0.239 ms
64 bytes from 10.0.35.1: icmp_req=57 ttl=63 time=0.239 ms
64 bytes from 10.0.35.1: icmp_req=58 ttl=63 time=0.232 ms
64 bytes from 10.0.35.1: icmp_req=59 ttl=63 time=0.224 ms
64 bytes from 10.0.35.1: icmp_req=60 ttl=63 time=0.233 ms
64 bytes from 10.0.35.1: icmp_req=61 ttl=63 time=0.257 ms
^C
--- 10.0.35.1 ping statistics ---
61 packets transmitted, 14 received, 77% packet loss, time 60376ms
rtt min/avg/max/mdev = 0.210/0.405/2.520/0.587 ms
/dev/pts/7
14:15:16
#ping 10.0.35.1
PING 10.0.35.1 (10.0.35.1) 56(84) bytes of data.
^E64 bytes from 10.0.35.1: icmp_req=48 ttl=63 time=2.52 ms
64 bytes from 10.0.35.1: icmp_req=49 ttl=63 time=0.253 ms
64 bytes from 10.0.35.1: icmp_req=50 ttl=63 time=0.248 ms
64 bytes from 10.0.35.1: icmp_req=51 ttl=63 time=0.280 ms
64 bytes from 10.0.35.1: icmp_req=52 ttl=63 time=0.235 ms
64 bytes from 10.0.35.1: icmp_req=53 ttl=63 time=0.256 ms
64 bytes from 10.0.35.1: icmp_req=54 ttl=63 time=0.210 ms
64 bytes from 10.0.35.1: icmp_req=55 ttl=63 time=0.253 ms
64 bytes from 10.0.35.1: icmp_req=56 ttl=63 time=0.239 ms
64 bytes from 10.0.35.1: icmp_req=57 ttl=63 time=0.239 ms
64 bytes from 10.0.35.1: icmp_req=58 ttl=63 time=0.232 ms
64 bytes from 10.0.35.1: icmp_req=59 ttl=63 time=0.224 ms
64 bytes from 10.0.35.1: icmp_req=60 ttl=63 time=0.233 ms
64 bytes from 10.0.35.1: icmp_req=61 ttl=63 time=0.257 ms
^C
--- 10.0.35.1 ping statistics ---
61 packets transmitted, 14 received, 77% packet loss, time 60376ms
rtt min/avg/max/mdev = 0.210/0.405/2.520/0.587 ms
/dev/pts/1
14:16:20
#apt-get install dhcp-server
Чтение списков пакетов... Готово
Построение дерева зависимостей
Чтение информации о состоянии... Готово
E: Не удалось найти пакет dhcp-server
/dev/pts/7
14:16:20
#apt-get install dhcp-server
Чтение списков пакетов... Готово
Построение дерева зависимостей
Чтение информации о состоянии... Готово
E: Не удалось найти пакет dhcp-server
/dev/pts/1
14:16:30
#apt-get install dhcp3-server
Чтение списков пакетов... Готово
Построение дерева зависимостей
Чтение информации о состоянии... Готово
Будут установлены следующие дополнительные пакеты:
  isc-dhcp-server
Предлагаемые пакеты:
  isc-dhcp-server-ldap
НОВЫЕ пакеты, которые будут установлены:
  dhcp3-server isc-dhcp-server
обновлено 0, установлено 2 новых пакетов, для удаления отмечено 0 пакетов, и 0 пакетов не обновлено.
...
Распаковывается пакет isc-dhcp-server (из файла .../isc-dhcp-server_4.1.1-P1-15+squeeze3_i386.deb)...
Выбор ранее не выбранного пакета dhcp3-server.
Распаковывается пакет dhcp3-server (из файла .../dhcp3-server_4.1.1-P1-15+squeeze3_all.deb)...
Обрабатываются триггеры для man-db ...
Настраивается пакет isc-dhcp-server (4.1.1-P1-15+squeeze3) ...
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.
Настраивается пакет dhcp3-server (4.1.1-P1-15+squeeze3) ...
/dev/pts/7
14:16:30
#apt-get install dhcp3-server
Чтение списков пакетов... Готово
Построение дерева зависимостей
Чтение информации о состоянии... Готово
Будут установлены следующие дополнительные пакеты:
  isc-dhcp-server
Предлагаемые пакеты:
  isc-dhcp-server-ldap
НОВЫЕ пакеты, которые будут установлены:
  dhcp3-server isc-dhcp-server
обновлено 0, установлено 2 новых пакетов, для удаления отмечено 0 пакетов, и 0 пакетов не обновлено.
...
Распаковывается пакет isc-dhcp-server (из файла .../isc-dhcp-server_4.1.1-P1-15+squeeze3_i386.deb)...
Выбор ранее не выбранного пакета dhcp3-server.
Распаковывается пакет dhcp3-server (из файла .../dhcp3-server_4.1.1-P1-15+squeeze3_all.deb)...
Обрабатываются триггеры для man-db ...
Настраивается пакет isc-dhcp-server (4.1.1-P1-15+squeeze3) ...
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.
Настраивается пакет dhcp3-server (4.1.1-P1-15+squeeze3) ...
/dev/pts/1
14:16:39
#cd /etc/dhcp/

/dev/pts/7
14:16:39
#cd /etc/dhcp/

/dev/pts/1
14:16:55
#ls
dhclient.conf  dhclient-enter-hooks.d  dhclient-exit-hooks.d  dhcpd.conf
14:16:55
#cp dhcpd.conf dhcpd.conf0

/dev/pts/7
14:16:55
#ls
dhclient.conf  dhclient-enter-hooks.d  dhclient-exit-hooks.d  dhcpd.conf
14:16:55
#cp dhcpd.conf dhcpd.conf0

/dev/pts/1
14:17:03
#vi dhcpd.conf
/dev/pts/7
14:17:03
#vi dhcpd.conf
--- /tmp/l3-saved-3613.9389.32102	2011-10-17 15:17:06.000000000 +0300
+++ dhcpd.conf	2011-10-17 15:19:55.000000000 +0300
@@ -10,8 +10,8 @@
 ddns-update-style none;
 
 # option definitions common to all supported networks...
-option domain-name "example.org";
-option domain-name-servers ns1.example.org, ns2.example.org;
+option domain-name "nt-voip";
+option domain-name-servers 10.0.35.1;
 
 default-lease-time 600;
 max-lease-time 7200;
@@ -47,15 +47,15 @@
 #}
 
 # A slightly different configuration for an internal subnet.
-#subnet 10.5.5.0 netmask 255.255.255.224 {
-#  range 10.5.5.26 10.5.5.30;
+subnet 192.168.10.0 netmask 255.255.255.0 {
+  range 192.168.10.200 192.168.10.250;
 #  option domain-name-servers ns1.internal.example.org;
 #  option domain-name "internal.example.org";
-#  option routers 10.5.5.1;
+  option routers 192.168.10.254;
 #  option broadcast-address 10.5.5.31;
 #  default-lease-time 600;
 #  max-lease-time 7200;
-#}
+}
 
 # Hosts which require special configuration options can be listed in
 # host statements.   If no address is specified, the address will be
/dev/pts/1
14:19:55
#/etc/init.d/isc-dhcp-server restart
Stopping ISC DHCP server: dhcpd failed!
Starting ISC DHCP server: dhcpd.
/dev/pts/7
14:19:55
#/etc/init.d/isc-dhcp-server restart
Stopping ISC DHCP server: dhcpd failed!
Starting ISC DHCP server: dhcpd.
/dev/pts/1
14:20:13
#tail -f /var/log/messages
Oct 17 15:20:09 linux1 dhcpd: For info, please visit https://www.isc.org/software/dhcp/
Oct 17 15:20:11 linux1 dhcpd: Internet Systems Consortium DHCP Server 4.1.1-P1
Oct 17 15:20:11 linux1 dhcpd: Copyright 2004-2010 Internet Systems Consortium.
Oct 17 15:20:11 linux1 dhcpd: All rights reserved.
Oct 17 15:20:11 linux1 dhcpd: For info, please visit https://www.isc.org/software/dhcp/
Oct 17 15:20:11 linux1 dhcpd: Internet Systems Consortium DHCP Server 4.1.1-P1
Oct 17 15:20:11 linux1 dhcpd: Copyright 2004-2010 Internet Systems Consortium.
Oct 17 15:20:11 linux1 dhcpd: All rights reserved.
Oct 17 15:20:11 linux1 dhcpd: For info, please visit https://www.isc.org/software/dhcp/
Oct 17 15:20:11 linux1 dhcpd: Wrote 0 leases to leases file.
...
Oct 17 15:20:46 linux1 dhcpd: DHCPREQUEST for 192.168.10.200 (192.168.10.1) from f0:4d:a2:cc:4f:9b (sm-nb014) via eth0
Oct 17 15:20:46 linux1 dhcpd: DHCPACK on 192.168.10.200 to f0:4d:a2:cc:4f:9b (sm-nb014) via eth0
^[[A^[[B
Oct 17 15:24:24 linux1 dhcpd: DHCPDISCOVER from 00:04:13:24:e5:7e via eth0
Oct 17 15:24:25 linux1 dhcpd: DHCPOFFER on 192.168.10.201 to 00:04:13:24:e5:7e via eth0
Oct 17 15:24:25 linux1 dhcpd: DHCPREQUEST for 192.168.10.201 (192.168.10.1) from 00:04:13:24:e5:7e via eth0
Oct 17 15:24:25 linux1 dhcpd: DHCPACK on 192.168.10.201 to 00:04:13:24:e5:7e via eth0
Oct 17 15:24:41 linux1 dhcpd: DHCPREQUEST for 192.168.10.200 from f0:4d:a2:cc:4f:9b (sm-nb014) via eth0
Oct 17 15:24:41 linux1 dhcpd: DHCPACK on 192.168.10.200 to f0:4d:a2:cc:4f:9b (sm-nb014) via eth0
^C
/dev/pts/7
14:20:13
#tail -f /var/log/messages
Oct 17 15:20:09 linux1 dhcpd: For info, please visit https://www.isc.org/software/dhcp/
Oct 17 15:20:11 linux1 dhcpd: Internet Systems Consortium DHCP Server 4.1.1-P1
Oct 17 15:20:11 linux1 dhcpd: Copyright 2004-2010 Internet Systems Consortium.
Oct 17 15:20:11 linux1 dhcpd: All rights reserved.
Oct 17 15:20:11 linux1 dhcpd: For info, please visit https://www.isc.org/software/dhcp/
Oct 17 15:20:11 linux1 dhcpd: Internet Systems Consortium DHCP Server 4.1.1-P1
Oct 17 15:20:11 linux1 dhcpd: Copyright 2004-2010 Internet Systems Consortium.
Oct 17 15:20:11 linux1 dhcpd: All rights reserved.
Oct 17 15:20:11 linux1 dhcpd: For info, please visit https://www.isc.org/software/dhcp/
Oct 17 15:20:11 linux1 dhcpd: Wrote 0 leases to leases file.
...
Oct 17 15:20:46 linux1 dhcpd: DHCPREQUEST for 192.168.10.200 (192.168.10.1) from f0:4d:a2:cc:4f:9b (sm-nb014) via eth0
Oct 17 15:20:46 linux1 dhcpd: DHCPACK on 192.168.10.200 to f0:4d:a2:cc:4f:9b (sm-nb014) via eth0
^[[A^[[B
Oct 17 15:24:24 linux1 dhcpd: DHCPDISCOVER from 00:04:13:24:e5:7e via eth0
Oct 17 15:24:25 linux1 dhcpd: DHCPOFFER on 192.168.10.201 to 00:04:13:24:e5:7e via eth0
Oct 17 15:24:25 linux1 dhcpd: DHCPREQUEST for 192.168.10.201 (192.168.10.1) from 00:04:13:24:e5:7e via eth0
Oct 17 15:24:25 linux1 dhcpd: DHCPACK on 192.168.10.201 to 00:04:13:24:e5:7e via eth0
Oct 17 15:24:41 linux1 dhcpd: DHCPREQUEST for 192.168.10.200 from f0:4d:a2:cc:4f:9b (sm-nb014) via eth0
Oct 17 15:24:41 linux1 dhcpd: DHCPACK on 192.168.10.200 to f0:4d:a2:cc:4f:9b (sm-nb014) via eth0
^C
/dev/pts/1
14:25:06
#route
Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
default         gw1.unix.nt     0.0.0.0         UG    0      0        0 eth0
192.168.10.0    *               255.255.255.0   U     0      0        0 eth0
/dev/pts/7
14:25:06
#route
Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
default         gw1.unix.nt     0.0.0.0         UG    0      0        0 eth0
192.168.10.0    *               255.255.255.0   U     0      0        0 eth0
/dev/pts/1
14:25:08
#route
Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
default         gw1.unix.nt     0.0.0.0         UG    0      0        0 eth0
192.168.10.0    *               255.255.255.0   U     0      0        0 eth0
/dev/pts/7
14:25:08
#route
Kernel IP routing table
Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
default         gw1.unix.nt     0.0.0.0         UG    0      0        0 eth0
192.168.10.0    *               255.255.255.0   U     0      0        0 eth0
прошло 36 минут
/dev/pts/1
15:01:26
#vi dhcpd.conf
/dev/pts/7
15:01:26
#vi dhcpd.conf
--- /tmp/l3-saved-3613.14504.24215	2011-10-17 16:01:31.000000000 +0300
+++ dhcpd.conf	2011-10-17 16:01:47.000000000 +0300
@@ -10,7 +10,7 @@
 ddns-update-style none;
 
 # option definitions common to all supported networks...
-option domain-name "nt-voip";
+option domain-name "unix.nt";
 option domain-name-servers 10.0.35.1;
 
 default-lease-time 600;
/dev/pts/1
15:01:47
#/etc/init.d/isc-dhcp-server restart
Stopping ISC DHCP server: dhcpd.
Starting ISC DHCP server: dhcpd.
/dev/pts/7
15:01:47
#/etc/init.d/isc-dhcp-server restart
Stopping ISC DHCP server: dhcpd.
Starting ISC DHCP server: dhcpd.
прошла 41 минута
/dev/pts/1
15:43:09
#apt-get install tcpdump
Чтение списков пакетов... Готово
Построение дерева зависимостей
Чтение информации о состоянии... Готово
НОВЫЕ пакеты, которые будут установлены:
  tcpdump
обновлено 0, установлено 1 новых пакетов, для удаления отмечено 0 пакетов, и 0 пакетов не обновлено.
Необходимо скачать 376 kБ архивов.
После данной операции, объём занятого дискового пространства возрастёт на 901 kB.
Получено:1 http://10.0.35.1/debian/ squeeze/main tcpdump i386 4.1.1-1 [376 kB]
Получено 376 kБ за 0с (4 854 kБ/c)
Выбор ранее не выбранного пакета tcpdump.
(Чтение базы данных ... на данный момент установлен 115391 файл и каталог.)
Распаковывается пакет tcpdump (из файла .../tcpdump_4.1.1-1_i386.deb)...
Обрабатываются триггеры для man-db ...
Настраивается пакет tcpdump (4.1.1-1) ...
/dev/pts/7
15:43:09
#apt-get install tcpdump
Чтение списков пакетов... Готово
Построение дерева зависимостей
Чтение информации о состоянии... Готово
НОВЫЕ пакеты, которые будут установлены:
  tcpdump
обновлено 0, установлено 1 новых пакетов, для удаления отмечено 0 пакетов, и 0 пакетов не обновлено.
Необходимо скачать 376 kБ архивов.
После данной операции, объём занятого дискового пространства возрастёт на 901 kB.
Получено:1 http://10.0.35.1/debian/ squeeze/main tcpdump i386 4.1.1-1 [376 kB]
Получено 376 kБ за 0с (4 854 kБ/c)
Выбор ранее не выбранного пакета tcpdump.
(Чтение базы данных ... на данный момент установлен 115391 файл и каталог.)
Распаковывается пакет tcpdump (из файла .../tcpdump_4.1.1-1_i386.deb)...
Обрабатываются триггеры для man-db ...
Настраивается пакет tcpdump (4.1.1-1) ...
/dev/pts/1
15:43:26
#tcpdump
tcpdump: verbose output suppressed, use -v or -vv for full protocol decode
listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes
16:43:31.785341 IP linux1.unix.nt.34454 > 192.168.10.201.www: Flags [S], seq 1374056296, win 14600, options [mss 1460,sackOK,TS val 3495960 ecr 0,nop,wscale 6], length 0
16:43:31.785661 IP linux1.unix.nt.46698 > 10.0.35.1.domain: 63898+ PTR? 201.10.168.192.in-addr.arpa. (45)
16:43:31.785849 IP 192.168.10.201.www > linux1.unix.nt.34454: Flags [R.], seq 0, ack 1374056297, win 0, length 0
16:43:31.786065 IP 10.0.35.1.domain > linux1.unix.nt.46698: 63898 NXDomain* 0/1/0 (95)
16:43:31.886451 IP6 fe80::2e27:d7ff:fe46:198f.mdns > ff02::fb.mdns: 0 PTR (QM)? 201.10.168.192.in-addr.arpa. (45)
16:43:31.886479 IP linux1.unix.nt.mdns > 224.0.0.251.mdns: 0 PTR (QM)? 201.10.168.192.in-addr.arpa. (45)
16:43:31.964255 IP 192.168.10.200.53653 > 229.111.112.12.3071: UDP, length 4
16:43:32.887697 IP6 fe80::2e27:d7ff:fe46:198f.mdns > ff02::fb.mdns: 0 PTR (QM)? 201.10.168.192.in-addr.arpa. (45)
...
16:57:10.534949 IP 192.168.10.200.53653 > 229.111.112.12.3071: UDP, length 4
16:57:15.135788 IP linux1.unix.nt.33027 > note.unix.nt.ssh: Flags [P.], seq 2066:2130, ack 4937, win 402, options [nop,nop,TS val 3701798 ecr 26240734], length 64
16:57:15.138947 IP note.unix.nt.ssh > linux1.unix.nt.33027: Flags [P.], seq 4937:5641, ack 2130, win 223, options [nop,nop,TS val 26241975 ecr 3701798], length 704
16:57:15.138958 IP linux1.unix.nt.33027 > note.unix.nt.ssh: Flags [.], ack 5641, win 446, options [nop,nop,TS val 3701798 ecr 26241975], length 0
16:57:16.773849 IP 192.168.10.200.53653 > 229.111.112.12.3071: UDP, length 4
^C16:57:17.703415 LLDP, name ProCurve Switch 3400cl-24G, length 166
^C
3040 packets captured
3266 packets received by filter
226 packets dropped by kernel
/dev/pts/7
15:43:26
#tcpdump
tcpdump: verbose output suppressed, use -v or -vv for full protocol decode
listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes
16:43:31.785341 IP linux1.unix.nt.34454 > 192.168.10.201.www: Flags [S], seq 1374056296, win 14600, options [mss 1460,sackOK,TS val 3495960 ecr 0,nop,wscale 6], length 0
16:43:31.785661 IP linux1.unix.nt.46698 > 10.0.35.1.domain: 63898+ PTR? 201.10.168.192.in-addr.arpa. (45)
16:43:31.785849 IP 192.168.10.201.www > linux1.unix.nt.34454: Flags [R.], seq 0, ack 1374056297, win 0, length 0
16:43:31.786065 IP 10.0.35.1.domain > linux1.unix.nt.46698: 63898 NXDomain* 0/1/0 (95)
16:43:31.886451 IP6 fe80::2e27:d7ff:fe46:198f.mdns > ff02::fb.mdns: 0 PTR (QM)? 201.10.168.192.in-addr.arpa. (45)
16:43:31.886479 IP linux1.unix.nt.mdns > 224.0.0.251.mdns: 0 PTR (QM)? 201.10.168.192.in-addr.arpa. (45)
16:43:31.964255 IP 192.168.10.200.53653 > 229.111.112.12.3071: UDP, length 4
16:43:32.887697 IP6 fe80::2e27:d7ff:fe46:198f.mdns > ff02::fb.mdns: 0 PTR (QM)? 201.10.168.192.in-addr.arpa. (45)
...
16:57:10.534949 IP 192.168.10.200.53653 > 229.111.112.12.3071: UDP, length 4
16:57:15.135788 IP linux1.unix.nt.33027 > note.unix.nt.ssh: Flags [P.], seq 2066:2130, ack 4937, win 402, options [nop,nop,TS val 3701798 ecr 26240734], length 64
16:57:15.138947 IP note.unix.nt.ssh > linux1.unix.nt.33027: Flags [P.], seq 4937:5641, ack 2130, win 223, options [nop,nop,TS val 26241975 ecr 3701798], length 704
16:57:15.138958 IP linux1.unix.nt.33027 > note.unix.nt.ssh: Flags [.], ack 5641, win 446, options [nop,nop,TS val 3701798 ecr 26241975], length 0
16:57:16.773849 IP 192.168.10.200.53653 > 229.111.112.12.3071: UDP, length 4
^C16:57:17.703415 LLDP, name ProCurve Switch 3400cl-24G, length 166
^C
3040 packets captured
3266 packets received by filter
226 packets dropped by kernel
прошло 11 минут
/dev/pts/5
15:55:25
#ssh user@192.168.90.1
The authenticity of host '192.168.90.1 (192.168.90.1)' can't be established.
RSA key fingerprint is f0:05:a6:a6:88:29:cd:4d:7a:23:9b:50:fa:00:de:0c.
Are you sure you want to continue connecting (yes/no)? yes
Warning: Permanently added '192.168.90.1' (RSA) to the list of known hosts.
user@192.168.90.1's password:
Permission denied, please try again.
user@192.168.90.1's password:
Permission denied, please try again.
user@192.168.90.1's password:
Permission denied (publickey,password).
/dev/pts/4
15:55:25
#ssh user@192.168.90.1
The authenticity of host '192.168.90.1 (192.168.90.1)' can't be established.
RSA key fingerprint is f0:05:a6:a6:88:29:cd:4d:7a:23:9b:50:fa:00:de:0c.
Are you sure you want to continue connecting (yes/no)? yes
Warning: Permanently added '192.168.90.1' (RSA) to the list of known hosts.
user@192.168.90.1's password:
Permission denied, please try again.
user@192.168.90.1's password:
Permission denied, please try again.
user@192.168.90.1's password:
Permission denied (publickey,password).
15:55:52
#ssh user@192.168.90.1
user@192.168.90.1's password:
Permission denied, please try again.
user@192.168.90.1's password:
Permission denied, please try again.
user@192.168.90.1's password:
Permission denied (publickey,password).
/dev/pts/5
15:55:52
#ssh user@192.168.90.1
user@192.168.90.1's password:
Permission denied, please try again.
user@192.168.90.1's password:
Permission denied, please try again.
user@192.168.90.1's password:
Permission denied (publickey,password).
/dev/pts/4
15:56:41
#ssh user@192.168.15.252
[ulaw-phone](!)
        disallow=all
        allow=ulaw
[root@linux9:~]# mv /etc/asterisk/sip.conf /etc/asterisk/sip.conf.SAVED
[root@linux9:~]# cat /etc/asterisk/sip.conf.SAVED | sed 's/;.*//' | expand |  gr
[root@linux9:~]# cat /etc/asterisk/sip.conf.SAVED | sed 's/;.*//' | expand |  gr
[root@linux9:~]# cat /etc/asterisk/sip.conf
[general]
context=default
allowoverlap=no
...
tcpbindaddr=0.0.0.0
srvlookup=yes
[root@linux9:~]# cat /etc/asterisk/sip.conf
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
/dev/pts/5
15:56:41
#ssh user@192.168.15.252
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
[authentication]
[basic-options](!)
        dtmfmode=rfc2833
        context=from-office
        type=friend
[natted-phone](!,basic-options)
        nat=yes
...
tcpbindaddr=0.0.0.0
srvlookup=yes
[root@linux9:~]# cat /etc/asterisk/sip.conf
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
/dev/pts/1
15:57:17
#cd /etc/asterisk/

/dev/pts/7
15:57:17
#cd /etc/asterisk/

/dev/pts/1
15:57:21
#ls
adsi.conf                enum.conf               muted.conf
adtranvofr.conf          extconfig.conf          osp.conf
agents.conf              extensions.ael          oss.conf
ais.conf                 extensions.conf         phone.conf
alarmreceiver.conf       extensions.lua          phoneprov.conf
alsa.conf                extensions_minivm.conf  queuerules.conf
amd.conf                 features.conf           queues.conf
asterisk.adsi            festival.conf           res_config_sqlite.conf
asterisk.conf            followme.conf           res_ldap.conf
cdr_adaptive_odbc.conf   func_odbc.conf          res_odbc.conf
...
chan_dahdi.conf          jingle.conf             skinny.conf
cli_aliases.conf         logger.conf             sla.conf
cli.conf                 manager.conf            smdi.conf
cli_permissions.conf     manager.d               telcordia-1.adsi
codecs.conf              meetme.conf             udptl.conf
console.conf             mgcp.conf               unistim.conf
dbsep.conf               minivm.conf             usbradio.conf
dnsmgr.conf              misdn.conf              users.conf
dsp.conf                 modules.conf            voicemail.conf
dundi.conf               musiconhold.conf        vpb.conf
/dev/pts/7
15:57:21
#ls
adsi.conf                enum.conf               muted.conf
adtranvofr.conf          extconfig.conf          osp.conf
agents.conf              extensions.ael          oss.conf
ais.conf                 extensions.conf         phone.conf
alarmreceiver.conf       extensions.lua          phoneprov.conf
alsa.conf                extensions_minivm.conf  queuerules.conf
amd.conf                 features.conf           queues.conf
asterisk.adsi            festival.conf           res_config_sqlite.conf
asterisk.conf            followme.conf           res_ldap.conf
cdr_adaptive_odbc.conf   func_odbc.conf          res_odbc.conf
...
chan_dahdi.conf          jingle.conf             skinny.conf
cli_aliases.conf         logger.conf             sla.conf
cli.conf                 manager.conf            smdi.conf
cli_permissions.conf     manager.d               telcordia-1.adsi
codecs.conf              meetme.conf             udptl.conf
console.conf             mgcp.conf               unistim.conf
dbsep.conf               minivm.conf             usbradio.conf
dnsmgr.conf              misdn.conf              users.conf
dsp.conf                 modules.conf            voicemail.conf
dundi.conf               musiconhold.conf        vpb.conf
/dev/pts/1
15:57:22
#cp sip.conf sip.conf0

/dev/pts/7
15:57:22
#cp sip.conf sip.conf0

/dev/pts/1
15:57:28
#cat sip.conf
;
; SIP Configuration example for Asterisk
;
; SIP dial strings
;-----------------------------------------------------------
; In the dialplan (extensions.conf) you can use several
; syntaxes for dialing SIP devices.
;        SIP/devicename
;        SIP/username@domain   (SIP uri)
;        SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
...
;[pre14-asterisk]
;type=friend
;secret=digium
;host=dynamic
;rfc2833compensate=yes          ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
                                ; You must have this turned on or DTMF reception will work improperly.
;t38pt_usertpsource=yes         ; Use the source IP address of RTP as the destination IP address for UDPTL packets
                                ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
                                ; external IP address of the remote device. If port forwarding is done at the client side
                                ; then UDPTL will flow to the remote device.
/dev/pts/7
15:57:28
#cat sip.conf
;
; SIP Configuration example for Asterisk
;
; SIP dial strings
;-----------------------------------------------------------
; In the dialplan (extensions.conf) you can use several
; syntaxes for dialing SIP devices.
;        SIP/devicename
;        SIP/username@domain   (SIP uri)
;        SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
...
;[pre14-asterisk]
;type=friend
;secret=digium
;host=dynamic
;rfc2833compensate=yes          ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
                                ; You must have this turned on or DTMF reception will work improperly.
;t38pt_usertpsource=yes         ; Use the source IP address of RTP as the destination IP address for UDPTL packets
                                ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
                                ; external IP address of the remote device. If port forwarding is done at the client side
                                ; then UDPTL will flow to the remote device.
/dev/pts/1
15:57:39
#cat sip.conf | sed 's/;.*//'
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
[authentication]
[basic-options](!)
        dtmfmode=rfc2833
...
[my-codecs](!)
        disallow=all
        allow=ilbc
        allow=g729
        allow=gsm
        allow=g723
        allow=ulaw
[ulaw-phone](!)
        disallow=all
        allow=ulaw
/dev/pts/7
15:57:39
#cat sip.conf | sed 's/;.*//'
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
[authentication]
[basic-options](!)
        dtmfmode=rfc2833
...
[my-codecs](!)
        disallow=all
        allow=ilbc
        allow=g729
        allow=gsm
        allow=g723
        allow=ulaw
[ulaw-phone](!)
        disallow=all
        allow=ulaw
/dev/pts/1
15:57:55
#cat sip.conf | sed 's/;.*//' | expand
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
[authentication]
[basic-options](!)
        dtmfmode=rfc2833
...
[my-codecs](!)
        disallow=all
        allow=ilbc
        allow=g729
        allow=gsm
        allow=g723
        allow=ulaw
[ulaw-phone](!)
        disallow=all
        allow=ulaw
/dev/pts/7
15:57:55
#cat sip.conf | sed 's/;.*//' | expand
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
[authentication]
[basic-options](!)
        dtmfmode=rfc2833
...
[my-codecs](!)
        disallow=all
        allow=ilbc
        allow=g729
        allow=gsm
        allow=g723
        allow=ulaw
[ulaw-phone](!)
        disallow=all
        allow=ulaw
/dev/pts/1
15:58:02
#cat sip.conf | sed 's/;.*//' | expand
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
[authentication]
[basic-options](!)
        dtmfmode=rfc2833
...
[my-codecs](!)
        disallow=all
        allow=ilbc
        allow=g729
        allow=gsm
        allow=g723
        allow=ulaw
[ulaw-phone](!)
        disallow=all
        allow=ulaw
/dev/pts/7
15:58:02
#cat sip.conf | sed 's/;.*//' | expand
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
[authentication]
[basic-options](!)
        dtmfmode=rfc2833
...
[my-codecs](!)
        disallow=all
        allow=ilbc
        allow=g729
        allow=gsm
        allow=g723
        allow=ulaw
[ulaw-phone](!)
        disallow=all
        allow=ulaw
/dev/pts/1
15:58:19
#vi sip.conf
/dev/pts/7
15:58:19
#vi sip.conf
/dev/pts/1
15:58:19
#vi sip.conf
/dev/pts/7
15:58:19
#vi sip.conf
--- /tmp/l3-saved-3613.9538.10453	2011-10-17 16:58:45.000000000 +0300
+++ sip.conf	2011-10-17 16:59:03.000000000 +0300
@@ -1,1153 +1,7 @@
-;
-; SIP Configuration example for Asterisk
-;
-; SIP dial strings
-;-----------------------------------------------------------
-; In the dialplan (extensions.conf) you can use several 
-; syntaxes for dialing SIP devices.
-;        SIP/devicename
-;        SIP/username@domain   (SIP uri)
-;        SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
-;        SIP/devicename/extension
-;
-;
-; 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
-;
-; 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
-;
-; 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
-;
-;   module reload chan_sip.so    Reload configuration file
-;
-;------- 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.
-;-----------------------------------------------------------------------------
-
-; ** Deprecated configuration options **
-; The "call-limit" configuation option is deprecated. It still works in
-; this version of Asterisk, but will disappear in the next version.
-; 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)
-;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)
-;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
-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)
-
-;
-; 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
-
-;tlscertfile=asterisk.pem       ; Certificate file (*.pem only) to use for TLS connections 
-                                ; default is to look for "asterisk.pem" in current directory
-
-;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.
-
-;tlscadir=</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
-
-;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)
-
-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 "no")
-
-; See qos.tex or Quality of Service section of asterisk.pdf 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
-;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
-;qualifygap=100			; Number of milliseconds between each group of peers being qualified
-;qualifypeers=1			; Number of peers in a group to be qualified at the same time
-;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.
-;vmexten=voicemail              ; dialplan extension to reach mailbox sets the 
-                                ; Message-Account in the MWI notify message 
-                                ; defaults to "asterisk"
-;disallow=all                   ; First disallow all codecs
-;allow=ulaw                     ; Allow codecs in order of preference
-;allow=ilbc                     ; see doc/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
-;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 "no" will stop any media before we have
-				; call progress. Default is "yes".
-
-;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.
-;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
-;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.
-
-;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 
-;                                               ; (could also be tcp,udp) - defining transports on the proxy line only
-;                                               ; applies for the global proxy, otherwise use the transport= option
-;matchexterniplocally = yes     ; Only substitute the externip 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.
-
-; 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
-
-;
-; 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
-;
-;--------------------------- 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
-;
-;--------------------------- HASH TABLE SIZES ------------------------------------------------
-; For maximum efficiency, adjust the following
-; values to be slightly larger than the maximum number of in-memory objects (devices).
-; Too large, and space is wasted. Too small, and things will run slower.
-; 563 is probably way too big for small (home) applications, but it
-; should cover most small/medium sites.
-; It is recommended to make the sizes be a prime number!
-; This was internally set to 17 for small-memory applications...
-; All tables default to 563, except when compiled in LOW_MEMORY mode,
-; in which case, they default to 17. You can override this by uncommenting
-; the following, and changing the values.
-;hash_users=563
-;hash_peers=563
-;hash_dialogs=563
-
-;--------------------------- 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
-; faxdetect = both		; Enables both CNG and T.38 detection (same as 'yes')
-;
-;----------------------------------------- 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.
-; Format for the mwi register statement is:
-;       mwi => user[:secret[:authuser]]@host[:port][/mailbox]
-;
-; Examples:
-;mwi => 1234:password@mysipprovider.com/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. "externip = 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 "bindport" value (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:
-;
-;        externip = 12.34.56.78          ; use this address.
-;        externip = 12.34.56.78:9900     ; use this address and port.
-;        externip = mynat.my.org:12600   ; Public address of my nat box.
-;
-;   b. "externhost = hostname[:port]" is similar to "externip" 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
-;
-;   c. "stunaddr = stun.server[:port]" queries the STUN server specified
-;      as an argument to obtain the external address/port.
-;      Queries are also sent periodically every "externrefresh" seconds
-;      (as a side effect, sending the query also acts as a keepalive for
-;      the state entry on the nat box):
-;
-;        stunaddr = foo.stun.com:3478
-;        externrefresh = 15
-;
-;   Note that at the moment all these mechanism work only for the SIP socket.
-;   The IP address discovered with externip/externhost/STUN 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 "externip" and
-; "externhost" might not help you configure addresses properly, and you
-; really need to use STUN.
-;
-; NOTE 2: when using "externip" or "externhost", the address part is
-; also used as the external address for media sessions.
-; If you use "stunaddr", STUN queries will be sent to the same server
-; also from media sockets, and this should permit a correct mapping of
-; the port numbers as well.
-;
-; 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                ; default. Use NAT mode only according to RFC3581 (;rport)
-;        nat = yes               ; Always ignore info and assume NAT
-;        nat = never             ; Never attempt NAT mode or RFC3581 support
-;        nat = route             ; route = Assume NAT, don't send rport 
-;                                ; (work around more UNIDEN bugs)
-
-;----------------------------------- 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).
-
-;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.
-
-                                ; 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'.
-
-;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.
-
-;----------------------------------------- REALTIME SUPPORT ------------------------
-; For additional information on ARA, the Asterisk Realtime Architecture,
-; please read realtime.txt and extconfig.txt in the /doc directory of the
-; source code.
-;
-;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. 
-
-;------------------------------ 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".
-;-----------------------------------------------------------------------------------
-
-[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
-; 
-; The SIP channel has two types of devices, the friend and the peer.
-; * The type=friend is a device type that accepts both incoming and outbound calls,
-;   where Asterisk match on the From: username on incoming calls.
-;   (A synonym for friend is "user"). This is a type you use for your local
-;   SIP phones.
-; * The type=peer also handles both incoming and outbound calls. On inbound calls,
-;   Asterisk only matches on IP/port, not on names. This is mostly used for SIP
-;   trunks.
-;
-; 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).
-
-;[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.
-
-;
-; 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
-        nat=yes
-        directmedia=no
-        host=dynamic
-
-[public-phone](!,basic-options)   ; another template inheriting basic-options
-        nat=no
-        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
-;nat=no                          ; there is not NAT between phone and Asterisk
-;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
-;nat=yes                         ; X-Lite is behind a NAT router
-;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
-
-;[cisco1]
-;type=friend
-;secret=blah
-;qualify=200                     ; Qualify peer is no more than 200ms away
-;nat=yes                         ; This phone may be natted
-                                 ; Send SIP and RTP to the IP address that packet is 
-                                 ; received from instead of trusting SIP headers 
-;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.
+context=default
+allowoverlap=no
+udpbindaddr=0.0.0.0
+tcpenable=no
+tcpbindaddr=0.0.0.0
+srvlookup=yes
/dev/pts/1
15:59:03
#vi sip.conf
/dev/pts/7
15:59:03
#vi sip.conf
--- /tmp/l3-saved-3613.22321.31742	2011-10-17 16:59:05.000000000 +0300
+++ sip.conf	2011-10-17 17:14:56.000000000 +0300
@@ -5,3 +5,13 @@
 tcpenable=no
 tcpbindaddr=0.0.0.0
 srvlookup=yes
+
+[1101]
+type=friend
+secret=1234
+host=192.168.10.201
+
+[1102]
+type=friend
+secret=1234
+host=192.168.10.200
прошло 15 минут
/dev/pts/1
16:14:56
#asterisk -rv
Asterisk 1.6.2.9-2+squeeze3, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.6.2.9-2+squeeze3 currently running on linux1 (pid = 6558)
Verbosity is at least 2
linux1*CLI> sip reload
...
1101                       192.168.10.201              5060     Unmonitored
1102                       192.168.10.200              5060     Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
linux1*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
1101                       192.168.10.201              5060     Unmonitored
1102                       192.168.10.200              5060     Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
linux1*CLI> exit
Executing last minute cleanups
/dev/pts/7
16:14:56
#asterisk -rv
Asterisk 1.6.2.9-2+squeeze3, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.6.2.9-2+squeeze3 currently running on linux1 (pid = 6558)
Verbosity is at least 2
linux1*CLI> sip reload
...
1101                       192.168.10.201              5060     Unmonitored
1102                       192.168.10.200              5060     Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
linux1*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
1101                       192.168.10.201              5060     Unmonitored
1102                       192.168.10.200              5060     Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
linux1*CLI> exit
Executing last minute cleanups
/dev/pts/1
16:24:33
#asterisk -rvvvvvvvvvvvvvvv
Asterisk 1.6.2.9-2+squeeze3, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.2.9-2+squeeze3 currently running on linux1 (pid = 6558)
...
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
linux1*CLI> sip set debug peer 1101
SIP Debugging Enabled for IP: 192.168.10.201:5060
linux1*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
1101                       192.168.10.201              5060     Unmonitored
1102                       192.168.10.200              5060     Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
linux1*CLI> exit
Executing last minute cleanups
/dev/pts/7
16:24:33
#asterisk -rvvvvvvvvvvvvvvv
Asterisk 1.6.2.9-2+squeeze3, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.2.9-2+squeeze3 currently running on linux1 (pid = 6558)
...
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
linux1*CLI> sip set debug peer 1101
SIP Debugging Enabled for IP: 192.168.10.201:5060
linux1*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
1101                       192.168.10.201              5060     Unmonitored
1102                       192.168.10.200              5060     Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
linux1*CLI> exit
Executing last minute cleanups
/dev/pts/1
16:26:00
#vi sip.conf
/dev/pts/7
16:26:00
#vi sip.conf
--- /tmp/l3-saved-3613.8429.11535	2011-10-17 17:26:02.000000000 +0300
+++ sip.conf	2011-10-17 17:26:24.000000000 +0300
@@ -9,9 +9,11 @@
 [1101]
 type=friend
 secret=1234
-host=192.168.10.201
+host=dynamic
+monitor=yes
 
 [1102]
 type=friend
 secret=1234
-host=192.168.10.200
+host=dynamic
+monitor=yes
/dev/pts/1
16:26:24
#asterisk -rvvvvvvvvvvvvvvv
Asterisk 1.6.2.9-2+squeeze3, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.2.9-2+squeeze3 currently running on linux1 (pid = 6558)
...
linux1*CLI>
linux1*CLI>
linux1*CLI>
linux1*CLI>
linux1*CLI>
linux1*CLI>
linux1*CLI>
linux1*CLI>
linux1*CLI> exit
Executing last minute cleanups
/dev/pts/7
16:26:24
#asterisk -rvvvvvvvvvvvvvvv
Asterisk 1.6.2.9-2+squeeze3, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.2.9-2+squeeze3 currently running on linux1 (pid = 6558)
...
linux1*CLI>
linux1*CLI>
linux1*CLI>
linux1*CLI>
linux1*CLI>
linux1*CLI>
linux1*CLI>
linux1*CLI>
linux1*CLI> exit
Executing last minute cleanups
/dev/pts/1
16:27:29
#ifconfig
eth0      Link encap:Ethernet  HWaddr 2c:27:d7:46:19:8f
          inet addr:192.168.10.1  Bcast:192.168.10.255  Mask:255.255.255.0
          inet6 addr: fe80::2e27:d7ff:fe46:198f/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:58969 errors:0 dropped:0 overruns:0 frame:0
          TX packets:26344 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:45243393 (43.1 MiB)  TX bytes:4612827 (4.3 MiB)
          Interrupt:20 Memory:fe400000-fe420000
lo        Link encap:Local Loopback
          inet addr:127.0.0.1  Mask:255.0.0.0
          inet6 addr: ::1/128 Scope:Host
          UP LOOPBACK RUNNING  MTU:16436  Metric:1
          RX packets:255 errors:0 dropped:0 overruns:0 frame:0
          TX packets:255 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:26237 (25.6 KiB)  TX bytes:26237 (25.6 KiB)
/dev/pts/7
16:27:29
#ifconfig
eth0      Link encap:Ethernet  HWaddr 2c:27:d7:46:19:8f
          inet addr:192.168.10.1  Bcast:192.168.10.255  Mask:255.255.255.0
          inet6 addr: fe80::2e27:d7ff:fe46:198f/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:58969 errors:0 dropped:0 overruns:0 frame:0
          TX packets:26344 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:45243393 (43.1 MiB)  TX bytes:4612827 (4.3 MiB)
          Interrupt:20 Memory:fe400000-fe420000
lo        Link encap:Local Loopback
          inet addr:127.0.0.1  Mask:255.0.0.0
          inet6 addr: ::1/128 Scope:Host
          UP LOOPBACK RUNNING  MTU:16436  Metric:1
          RX packets:255 errors:0 dropped:0 overruns:0 frame:0
          TX packets:255 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:26237 (25.6 KiB)  TX bytes:26237 (25.6 KiB)
/dev/pts/1
16:27:31
#asterisk -rvvvvvvvvvvvvvvv
Asterisk 1.6.2.9-2+squeeze3, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.2.9-2+squeeze3 currently running on linux1 (pid = 6558)
...
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
linux1*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
1101                       (Unspecified)    D          5060     Unmonitored
1102/1102                  192.168.10.200   D          13826    Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
<--- SIP read from UDP:192.168.10.200:13826 --->
<------------->
linux1*CLI> exit
Executing last minute cleanups
/dev/pts/7
16:27:31
#asterisk -rvvvvvvvvvvvvvvv
Asterisk 1.6.2.9-2+squeeze3, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.2.9-2+squeeze3 currently running on linux1 (pid = 6558)
...
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
linux1*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
1101                       (Unspecified)    D          5060     Unmonitored
1102/1102                  192.168.10.200   D          13826    Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
<--- SIP read from UDP:192.168.10.200:13826 --->
<------------->
linux1*CLI> exit
Executing last minute cleanups
прошло 13 минут
/dev/pts/1
16:40:42
#vi /etc/dhcp/dhcpd.conf
/dev/pts/7
16:40:42
#vi /etc/dhcp/dhcpd.conf
/dev/pts/1
16:41:41
#cp extensions.conf extensions.conf0

/dev/pts/7
16:41:41
#cp extensions.conf extensions.conf0

/dev/pts/1
16:41:56
#vi extensions.conf
/dev/pts/7
16:41:56
#vi extensions.conf
--- /tmp/l3-saved-3613.16071.10840	2011-10-17 17:42:02.000000000 +0300
+++ extensions.conf	2011-10-17 17:43:40.000000000 +0300
@@ -1,846 +1,5 @@
-; extensions.conf - the Asterisk dial plan
-;
-; Static extension configuration file, used by
-; the pbx_config module. This is where you configure all your 
-; inbound and outbound calls in Asterisk. 
-; 
-; This configuration file is reloaded 
-; - With the "dialplan reload" command in the CLI
-; - With the "reload" command (that reloads everything) in the CLI
-
-;
-; The "General" category is for certain variables.  
-;
-[general]
-;
-; If static is set to no, or omitted, then the pbx_config will rewrite
-; this file when extensions are modified.  Remember that all comments
-; made in the file will be lost when that happens. 
-;
-; XXX Not yet implemented XXX
-;
-static=yes
-;
-; if static=yes and writeprotect=no, you can save dialplan by
-; CLI command "dialplan save" too
-;
-writeprotect=no
-;
-; If autofallthrough is set, then if an extension runs out of
-; things to do, it will terminate the call with BUSY, CONGESTION
-; or HANGUP depending on Asterisk's best guess. This is the default.
-;
-; If autofallthrough is not set, then if an extension runs out of 
-; things to do, Asterisk will wait for a new extension to be dialed 
-; (this is the original behavior of Asterisk 1.0 and earlier).
-;
-;autofallthrough=no
-;
-;
-;
-; If extenpatternmatchnew is set (true, yes, etc), then a new algorithm that uses
-; a Trie to find the best matching pattern is used. In dialplans
-; with more than about 20-40 extensions in a single context, this
-; new algorithm can provide a noticeable speedup. 
-; With 50 extensions, the speedup is 1.32x
-; with 88 extensions, the speedup is 2.23x
-; with 138 extensions, the speedup is 3.44x
-; with 238 extensions, the speedup is 5.8x
-; with 438 extensions, the speedup is 10.4x
-; With 1000 extensions, the speedup is ~25x
-; with 10,000 extensions, the speedup is 374x
-; Basically, the new algorithm provides a flat response 
-; time, no matter the number of extensions.
-;
-; By default, the old pattern matcher is used. 
-;
-; ****This is a new feature! *********************
-; The new pattern matcher is for the brave, the bold, and 
-; the desperate. If you have large dialplans (more than about 50 extensions
-; in a context), and/or high call volume, you might consider setting 
-; this value to "yes" !!
-; Please, if you try this out, and are forced to return to the
-; old pattern matcher, please report your reasons in a bug report
-; on bugs.digium.com. We have made good progress in providing something
-; compatible with the old matcher; help us finish the job!
-;
-; This value can be switched at runtime using the cli command "dialplan set extenpatternmatchnew true"
-; or "dialplan set extenpatternmatchnew false", so you can experiment to your hearts content.
-;
-;extenpatternmatchnew=no
-;
-; If clearglobalvars is set, global variables will be cleared 
-; and reparsed on a dialplan reload, or Asterisk reload.
-;
-; If clearglobalvars is not set, then global variables will persist
-; through reloads, and even if deleted from the extensions.conf or
-; one of its included files, will remain set to the previous value.
-;
-; NOTE: A complication sets in, if you put your global variables into
-; the AEL file, instead of the extensions.conf file. With clearglobalvars
-; set, a "reload" will often leave the globals vars cleared, because it
-; is not unusual to have extensions.conf (which will have no globals)
-; load after the extensions.ael file (where the global vars are stored).
-; So, with "reload" in this particular situation, first the AEL file will
-; clear and then set all the global vars, then, later, when the extensions.conf
-; file is loaded, the global vars are all cleared, and then not set, because
-; they are not stored in the extensions.conf file.
-;
-clearglobalvars=no
-;
-; If priorityjumping is set to 'yes', then applications that support
-; 'jumping' to a different priority based on the result of their operations
-; will do so (this is backwards compatible behavior with pre-1.2 releases
-; of Asterisk). Individual applications can also be requested to do this
-; by passing a 'j' option in their arguments.
-;
-;priorityjumping=yes
-;
-; User context is where entries from users.conf are registered.  The
-; default value is 'default'
-;
-;userscontext=default
-;
-; You can include other config files, use the #include command
-; (without the ';'). Note that this is different from the "include" command
-; that includes contexts within other contexts. The #include command works
-; in all asterisk configuration files.
-;#include "filename.conf"
-;#include <filename.conf>
-;#include filename.conf
-;
-; You can execute a program or script that produces config files, and they
-; will be inserted where you insert the #exec command. The #exec command 
-; works on all asterisk configuration files.  However, you will need to
-; activate them within asterisk.conf with the "execincludes" option.  They
-; are otherwise considered a security risk.
-;#exec /opt/bin/build-extra-contexts.sh
-;#exec /opt/bin/build-extra-contexts.sh --foo="bar"
-;#exec </opt/bin/build-extra-contexts.sh --foo="bar">
-;#exec "/opt/bin/build-extra-contexts.sh --foo=\"bar\""
-;
-
-; The "Globals" category contains global variables that can be referenced
-; in the dialplan with the GLOBAL dialplan function:
-; ${GLOBAL(VARIABLE)}
-; ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
-; Unix/Linux environmental variables can be reached with the ENV dialplan
-; function: ${ENV(VARIABLE)}
-;
-[globals]
-CONSOLE=Console/dsp				; Console interface for demo
-;CONSOLE=DAHDI/1
-;CONSOLE=Phone/phone0
-IAXINFO=guest					; IAXtel username/password
-;IAXINFO=myuser:mypass
-TRUNK=DAHDI/G2					; Trunk interface
-;
-; Note the 'G2' in the TRUNK variable above. It specifies which group (defined
-; in chan_dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use
-; in the specified group. The four possible options are:
-;
-; g: select the lowest-numbered non-busy DAHDI channel
-;    (aka. ascending sequential hunt group).
-; G: select the highest-numbered non-busy DAHDI channel
-;    (aka. descending sequential hunt group).
-; r: use a round-robin search, starting at the next highest channel than last
-;    time (aka. ascending rotary hunt group).
-; R: use a round-robin search, starting at the next lowest channel than last
-;    time (aka. descending rotary hunt group).
-;
-TRUNKMSD=1					; MSD digits to strip (usually 1 or 0)
-;TRUNK=IAX2/user:pass@provider
-
-;FREENUMDOMAIN=mydomain.com                     ; domain to send on outbound
-                                                ; freenum calls (uses outbound-freenum
-                                                ; context)
-
-;
-; WARNING WARNING WARNING WARNING
-; If you load any other extension configuration engine, such as pbx_ael.so,
-; your global variables may be overridden by that file.  Please take care to
-; use only one location to set global variables, and you will likely save
-; yourself a ton of grief.
-; WARNING WARNING WARNING WARNING
-;
-; Any category other than "General" and "Globals" represent 
-; extension contexts, which are collections of extensions.  
-;
-; Extension names may be numbers, letters, or combinations
-; thereof. If an extension name is prefixed by a '_'
-; character, it is interpreted as a pattern rather than a
-; literal.  In patterns, some characters have special meanings:
-;
-;   X - any digit from 0-9
-;   Z - any digit from 1-9
-;   N - any digit from 2-9
-;   [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
-;   . - wildcard, matches anything remaining (e.g. _9011. matches 
-;	anything starting with 9011 excluding 9011 itself)
-;   ! - wildcard, causes the matching process to complete as soon as
-;       it can unambiguously determine that no other matches are possible
-;
-; For example, the extension _NXXXXXX would match normal 7 digit dialings, 
-; while _1NXXNXXXXXX would represent an area code plus phone number
-; preceded by a one.
-;
-; Each step of an extension is ordered by priority, which must always start
-; with 1 to be considered a valid extension.  The priority "next" or "n" means
-; the previous priority plus one, regardless of whether the previous priority
-; was associated with the current extension or not.  The priority "same" or "s"
-; means the same as the previously specified priority, again regardless of
-; whether the previous entry was for the same extension.  Priorities may be
-; immediately followed by a plus sign and another integer to add that amount
-; (most useful with 's' or 'n').  Priorities may then also have an alias, or
-; label, in parentheses after their name which can be used in goto situations.
-;
-; Contexts contain several lines, one for each step of each extension.  One may
-; include another context in the current one as well, optionally with a date
-; and time.  Included contexts are included in the order they are listed.
-; Switches may also be included within a context.  The order of matching within
-; a context is always exact extensions, pattern match extensions, includes, and
-; switches.  Includes are always processed depth-first.  So for example, if you
-; would like a switch "A" to match before context "B", simply put switch "A" in
-; an included context "C", where "C" is included in your original context
-; before "B".
-;
-;[context]
-;exten => someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,...)
-;
-; Timing list for includes is 
-;
-;   <time range>,<days of week>,<days of month>,<months>[,<timezone>]
-;
-; Note that ranges may be specified to wrap around the ends.  Also, minutes are
-; fine-grained only down to the closest even minute.
-;
-;include => daytime,9:00-17:00,mon-fri,*,*
-;include => weekend,*,sat-sun,*,*
-;include => weeknights,17:02-8:58,mon-fri,*,*
-;
-; ignorepat can be used to instruct drivers to not cancel dialtone upon receipt
-; of a particular pattern.  The most commonly used example is of course '9'
-; like this:
-;
-;ignorepat => 9
-;
-; so that dialtone remains even after dialing a 9.  Please note that ignorepat
-; only works with channels which receive dialtone from the PBX, such as DAHDI,
-; Phone, and VPB.  Other channels, such as SIP and MGCP, which generate their
-; own dialtone and converse with the PBX only after a number is complete, are
-; generally unaffected by ignorepat (unless DISA or another method is used to
-; generate a dialtone after answering the channel).
-;
-
-;
-; Sample entries for extensions.conf
-;
-;
-[dundi-e164-canonical]
-;include => stdexten
-;
-; List canonical entries here
-;
-;exten => 12564286000,1,Gosub(6000,stdexten(IAX2/foo))
-;exten => 12564286000,n,Goto(default,s,1)	; exited Voicemail
-;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})
-
-[dundi-e164-customers]
-;
-; If you are an ITSP or Reseller, list your customers here.
-;
-;exten => _12564286000,1,Dial(SIP/customer1)
-;exten => _12564286001,1,Dial(IAX2/customer2)
-
-[dundi-e164-via-pstn]
-;
-; If you are freely delivering calls to the PSTN, list them here
-;
-;exten => _1256428XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428 
-;exten => _1256325XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Ditto for 256-325
-
-[dundi-e164-local]
-;
-; Context to put your dundi IAX2 or SIP user in for
-; full access
-;
-include => dundi-e164-canonical
-include => dundi-e164-customers
-include => dundi-e164-via-pstn
-
-[dundi-e164-switch]
-;
-; Just a wrapper for the switch
-;
-switch => DUNDi/e164
-
-[dundi-e164-lookup]
-;
-; Locally to lookup, try looking for a local E.164 solution
-; then try DUNDi if we don't have one.
-;
-include => dundi-e164-local
-include => dundi-e164-switch
-;
-; DUNDi can also be implemented as a Macro instead of using 
-; the Local channel driver. 
-;
-[macro-dundi-e164]
-;
-; ARG1 is the extension to Dial
-;
-; Extension "s" is not a wildcard extension that matches "anything".
-; In macros, it is the start extension. In most other cases, 
-; you have to goto "s" to execute that extension.
-;
-; For wildcard matches, see above - all pattern matches start with
-; an underscore.
-exten => s,1,Goto(${ARG1},1)
-include => dundi-e164-lookup
-
-;
-; Here are the entries you need to participate in the IAXTEL
-; call routing system.  Most IAXTEL numbers begin with 1-700, but
-; there are exceptions.  For more information, and to sign
-; up, please go to www.gnophone.com or www.iaxtel.com
-;
-[iaxtel700]
-exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)
-
-;
-; The SWITCH statement permits a server to share the dialplan with
-; another server. Use with care: Reciprocal switch statements are not
-; allowed (e.g. both A -> B and B -> A), and the switched server needs
-; to be on-line or else dialing can be severly delayed.
-;
-[iaxprovider]
-;switch => IAX2/user:[key]@myserver/mycontext
-
-[trunkint]
-;
-; International long distance through trunk
-;
-exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
-exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${FILTER(0-9,${EXTEN:${GLOBAL(TRUNKMSD)}})})
-
-[trunkld]
-;
-; Long distance context accessed through trunk
-;
-exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
-exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
-
-[trunklocal]
-;
-; Local seven-digit dialing accessed through trunk interface
-;
-exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
-
-[trunktollfree]
-;
-; Long distance context accessed through trunk interface
-;
-exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
-exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
-exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
-exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
-
-[international]
-;
-; Master context for international long distance
-;
-ignorepat => 9
-include => longdistance
-include => trunkint
-
-[longdistance]
-;
-; Master context for long distance
-;
-ignorepat => 9
-include => local
-include => trunkld
-
-[local]
-;
-; Master context for local, toll-free, and iaxtel calls only
-;
-ignorepat => 9
-include => default
-include => trunklocal
-include => iaxtel700
-include => trunktollfree
-include => iaxprovider
-
-;Include parkedcalls (or the context you define in features conf)
-;to enable call parking.
-include => parkedcalls
-;
-; You can use an alternative switch type as well, to resolve
-; extensions that are not known here, for example with remote 
-; IAX switching you transparently get access to the remote
-; Asterisk PBX
-; 
-; switch => IAX2/user:password@bigserver/local
-;
-; An "lswitch" is like a switch but is literal, in that
-; variable substitution is not performed at load time
-; but is passed to the switch directly (presumably to
-; be substituted in the switch routine itself)
-;
-; lswitch => Loopback/12${EXTEN}@othercontext
-;
-; An "eswitch" is like a switch but the evaluation of
-; variable substitution is performed at runtime before
-; being passed to the switch routine.
-;
-; eswitch => IAX2/context@${CURSERVER}
-
-; The following two contexts are a template to enable the ability to dial
-; ISN numbers. For more information about what an ISN number is, please see
-; http://www.freenum.org.
-;
-; This is the dialing hook.  use:
-; include => outbound-freenum
-
-[outbound-freenum]
-; We'll add more digits as needed. The purpose is to dial things
-; like extension numbers at domains (ITAD number) so we're matching
-; on lengths of 1 through 6 prior to the separator (the asterisk [*])
-;
-exten => _X*X!,1,Goto(outbound-freenum2,${EXTEN},1)
-exten => _XX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
-exten => _XXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
-exten => _XXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
-exten => _XXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
-exten => _XXXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
-
-[outbound-freenum2]
-; This is the handler which performs the dialing logic. It is called
-; from the [outbound-freenum] context
-;
-exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
-same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)})                                ; make sure the suffix is all digits as well
-same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION,1)
-                                                                        ; filter out bad characters per the README-SERIOUSLY.best-practices.txt document
-same => n,Set(TIMEOUT(absolute)=10800)
-same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)})     ; perform our lookup with freenum.org
-same => n,GotoIf($["${isnresult}" != ""]?from)
-same => n,Set(DIALSTATUS=CONGESTION)
-same => n,Goto(fn-CONGESTION,1)
-same => n(from),Set(SIPFROMUSER=${CALLERID(num)})
-same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial)               ; check if we set the FREENUMDOMAIN global variable in [global]
-same => n,Set(SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})                   ;    if we did set it, then we'll use it for our outbound dialing domain
-same => n(dial),Dial(SIP/${isnresult},40)
-same => n,Goto(fn-${DIALSTATUS},1)
-
-exten => fn-BUSY,1,Busy()
-
-exten => _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
-same => n,Congestion()
-
-[macro-trunkdial]
-;
-; Standard trunk dial macro (hangs up on a dialstatus that should 
-; terminate call)
-;   ${ARG1} - What to dial
-;
-exten => s,1,Dial(${ARG1})
-exten => s,n,Goto(s-${DIALSTATUS},1)
-exten => s-NOANSWER,1,Hangup
-exten => s-BUSY,1,Hangup
-exten => _s-.,1,NoOp
-
-[stdexten]
-;
-; Standard extension subroutine:
-;   ${EXTEN} - Extension
-;   ${ARG1} - Device(s) to ring
-;   ${ARG2} - Optional context in Voicemail (if empty, then "default")
-;
-; Note that the current version will drop through to the next priority in the
-; case of their pressing '#'.  This gives more flexibility in what do to next:
-; you can prompt for a new extension, or drop the call, or send them to a
-; general delivery mailbox, or...
-;
-; The use of the LOCAL() function is purely for convenience.  Any variable
-; initially declared as LOCAL() will disappear when the innermost Gosub context
-; in which it was declared returns.  Note also that you can declare a LOCAL()
-; variable on top of an existing variable, and its value will revert to its
-; previous value (before being declared as LOCAL()) upon Return.
-;
-exten => _X.,50000(stdexten),NoOp(Start stdexten)
-exten => _X.,n,Set(LOCAL(ext)=${EXTEN})
-exten => _X.,n,Set(LOCAL(dev)=${ARG1})
-exten => _X.,n,Set(LOCAL(cntx)=${ARG2})
-
-exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
-exten => _X.,n,Dial(${dev},20)			; Ring the interface, 20 seconds maximum
-exten => _X.,n,Goto(stdexten-${DIALSTATUS},1)		; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
-
-exten => stdexten-NOANSWER,1,Voicemail(${mbx},u)	; If unavailable, send to voicemail w/ unavail announce
-exten => stdexten-NOANSWER,n,NoOp(Finish stdexten NOANSWER)
-exten => stdexten-NOANSWER,n,Return()			; If they press #, return to start
-
-exten => stdexten-BUSY,1,Voicemail(${mbx},b)
-						; If busy, send to voicemail w/ busy announce
-exten => stdexten-BUSY,n,NoOp(Finish stdexten BUSY)
-exten => stdexten-BUSY,n,Return()			; If they press #, return to start
-
-exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1)	; Treat anything else as no answer
-
-exten => a,1,VoicemailMain(${mbx})		; If they press *, send the user into VoicemailMain
-exten => a,n,Return()
-
-[stdPrivacyexten]
-;
-; Standard extension subroutine:
-;   ${ARG1} - Extension
-;   ${ARG2} - Device(s) to ring
-;   ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 extension-priority)
-;   ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 extension-priority)`
-;   ${ARG5} - Context in voicemail (if empty, then "default")
-;
-; See above note in stdexten about priority handling on exit.
-;
-exten => _X.,60000(stdPrivacyexten),NoOp(Start stdPrivacyexten)
-exten => _X.,n,Set(LOCAL(ext)=${ARG1})
-exten => _X.,n,Set(LOCAL(dev)=${ARG2})
-exten => _X.,n,Set(LOCAL(dontcntx)=${ARG3})
-exten => _X.,n,Set(LOCAL(tortcntx)=${ARG4})
-exten => _X.,n,Set(LOCAL(cntx)=${ARG5})
-
-exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
-exten => _X.,n,Dial(${dev},20,p)			; Ring the interface, 20 seconds maximum, call screening 
-						; option (or use P for databased call _X.creening)
-exten => _X.,n,Goto(stdexten-${DIALSTATUS},1)		; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
-
-exten => stdexten-NOANSWER,1,Voicemail(${mbx},u)	; If unavailable, send to voicemail w/ unavail announce
-exten => stdexten-NOANSWER,n,NoOp(Finish stdPrivacyexten NOANSWER)
-exten => stdexten-NOANSWER,n,Return()			; If they press #, return to start
-
-exten => stdexten-BUSY,1,Voicemail(${mbx},b)		; If busy, send to voicemail w/ busy announce
-exten => stdexten-BUSY,n,NoOp(Finish stdPrivacyexten BUSY)
-exten => stdexten-BUSY,n,Return()			; If they press #, return to start
-
-exten => stdexten-DONTCALL,1,Goto(${dontcntx},s,1)	; Callee chose to send this call to a polite "Don't call again" script.
-
-exten => stdexten-TORTURE,1,Goto(${tortcntx},s,1)	; Callee chose to send this call to a telemarketer torture script.
-
-exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1)	; Treat anything else as no answer
-
-exten => a,1,VoicemailMain(${mbx})		; If they press *, send the user into VoicemailMain
-exten => a,n,Return
-
-[macro-page];
-;
-; Paging macro:
-;
-;       Check to see if SIP device is in use and DO NOT PAGE if they are
-;
-;   ${ARG1} - Device to page
-
-exten => s,1,ChanIsAvail(${ARG1},s)			; s is for ANY call
-exten => s,n,GoToIf([${AVAILORIGCHAN} = ""]?fail:autoanswer)
-exten => s,n(autoanswer),Set(_ALERT_INFO="RA")			; This is for the PolyComs
-exten => s,n,SIPAddHeader(Call-Info: Answer-After=0)	; This is for the Grandstream, Snoms, and Others
-exten => s,n,NoOp()					; Add others here and Post on the Wiki!!!!
-exten => s,n,Dial(${ARG1})
-exten => s,n(fail),Hangup
-
-
-[demo]
-include => stdexten
-;
-; We start with what to do when a call first comes in.
-;
-exten => s,1,Wait(1)			; Wait a second, just for fun
-exten => s,n,Answer			; Answer the line
-exten => s,n,Set(TIMEOUT(digit)=5)	; Set Digit Timeout to 5 seconds
-exten => s,n,Set(TIMEOUT(response)=10)	; Set Response Timeout to 10 seconds
-exten => s,n(restart),BackGround(demo-congrats)	; Play a congratulatory message
-exten => s,n(instruct),BackGround(demo-instruct)	; Play some instructions
-exten => s,n,WaitExten			; Wait for an extension to be dialed.
-
-exten => 2,1,BackGround(demo-moreinfo)	; Give some more information.
-exten => 2,n,Goto(s,instruct)
-
-exten => 3,1,Set(LANGUAGE()=fr)		; Set language to french
-exten => 3,n,Goto(s,restart)		; Start with the congratulations
-
-exten => 1000,1,Goto(default,s,1)
-;
-; We also create an example user, 1234, who is on the console and has
-; voicemail, etc.
-;
-exten => 1234,1,Playback(transfer,skip)		; "Please hold while..." 
-					; (but skip if channel is not up)
-exten => 1234,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}))
-exten => 1234,n,Goto(default,s,1)		; exited Voicemail
-
-exten => 1235,1,Voicemail(1234,u)		; Right to voicemail
-
-exten => 1236,1,Dial(Console/dsp)		; Ring forever
-exten => 1236,n,Voicemail(1234,b)		; Unless busy
-
-;
-; # for when they're done with the demo
-;
-exten => #,1,Playback(demo-thanks)	; "Thanks for trying the demo"
-exten => #,n,Hangup			; Hang them up.
-
-;
-; A timeout and "invalid extension rule"
-;
-exten => t,1,Goto(#,1)			; If they take too long, give up
-exten => i,1,Playback(invalid)		; "That's not valid, try again"
-
-;
-; Create an extension, 500, for dialing the
-; Asterisk demo.
-;
-exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
-exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default)	; Call the Asterisk demo
-exten => 500,n,Playback(demo-nogo)	; Couldn't connect to the demo site
-exten => 500,n,Goto(s,6)		; Return to the start over message.
-
-;
-; Create an extension, 600, for evaluating echo latency.
-;
-exten => 600,1,Playback(demo-echotest)	; Let them know what's going on
-exten => 600,n,Echo			; Do the echo test
-exten => 600,n,Playback(demo-echodone)	; Let them know it's over
-exten => 600,n,Goto(s,6)		; Start over
-
-;
-;	You can use the Macro Page to intercom a individual user
-exten => 76245,1,Macro(page,SIP/Grandstream1)
-; or if your peernames are the same as extensions
-exten => _7XXX,1,Macro(page,SIP/${EXTEN})
-;
-;
-; System Wide Page at extension 7999
-;
-exten => 7999,1,Set(TIMEOUT(absolute)=60)
-exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n,d)
-
-; Give voicemail at extension 8500
-;
-exten => 8500,1,VoicemailMain
-exten => 8500,n,Goto(s,6)
-;
-; Here's what a phone entry would look like (IXJ for example)
-;
-;exten => 1265,1,Dial(Phone/phone0,15)
-;exten => 1265,n,Goto(s,5)
-
-;
-;	The page context calls up the page macro that sets variables needed for auto-answer
-;	It is in is own context to make calling it from the Page() application as simple as 
-;	Local/{peername}@page
-;
-[page]
-exten => _X.,1,Macro(page,SIP/${EXTEN})
-
-;[mainmenu]
-;
-; Example "main menu" context with submenu
-;
-;exten => s,1,Answer
-;exten => s,n,Background(thanks)		; "Thanks for calling press 1 for sales, 2 for support, ..."
-;exten => s,n,WaitExten
-;exten => 1,1,Goto(submenu,s,1)
-;exten => 2,1,Hangup
-;include => default
-;
-;[submenu]
-;exten => s,1,Ringing					; Make them comfortable with 2 seconds of ringback
-;exten => s,n,Wait,2
-;exten => s,n,Background(submenuopts)	; "Thanks for calling the sales department.  Press 1 for steve, 2 for..."
-;exten => s,n,WaitExten
-;exten => 1,1,Goto(default,steve,1)
-;exten => 2,1,Goto(default,mark,2)
-
 [default]
-;
-; By default we include the demo.  In a production system, you 
-; probably don't want to have the demo there.
-;
-include => demo
-
-;
-; An extension like the one below can be used for FWD, Nikotel, sipgate etc.
-; Note that you must have a [sipprovider] section in sip.conf
-;
-;exten => _41X.,1,Dial(SIP/${FILTER(0-9,${EXTEN:2})}@sipprovider,,r)
-
-; Real extensions would go here. Generally you want real extensions to be
-; 4 or 5 digits long (although there is no such requirement) and start with a
-; single digit that is fairly large (like 6 or 7) so that you have plenty of
-; room to overlap extensions and menu options without conflict.  You can alias
-; them with names, too, and use global variables
-
-;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1(Joe Schmoe) ; Channel hints for presence
-;exten => 6245,1,Dial(SIP/Grandstream1,20,rt)	; permit transfer
-;exten => 6245,n(dial),Dial(${HINT},20,rtT)	; Use hint as listed
-;exten => 6245,n,Voicemail(6245,u)		; Voicemail (unavailable)
-;exten => 6245,s+1,Hangup			; s+1, same as n
-;exten => 6245,dial+101,Voicemail(6245,b)	; Voicemail (busy)
-;exten => 6361,1,Dial(IAX2/JaneDoe,,rm)		; ring without time limit
-;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14)
-;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
-;exten => 6391,1,Dial(JINGLE/asterisk@digium.com/mogorman@astjab.org) ;Dial via jingle using asterisk as the transport and calling mogorman.
-;exten => 6394,1,Dial(Local/6275/n)		; this will dial ${MARK}
-
-;exten => 6275,1,Gosub(${EXTEN},stdexten(${MARK}))
-						; assuming ${MARK} is something like DAHDI/2
-;exten => 6275,n,Goto(default,s,1)		; exited Voicemail
-;exten => mark,1,Goto(6275,1)			; alias mark to 6275
-;exten => 6536,1,Gosub(${EXTEN},stdexten(${WIL}))
-						; Ditto for wil
-;exten => 6536,n,Goto(default,s,1)		; exited Voicemail
-;exten => wil,1,Goto(6236,1)
-
-;If you want to subscribe to the status of a parking space, this is
-;how you do it. Subscribe to extension 6600 in sip, and you will see
-;the status of the first parking lot with this extensions' help
-;exten => 6600,hint,park:701@parkedcalls
-;exten => 6600,1,noop
-;
-; Some other handy things are an extension for checking voicemail via
-; voicemailmain
-;
-;exten => 8500,1,VoicemailMain
-;exten => 8500,n,Hangup
-;
-; Or a conference room (you'll need to edit meetme.conf to enable this room)
-;
-;exten => 8600,1,Meetme(1234)
-;
-; Or playing an announcement to the called party, as soon it answers
-;
-;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
-;
-
-; example of a compartmentalized company called "acme"
-;
-; this is the context that your incoming IAX/SIP trunk dumps you in...
-;[acme-incoming]
-;exten => s,1,Wait(1)
-;exten => s,n,Answer()
-;exten => s,n(menu),Playback(acme/vm-brief-menu)
-;exten => s,n(exten),Background(vm-enter-num-to-call)
-;exten => s,n,WaitExten(5)
-;exten => s,n(goodbye),Playback(vm-goodbye)
-;exten => s,n(end),Hangup()
-;
-;include  => acme-extens
-;
-;exten => i,1,Playback(vm-invalid)
-;exten => i,n,Goto(s,exten)			; optionally, transfer to operator
-;
-;exten => t,1,Goto(s,goodbye)
-;
-; this is the context our internal SIP hardphones use (see sip.conf)
-;
-;[acme-internal]
-;exten => s,1,Answer()
-;exten => s,n(exten),Background(vm-enter-num-to-call)
-;exten => s,n,WaitExten(5)
-;exten => s,n(goodbye),Playback(vm-goodbye)
-;exten => s,n(end),Hangup()
-;
-;include => trunkint
-;include => trunkld
-;include => trunklocal
-;
-;include => acme-extens
-;
-; you can test what your system sounds like to outside callers by dialing this
-;exten => 777,1,DISA(no-password,acme-incoming)
-;
-; grouping of acme's extensions... never used directly, always included.
-;
-;[acme-extens]
-;include => stdexten
-;exten => 111,1,Gosub(111,stdexten(SIP/pete_1,acme))
-;exten => 111,n,Goto(s,exten)
-;
-;exten => 112,1,Gosub(112,stdexten(SIP/nancy_1,acme))
-;exten => 112,n,Goto(s,end)
-;
-; end of acme example
-
-;
-; Time context: you can patch this in via the following.
-;
-; [acme-internal]
-; ...
-; exten => 777,1,Gosub(time)
-; exten => 777,n,Hangup()
-;
-; ...
-; include => time
-;
-; Note: if you're geographically spread out, you can have SIP extensions
-; specify their own local timezone in sip.conf as:
-;
-; [boi]
-; type=friend
-; context=acme-internal
-; callerid="Boise Ofc. <2083451111>"
-; ...
-; ; use system-wide default timezone of MST7MDT
-;
-; [lws]
-; type=friend
-; context=acme-internal
-; callerid="Lewiston Ofc. <2087431111>"
-; ...
-; setvar=timezone=PST8PDT
-;
-; "timezone" isn't a 'reserved' name in any way, and other places where
-; the timezone is significant (e.g. calls to "SayUnixTime()", etc) will
-; require modification as well.  Note that voicemail.conf already has
-; a mechanism for timezones.
-;
-
-[time]
-exten => _X.,30000(time),NoOp(Time: ${EXTEN} ${timezone})
-exten => _X.,n,Wait(0.25)
-exten => _X.,n,Answer()
-; the amount of delay is set for English; you may need to adjust this time
-; for other languages if there's no pause before the synchronizing beep.
-exten => _X.,n,Set(FUTURETIME=$[${EPOCH} + 12])
-exten => _X.,n,SayUnixTime(${FUTURETIME},Zulu,HNS)
-exten => _X.,n,SayPhonetic(z)
-; use the timezone associated with the extension (sip only), or system-wide
-; default if one hasn't been set.
-exten => _X.,n,SayUnixTime(${FUTURETIME},${timezone},HNS)
-exten => _X.,n,Playback(spy-local)
-exten => _X.,n,WaitUntil(${FUTURETIME})
-exten => _X.,n,Playback(beep)
-exten => _X.,n,Return()
-
-;
-; ANI context: use in the same way as "time" above
-;
-
-[ani]
-exten => _X.,40000(ani),NoOp(ANI: ${EXTEN})
-exten => _X.,n,Wait(0.25)
-exten => _X.,n,Answer()
-exten => _X.,n,Playback(vm-from)
-exten => _X.,n,SayDigits(${CALLERID(ani)})
-exten => _X.,n,Wait(1.25)
-exten => _X.,n,SayDigits(${CALLERID(ani)})	; playback again in case of missed digit
-exten => _X.,n,Return()
 
-; For more information on applications, just type "core show applications" at your
-; friendly Asterisk CLI prompt.
-;
-; "core show application <command>" will show details of how you
-; use that particular application in this file, the dial plan. 
-; "core show functions" will list all dialplan functions
-; "core show function <COMMAND>" will show you more information about
-; one function. Remember that function names are UPPER CASE.
+exten => 1199,1,Playback(demo-thanks)
+exten => 1199,n,Playback(demo-thanks)
+exten => 1199,n,Playback(demo-thanks)
/dev/pts/1
16:47:29
#~
Asterisk 1.6.2.9-2+squeeze3, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.2.9-2+squeeze3 currently running on linux1 (pid = 6558)
...
Supported: replaces, timer
Expires: 0
Date: Mon, 17 Oct 2011 15:11:34 GMT
ontent-Length: 0
<------------>
Scheduling destruction of SIP dialog '3c2670c32f19-2974qgqwtolq' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog '3c2670c32f19-2974qgqwtolq' Method: REGISTER
linux1*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
/dev/pts/7
16:47:29
#~
Asterisk 1.6.2.9-2+squeeze3, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.2.9-2+squeeze3 currently running on linux1 (pid = 6558)
...
Supported: replaces, timer
Expires: 0
Date: Mon, 17 Oct 2011 15:11:34 GMT
ontent-Length: 0
<------------>
Scheduling destruction of SIP dialog '3c2670c32f19-2974qgqwtolq' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog '3c2670c32f19-2974qgqwtolq' Method: REGISTER
linux1*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
прошло 27 минут
/dev/pts/1
17:15:04
#vi extensions.conf
/dev/pts/7
17:15:04
#vi extensions.conf
--- /tmp/l3-saved-3613.16880.30418	2011-10-17 18:15:10.000000000 +0300
+++ extensions.conf	2011-10-17 18:16:06.000000000 +0300
@@ -3,3 +3,6 @@
 exten => 1199,1,Playback(demo-thanks)
 exten => 1199,n,Playback(demo-thanks)
 exten => 1199,n,Playback(demo-thanks)
+
+
+exten => 1101,1,Dial(SIP/${EXTEN})
/dev/pts/1
17:17:01
#asterisk -rvvvvvvvvvvvvvvv
Asterisk 1.6.2.9-2+squeeze3, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.2.9-2+squeeze3 currently running on linux1 (pid = 6558)
...
Server: Asterisk PBX 1.6.2.9-2+squeeze3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
  == Spawn extension (default, 1102, 1) exited non-zero on 'SIP/1101-0000000f'
Really destroying SIP dialog '3c2671326c6e-yj1l4vrsnanx' Method: BYE
linux1*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
/dev/pts/7
17:17:01
#asterisk -rvvvvvvvvvvvvvvv
Asterisk 1.6.2.9-2+squeeze3, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.2.9-2+squeeze3 currently running on linux1 (pid = 6558)
...
Server: Asterisk PBX 1.6.2.9-2+squeeze3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
  == Spawn extension (default, 1102, 1) exited non-zero on 'SIP/1101-0000000f'
Really destroying SIP dialog '3c2671326c6e-yj1l4vrsnanx' Method: BYE
linux1*CLI>
Disconnected from Asterisk server
Executing last minute cleanups

Вторник (10/18/11)

/dev/pts/4
08:39:31
#tail -f /var/log/messages
Oct 18 09:38:14 linux1 dhcpd: Internet Systems Consortium DHCP Server 4.1.1-P1
Oct 18 09:38:14 linux1 dhcpd: Copyright 2004-2010 Internet Systems Consortium.
Oct 18 09:38:14 linux1 dhcpd: All rights reserved.
Oct 18 09:38:14 linux1 dhcpd: For info, please visit https://www.isc.org/software/dhcp/
Oct 18 09:38:14 linux1 dhcpd: Wrote 2 leases to leases file.
Oct 18 09:38:16 linux1 kernel: [   16.511752] sshd (1707): /proc/1707/oom_adj is deprecated, please use /proc/1707/oom_score_adj instead.
Oct 18 09:38:17 linux1 dhcpd: DHCPDISCOVER from 00:04:13:24:e5:7e via eth0
Oct 18 09:38:18 linux1 dhcpd: DHCPOFFER on 192.168.10.201 to 00:04:13:24:e5:7e via eth0
Oct 18 09:38:18 linux1 dhcpd: DHCPREQUEST for 192.168.10.201 (192.168.10.1) from 00:04:13:24:e5:7e via eth0
Oct 18 09:38:18 linux1 dhcpd: DHCPACK on 192.168.10.201 to 00:04:13:24:e5:7e via eth0
...
Oct 18 09:48:19 linux1 dhcpd: DHCPREQUEST for 192.168.10.201 from 00:04:13:24:e5:7e via eth0
Oct 18 09:48:19 linux1 dhcpd: DHCPACK on 192.168.10.201 to 00:04:13:24:e5:7e via eth0
Oct 18 09:48:47 linux1 kernel: [  644.951458] usb 2-1.8: USB disconnect, device number 3
Oct 18 09:50:59 linux1 dhcpd: DHCPINFORM from 192.168.10.200 via eth0: not authoritative for subnet 192.168.10.0
Oct 18 09:51:02 linux1 dhcpd: DHCPINFORM from 192.168.10.200 via eth0: not authoritative for subnet 192.168.10.0
Oct 18 09:51:10 linux1 dhcpd: DHCPREQUEST for 192.168.10.200 from f0:4d:a2:cc:4f:9b (sm-nb014) via eth0
Oct 18 09:51:10 linux1 dhcpd: DHCPACK on 192.168.10.200 to f0:4d:a2:cc:4f:9b (sm-nb014) via eth0
Oct 18 09:53:19 linux1 dhcpd: DHCPREQUEST for 192.168.10.201 from 00:04:13:24:e5:7e via eth0
Oct 18 09:53:19 linux1 dhcpd: DHCPACK on 192.168.10.201 to 00:04:13:24:e5:7e via eth0
^C^
/dev/pts/1
08:39:31
#tail -f /var/log/messages
Oct 18 09:38:14 linux1 dhcpd: Internet Systems Consortium DHCP Server 4.1.1-P1
Oct 18 09:38:14 linux1 dhcpd: Copyright 2004-2010 Internet Systems Consortium.
Oct 18 09:38:14 linux1 dhcpd: All rights reserved.
Oct 18 09:38:14 linux1 dhcpd: For info, please visit https://www.isc.org/software/dhcp/
Oct 18 09:38:14 linux1 dhcpd: Wrote 2 leases to leases file.
Oct 18 09:38:16 linux1 kernel: [   16.511752] sshd (1707): /proc/1707/oom_adj is deprecated, please use /proc/1707/oom_score_adj instead.
Oct 18 09:38:17 linux1 dhcpd: DHCPDISCOVER from 00:04:13:24:e5:7e via eth0
Oct 18 09:38:18 linux1 dhcpd: DHCPOFFER on 192.168.10.201 to 00:04:13:24:e5:7e via eth0
Oct 18 09:38:18 linux1 dhcpd: DHCPREQUEST for 192.168.10.201 (192.168.10.1) from 00:04:13:24:e5:7e via eth0
Oct 18 09:38:18 linux1 dhcpd: DHCPACK on 192.168.10.201 to 00:04:13:24:e5:7e via eth0
...
Oct 18 09:48:19 linux1 dhcpd: DHCPREQUEST for 192.168.10.201 from 00:04:13:24:e5:7e via eth0
Oct 18 09:48:19 linux1 dhcpd: DHCPACK on 192.168.10.201 to 00:04:13:24:e5:7e via eth0
Oct 18 09:48:47 linux1 kernel: [  644.951458] usb 2-1.8: USB disconnect, device number 3
Oct 18 09:50:59 linux1 dhcpd: DHCPINFORM from 192.168.10.200 via eth0: not authoritative for subnet 192.168.10.0
Oct 18 09:51:02 linux1 dhcpd: DHCPINFORM from 192.168.10.200 via eth0: not authoritative for subnet 192.168.10.0
Oct 18 09:51:10 linux1 dhcpd: DHCPREQUEST for 192.168.10.200 from f0:4d:a2:cc:4f:9b (sm-nb014) via eth0
Oct 18 09:51:10 linux1 dhcpd: DHCPACK on 192.168.10.200 to f0:4d:a2:cc:4f:9b (sm-nb014) via eth0
Oct 18 09:53:19 linux1 dhcpd: DHCPREQUEST for 192.168.10.201 from 00:04:13:24:e5:7e via eth0
Oct 18 09:53:19 linux1 dhcpd: DHCPACK on 192.168.10.201 to 00:04:13:24:e5:7e via eth0
^C^
прошло 15 минут
/dev/pts/4
08:54:55
#asterisk -rv
Asterisk 1.6.2.9-2+squeeze3, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.6.2.9-2+squeeze3 currently running on linux1 (pid = 1256)
Verbosity was 0 and is now 1
linux1*CLI> sip set debug
Disconnected from Asterisk server
Executing last minute cleanups

Файлы

  • sip.conf
  • sip.conf
    >
    ;
    ; SIP Configuration example for Asterisk
    ;
    ; SIP dial strings
    ;-----------------------------------------------------------
    ; In the dialplan (extensions.conf) you can use several
    ; syntaxes for dialing SIP devices.
    ;        SIP/devicename
    ;        SIP/username@domain   (SIP uri)
    ;        SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
    ;        SIP/devicename/extension
    ;
    ;
    ; 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
    ;
    ; 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
    ;
    ; 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
    ;
    ;   module reload chan_sip.so    Reload configuration file
    ;
    ;------- 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.
    ;-----------------------------------------------------------------------------
    ; ** Deprecated configuration options **
    ; The "call-limit" configuation option is deprecated. It still works in
    ; this version of Asterisk, but will disappear in the next version.
    ; 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)
    ;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)
    ;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
    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)
    ;
    ; 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
    ;tlscertfile=asterisk.pem       ; Certificate file (*.pem only) to use for TLS connections
                                    ; default is to look for "asterisk.pem" in current directory
    ;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.
    ;tlscadir=</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
    ;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)
    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 "no")
    ; See qos.tex or Quality of Service section of asterisk.pdf 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
    ;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
    ;qualifygap=100                 ; Number of milliseconds between each group of peers being qualified
    ;qualifypeers=1                 ; Number of peers in a group to be qualified at the same time
    ;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.
    ;vmexten=voicemail              ; dialplan extension to reach mailbox sets the
                                    ; Message-Account in the MWI notify message
                                    ; defaults to "asterisk"
    ;disallow=all                   ; First disallow all codecs
    ;allow=ulaw                     ; Allow codecs in order of preference
    ;allow=ilbc                     ; see doc/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
    ;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 "no" will stop any media before we have
                                    ; call progress. Default is "yes".
    ;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.
    ;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
    ;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.
    ;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
    ;                                               ; (could also be tcp,udp) - defining transports on the proxy line only
    ;                                               ; applies for the global proxy, otherwise use the transport= option
    ;matchexterniplocally = yes     ; Only substitute the externip 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.
    ; 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
    ;
    ; 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
    ;
    ;--------------------------- 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
    ;
    ;--------------------------- HASH TABLE SIZES ------------------------------------------------
    ; For maximum efficiency, adjust the following
    ; values to be slightly larger than the maximum number of in-memory objects (devices).
    ; Too large, and space is wasted. Too small, and things will run slower.
    ; 563 is probably way too big for small (home) applications, but it
    ; should cover most small/medium sites.
    ; It is recommended to make the sizes be a prime number!
    ; This was internally set to 17 for small-memory applications...
    ; All tables default to 563, except when compiled in LOW_MEMORY mode,
    ; in which case, they default to 17. You can override this by uncommenting
    ; the following, and changing the values.
    ;hash_users=563
    ;hash_peers=563
    ;hash_dialogs=563
    ;--------------------------- 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
    ; faxdetect = both              ; Enables both CNG and T.38 detection (same as 'yes')
    ;
    ;----------------------------------------- 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.
    ; Format for the mwi register statement is:
    ;       mwi => user[:secret[:authuser]]@host[:port][/mailbox]
    ;
    ; Examples:
    ;mwi => 1234:password@mysipprovider.com/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. "externip = 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 "bindport" value (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:
    ;
    ;        externip = 12.34.56.78          ; use this address.
    ;        externip = 12.34.56.78:9900     ; use this address and port.
    ;        externip = mynat.my.org:12600   ; Public address of my nat box.
    ;
    ;   b. "externhost = hostname[:port]" is similar to "externip" 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
    ;
    ;   c. "stunaddr = stun.server[:port]" queries the STUN server specified
    ;      as an argument to obtain the external address/port.
    ;      Queries are also sent periodically every "externrefresh" seconds
    ;      (as a side effect, sending the query also acts as a keepalive for
    ;      the state entry on the nat box):
    ;
    ;        stunaddr = foo.stun.com:3478
    ;        externrefresh = 15
    ;
    ;   Note that at the moment all these mechanism work only for the SIP socket.
    ;   The IP address discovered with externip/externhost/STUN 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 "externip" and
    ; "externhost" might not help you configure addresses properly, and you
    ; really need to use STUN.
    ;
    ; NOTE 2: when using "externip" or "externhost", the address part is
    ; also used as the external address for media sessions.
    ; If you use "stunaddr", STUN queries will be sent to the same server
    ; also from media sockets, and this should permit a correct mapping of
    ; the port numbers as well.
    ;
    ; 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                ; default. Use NAT mode only according to RFC3581 (;rport)
    ;        nat = yes               ; Always ignore info and assume NAT
    ;        nat = never             ; Never attempt NAT mode or RFC3581 support
    ;        nat = route             ; route = Assume NAT, don't send rport
    ;                                ; (work around more UNIDEN bugs)
    ;----------------------------------- 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).
    ;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.
                                    ; 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'.
    ;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.
    ;----------------------------------------- REALTIME SUPPORT ------------------------
    ; For additional information on ARA, the Asterisk Realtime Architecture,
    ; please read realtime.txt and extconfig.txt in the /doc directory of the
    ; source code.
    ;
    ;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.
    ;------------------------------ 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".
    ;-----------------------------------------------------------------------------------
    [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
    ;
    ; The SIP channel has two types of devices, the friend and the peer.
    ; * The type=friend is a device type that accepts both incoming and outbound calls,
    ;   where Asterisk match on the From: username on incoming calls.
    ;   (A synonym for friend is "user"). This is a type you use for your local
    ;   SIP phones.
    ; * The type=peer also handles both incoming and outbound calls. On inbound calls,
    ;   Asterisk only matches on IP/port, not on names. This is mostly used for SIP
    ;   trunks.
    ;
    ; 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).
    ;[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.
    ;
    ; 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
            nat=yes
            directmedia=no
            host=dynamic
    [public-phone](!,basic-options)   ; another template inheriting basic-options
            nat=no
            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
    ;nat=no                          ; there is not NAT between phone and Asterisk
    ;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
    ;nat=yes                         ; X-Lite is behind a NAT router
    ;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
    ;[cisco1]
    ;type=friend
    ;secret=blah
    ;qualify=200                     ; Qualify peer is no more than 200ms away
    ;nat=yes                         ; This phone may be natted
                                     ; Send SIP and RTP to the IP address that packet is
                                     ; received from instead of trusting SIP headers
    ;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.
    

    Статистика

    Время первой команды журнала14:09:15 2011-10-17
    Время последней команды журнала08:54:55 2011-10-18
    Количество командных строк в журнале101
    Процент команд с ненулевым кодом завершения, %11.88
    Процент синтаксически неверно набранных команд, % 0.00
    Суммарное время работы с терминалом *, час 2.09
    Количество командных строк в единицу времени, команда/мин 0.80
    Частота использования команд
    vi18|===============| 15.38%
    asterisk11|=========| 9.40%
    ping10|========| 8.55%
    cat8|======| 6.84%
    ifconfig6|=====| 5.13%
    '6|=====| 5.13%
    cp6|=====| 5.13%
    route6|=====| 5.13%
    apt-get6|=====| 5.13%
    sed6|=====| 5.13%
    ssh6|=====| 5.13%
    expand4|===| 3.42%
    cd4|===| 3.42%
    tail4|===| 3.42%
    /etc/init.d/isc-dhcp-server4|===| 3.42%
    ls4|===| 3.42%
    tcpdump2|=| 1.71%
    ifup2|=| 1.71%
    netstat2|=| 1.71%
    ~2|=| 1.71%
    ____
    *) 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

    $Id$