/l3/users/vadik/nt-voip/linux1.unix.nt/root :1 :2 :3 :4 :5 :6 :7 :8 :9 :10 :11 :12 :13 :14 :15 :16 :17 :18 :19 :20 |
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#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) |
#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) |
#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) |
#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) |
#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 |
#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 |
#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 |
#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 |
#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 |
#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 |
#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 |
#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 |
#route
![]() Kernel IP routing table Destination Gateway Genmask Flags Metric Ref Use Iface ^C |
#route
![]() Kernel IP routing table Destination Gateway Genmask Flags Metric Ref Use Iface ^C |
#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 |
#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 |
#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 |
#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 |
#apt-get install dhcp-server
![]() Чтение списков пакетов... Готово Построение дерева зависимостей Чтение информации о состоянии... Готово E: Не удалось найти пакет dhcp-server |
#apt-get install dhcp-server
![]() Чтение списков пакетов... Готово Построение дерева зависимостей Чтение информации о состоянии... Готово E: Не удалось найти пакет dhcp-server |
#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) ... |
#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) ... |
#cp dhcpd.conf dhcpd.conf0
|
#cp dhcpd.conf dhcpd.conf0
|
#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 |
#/etc/init.d/isc-dhcp-server restart
![]() Stopping ISC DHCP server: dhcpd failed! Starting ISC DHCP server: dhcpd. |
#/etc/init.d/isc-dhcp-server restart
Stopping ISC DHCP server: dhcpd failed! Starting ISC DHCP server: dhcpd. |
#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 |
#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 |
#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 |
#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 |
#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 |
#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 |
#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; |
#/etc/init.d/isc-dhcp-server restart
![]() Stopping ISC DHCP server: dhcpd. Starting ISC DHCP server: dhcpd. |
#/etc/init.d/isc-dhcp-server restart
Stopping ISC DHCP server: dhcpd. Starting ISC DHCP server: dhcpd. |
#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) ... |
#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) ... |
#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 |
#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 |
#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). |
#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). |
#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). |
#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). |
#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 |
#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 |
#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 |
#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 |
#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. |
#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. |
#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 |
#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 |
#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 |
#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 |
#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 |
#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 |
#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 |
#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 |
#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 |
#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 |
#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 |
#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 |
#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 |
#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 |
#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 |
#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) |
#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) |
#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 |
#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 |
#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) |
#~
![]() 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 |
#~
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 |
#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}) |
#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 |
#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 |
#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^ |
#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^ |
#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 |
; ; 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