Журнал лабораторных работ

Содержание

Журнал

Вторник (10/26/10)

/dev/pts/0
09:36:54
#rm /etc/asterisk/.sip.conf.sw*

09:38:09
#vi /etc/asterisk/extensions.conf
--- /tmp/l3-saved-3961.21773.23764	2010-10-26 10:44:46.000000000 +0300
+++ /etc/asterisk/extensions.conf	2010-10-26 10:46:40.000000000 +0300
@@ -554,7 +554,17 @@
 exten =>2299,n,Playback(demo-thanks)
 exten =>2299,n,Hangup
 
+exten => _22XX,1,Dial(SIP/${EXTEN})
+exten => _22XXX,1,Dial(SIP/1${EXTEN:1:4})
+
+exten => _68XX,1,Wait(2)
+exten => _68XX,n,Playback(/tmp/warning1)
+exten => _68XX,n,Dial(SIP/22${EXTEN:2})
+
+
 exten =>2201,1,Dial(SIP/2201)
 exten =>2202,2,DIal(SIP/2202)
 ;exten =>2201,1,Dial(SIP/2201)
 exten =>2298,1,Record(/tmp/warning1:gsm,,3)
+
+
прошла 21 минута
/dev/pts/2
09:59:41
#vi /etc/asterisk/extensions.conf
прошло 10 минут
/dev/pts/4
10:10:02
#ping 192.168.110.1
PING 192.168.110.1 (192.168.110.1) 56(84) bytes of data.
64 bytes from 192.168.110.1: icmp_seq=1 ttl=63 time=0.191 ms
64 bytes from 192.168.110.1: icmp_seq=2 ttl=63 time=0.220 ms
^C
--- 192.168.110.1 ping statistics ---
2 packets transmitted, 2 received, 0% packet loss, time 999ms
rtt min/avg/max/mdev = 0.191/0.205/0.220/0.020 ms
10:10:12
#ping 192.168.109.1
PING 192.168.109.1 (192.168.109.1) 56(84) bytes of data.
64 bytes from 192.168.109.1: icmp_seq=1 ttl=63 time=0.434 ms
^C
--- 192.168.109.1 ping statistics ---
1 packets transmitted, 1 received, 0% packet loss, time 0ms
rtt min/avg/max/mdev = 0.434/0.434/0.434/0.000 ms
10:10:16
#namp 192.168.109.1
bash: namp: command not found
10:10:22
#nmap 192.168.109.1
Starting Nmap 4.62 ( http://nmap.org ) at 2010-10-26 11:10 EEST
Interesting ports on 192.168.109.1:
Not shown: 1712 closed ports
PORT     STATE SERVICE
22/tcp   open  ssh
111/tcp  open  rpcbind
2000/tcp open  callbook
Nmap done: 1 IP address (1 host up) scanned in 0.077 seconds
10:10:26
#nmap 192.168.110.1
Starting Nmap 4.62 ( http://nmap.org ) at 2010-10-26 11:10 EEST
Interesting ports on 192.168.110.1:
Not shown: 1713 closed ports
PORT     STATE SERVICE
111/tcp  open  rpcbind
2000/tcp open  callbook
Nmap done: 1 IP address (1 host up) scanned in 0.087 seconds
10:10:35
#nmap 192.168.112.1
Starting Nmap 4.62 ( http://nmap.org ) at 2010-10-26 11:10 EEST
Interesting ports on 192.168.112.1:
Not shown: 1712 closed ports
PORT     STATE SERVICE
22/tcp   open  ssh
111/tcp  open  rpcbind
2000/tcp open  callbook
Nmap done: 1 IP address (1 host up) scanned in 0.197 seconds
10:10:48
#cat /etc/asterisk/sip.conf
[general]
context=default                 ; Default context for incoming calls
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
register=>mn:1234@192.168.111.1/bb
register=>mn:1234@192.168.110.1/br
register=>mn:1234@192.168.109.1/gm
[authentication]
...
[br]
type=friend
secret=1234
host=dynamic
username=mn
[gm]
type=friend
secret=1234
host=dynamic
username=mn
10:15:32
#vi /etc/asterisk/sip.conf
10:17:43
#asterisk -rx "sip reload"

/dev/pts/6
10:18:36
#asterisk -rcvvvv
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
    -- Executing [2202@default:2] Dial("SIP/2203-095c0140", "SIP/2202") in new stack
    -- Called 2202
    -- SIP/2202-095c8030 is ringing
    -- Got SIP response 486 "Busy Here" back from 192.168.112.2
    -- SIP/2202-095c8030 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
  == Auto fallthrough, channel 'SIP/2203-095c0140' status is 'BUSY'
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
10:24:10
#vi /etc/asterisk/sip.conf
--- /tmp/l3-saved-4784.12021.31995	2010-10-26 11:24:23.000000000 +0300
+++ /etc/asterisk/sip.conf	2010-10-26 11:25:45.000000000 +0300
@@ -26,6 +26,10 @@
 secret=1234
 host=dynamic
 
+[2204]
+type=friend
+secret=1234
+host=dynamic
 
 [bb]
 type=friend
10:25:45
#asterisk -rcv
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
    -- Added extension '_68XX' priority 3 to default
    -- Added extension '2201' priority 1 to default
    -- Added extension '2202' priority 2 to default
    -- Added extension '2203' priority 3 to default
    -- Added extension '2204' priority 4 to default
    -- Added extension '2298' priority 1 to default
  == Parsing '/etc/asterisk/users.conf': Found
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
/dev/pts/0
10:33:15
#asterisk -rcv
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
[Oct 26 11:34:36] NOTICE[2616]: chan_sip.c:7515 sip_reg_timeout:    -- Registration for 'mn@192.168.110.1' timed out, trying again (Attempt #3)
[Oct 26 11:34:56] NOTICE[2616]: chan_sip.c:7515 sip_reg_timeout:    -- Registration for 'mn@192.168.110.1' timed out, trying again (Attempt #4)
[Oct 26 11:35:16] NOTICE[2616]: chan_sip.c:7515 sip_reg_timeout:    -- Registration for 'mn@192.168.110.1' timed out, trying again (Attempt #5)
[Oct 26 11:35:36] NOTICE[2616]: chan_sip.c:7515 sip_reg_timeout:    -- Registration for 'mn@192.168.110.1' timed out, trying again (Attempt #6)
[Oct 26 11:35:56] NOTICE[2616]: chan_sip.c:7515 sip_reg_timeout:    -- Registration for 'mn@192.168.110.1' timed out, trying again (Attempt #7)
[Oct 26 11:36:16] NOTICE[2616]: chan_sip.c:7515 sip_reg_timeout:    -- Registration for 'mn@192.168.110.1' timed out, trying again (Attempt #8)
[Oct 26 11:36:36] NOTICE[2616]: chan_sip.c:7515 sip_reg_timeout:    -- Registration for 'mn@192.168.110.1' timed out, trying again (Attempt #9)
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
10:36:46
#ifconfig
eth0      Link encap:Ethernet  HWaddr 00:1b:fc:7d:bb:37
          inet addr:192.168.112.1  Bcast:192.168.112.255  Mask:255.255.255.0
          inet6 addr: fe80::21b:fcff:fe7d:bb37/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:18668 errors:0 dropped:0 overruns:0 frame:0
          TX packets:14422 errors:0 dropped:0 overruns:0 carrier:1
          collisions:0 txqueuelen:1000
          RX bytes:3574640 (3.4 MiB)  TX bytes:2151137 (2.0 MiB)
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:3488 errors:0 dropped:0 overruns:0 frame:0
          TX packets:3488 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:166174 (162.2 KiB)  TX bytes:166174 (162.2 KiB)
10:36:50
#asterisk -rcv
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
[Oct 26 11:38:15] NOTICE[2616]: chan_sip.c:7515 sip_reg_timeout:    -- Registration for 'mn@192.168.110.1' timed out, trying again (Attempt #1)
[Oct 26 11:38:35] NOTICE[2616]: chan_sip.c:7515 sip_reg_timeout:    -- Registration for 'mn@192.168.110.1' timed out, trying again (Attempt #2)
[Oct 26 11:38:55] NOTICE[2616]: chan_sip.c:7515 sip_reg_timeout:    -- Registration for 'mn@192.168.110.1' timed out, trying again (Attempt #3)
[Oct 26 11:39:09] NOTICE[2616]: chan_sip.c:15642 handle_request_register: Registration from 'sip:test@192.168.112.1' failed for '192.168.112.11' - No matching peer found
[Oct 26 11:39:15] NOTICE[2616]: chan_sip.c:7515 sip_reg_timeout:    -- Registration for 'mn@192.168.110.1' timed out, trying again (Attempt #4)
[Oct 26 11:39:35] NOTICE[2616]: chan_sip.c:7515 sip_reg_timeout:    -- Registration for 'mn@192.168.110.1' timed out, trying again (Attempt #5)
[Oct 26 11:39:55] NOTICE[2616]: chan_sip.c:7515 sip_reg_timeout:    -- Registration for 'mn@192.168.110.1' timed out, trying again (Attempt #6)
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
10:40:08
#ping 192.168.110.254
PING 192.168.110.254 (192.168.110.254) 56(84) bytes of data.
64 bytes from 192.168.110.254: icmp_seq=1 ttl=64 time=0.609 ms
64 bytes from 192.168.110.254: icmp_seq=2 ttl=64 time=0.591 ms
^C
--- 192.168.110.254 ping statistics ---
2 packets transmitted, 2 received, 0% packet loss, time 1004ms
rtt min/avg/max/mdev = 0.591/0.600/0.609/0.009 ms
10:40:18
#cat /etc/asterisk/sip.conf
[general]
context=default                 ; Default context for incoming calls
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
register=>mn:1234@192.168.111.1/bb
register=>mn:1234@192.168.110.1/br
register=>mn:1234@192.168.109.1/gm
[authentication]
...
[br]
type=friend
secret=1234
host=dynamic
username=mn
[gm]
type=friend
secret=1234
host=dynamic
username=mn
10:42:16
#asterisk -rcvv
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
br/mn                      (Unspecified)    D          0        Unmonitored
bb/mn                      192.168.111.1    D          5060     Unmonitored
2204/2204                  192.168.112.11   D          5060     Unmonitored
2203/2203                  192.168.112.2    D          5060     Unmonitored
2202/2202                  192.168.112.2    D          1193     Unmonitored
2201/2201                  (Unspecified)    D          0        Unmonitored
7 sip peers [Monitored: 0 online, 0 offline Unmonitored: 5 online, 2 offline]
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
/dev/pts/0
10:46:22
#asterisk -rcv
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
7 sip peers [Monitored: 0 online, 0 offline Unmonitored: 5 online, 2 offline]
    -- Executing [2204@default:1] Dial("SIP/2202-095b1150", "SIP/2204") in new stack
    -- Called 2204
    -- SIP/2204-095b52c0 is ringing
    -- SIP/2204-095b52c0 answered SIP/2202-095b1150
    -- Native bridging SIP/2202-095b1150 and SIP/2204-095b52c0
  == Spawn extension (default, 2204, 1) exited non-zero on 'SIP/2202-095b1150'
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
10:47:54
#vi /etc/asterisk/extensions.conf
--- /tmp/l3-saved-5131.9892.22288	2010-10-26 11:48:03.000000000 +0300
+++ /etc/asterisk/extensions.conf	2010-10-26 11:50:53.000000000 +0300
@@ -569,4 +569,7 @@
 ;exten =>2201,1,Dial(SIP/2201)
 exten =>2298,1,Record(/tmp/warning1:gsm,,3)
 
+exten =>_19XX,1,Dial,(SIP/ptr/${EXTEN})
+exten =>_20XX,1,Dial,(SIP/ptr/${EXTEN})
+exten =>_21XX,1,Dial,(SIP/ptr/${EXTEN})
 
10:50:53
#asterisk -rcv
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
br/mn                      192.168.110.1    D          5060     Unmonitored
bb/mn                      192.168.111.1    D          5060     Unmonitored
2204/2204                  192.168.112.11   D          5060     Unmonitored
2203/2203                  192.168.112.2    D          5060     Unmonitored
2202/2202                  192.168.112.2    D          1193     Unmonitored
2201/2201                  (Unspecified)    D          0        Unmonitored
7 sip peers [Monitored: 0 online, 0 offline Unmonitored: 6 online, 1 offline]
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
10:52:10
#vi /etc/asterisk/extensions.conf
--- /tmp/l3-saved-5131.2986.9744	2010-10-26 11:52:13.000000000 +0300
+++ /etc/asterisk/extensions.conf	2010-10-26 11:56:49.000000000 +0300
@@ -568,8 +568,12 @@
 exten =>2204,4,DIal(SIP/2204)
 ;exten =>2201,1,Dial(SIP/2201)
 exten =>2298,1,Record(/tmp/warning1:gsm,,3)
+;in Belorussia
+exten =>_19XX,1,Dial(SIP/ptr/${EXTEN})
+exten =>_20XX,1,Dial(SIP/ptr/${EXTEN})
+exten =>_21XX,1,Dial(SIP/ptr/${EXTEN})
+;in other locations
+exten =>_21XX,1,Dial(SIP/ptr/${EXTEN})
+exten =>_XXXX,1,Dial(SIP/ptr/${EXTEN})
 
-exten =>_19XX,1,Dial,(SIP/ptr/${EXTEN})
-exten =>_20XX,1,Dial,(SIP/ptr/${EXTEN})
-exten =>_21XX,1,Dial,(SIP/ptr/${EXTEN})
 
10:56:49
#asterisk -rcv
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
  == Everyone is busy/congested at this time (1:0/0/1)
  == Auto fallthrough, channel 'SIP/2202-095b1150' status is 'CHANUNAVAIL'
    -- Executing [2102@default:1] Dial("SIP/2202-095ba3c8", "SIP/ptr/2102") in new stack
[Oct 26 11:57:45] WARNING[5264]: chan_sip.c:2921 create_addr: No such host: ptr
[Oct 26 11:57:45] WARNING[5264]: app_dial.c:1202 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
  == Auto fallthrough, channel 'SIP/2202-095ba3c8' status is 'CHANUNAVAIL'
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
10:57:52
#vi /etc/asterisk/extensions.conf
--- /tmp/l3-saved-5131.15118.16460	2010-10-26 11:57:54.000000000 +0300
+++ /etc/asterisk/extensions.conf	2010-10-26 11:59:13.000000000 +0300
@@ -569,11 +569,11 @@
 ;exten =>2201,1,Dial(SIP/2201)
 exten =>2298,1,Record(/tmp/warning1:gsm,,3)
 ;in Belorussia
-exten =>_19XX,1,Dial(SIP/ptr/${EXTEN})
-exten =>_20XX,1,Dial(SIP/ptr/${EXTEN})
-exten =>_21XX,1,Dial(SIP/ptr/${EXTEN})
+exten =>_19XX,1,Dial(SIP/gm/${EXTEN})
+exten =>_20XX,1,Dial(SIP/br/${EXTEN})
+exten =>_21XX,1,Dial(SIP/bb/${EXTEN})
 ;in other locations
-exten =>_21XX,1,Dial(SIP/ptr/${EXTEN})
-exten =>_XXXX,1,Dial(SIP/ptr/${EXTEN})
+;exten =>_21XX,1,Dial(SIP/ptr/${EXTEN})
+exten =>_XXXX,1,Dial(SIP/bb/${EXTEN})
 
 
10:59:13
#asterisk -rcv
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
    -- Executing [2204@default:1] Dial("SIP/mn-095b52c0", "SIP/2204") in new stack
    -- Called 2204
    -- SIP/2204-095c0a38 is ringing
    -- SIP/2204-095c0a38 answered SIP/mn-095b52c0
    -- Native bridging SIP/mn-095b52c0 and SIP/2204-095c0a38
  == Spawn extension (default, 2204, 1) exited non-zero on 'SIP/mn-095b52c0'
[Oct 26 12:00:09] NOTICE[2616]: chan_sip.c:14668 handle_request_invite: Unable to create/find SIP channel for this INVITE
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
11:00:24
#vi /etc/asterisk/sip.conf
--- /tmp/l3-saved-5131.12803.5907	2010-10-26 12:00:45.000000000 +0300
+++ /etc/asterisk/sip.conf	2010-10-26 12:02:55.000000000 +0300
@@ -15,21 +15,25 @@
 type=friend
 secret=1234
 host=dynamic
+callerid="Minsk 2201"
 
 [2202]
 type=friend
 secret=1234
 host=dynamic
+callerid="Minsk Softphone 2202"
 
 [2203]
 type=friend
 secret=1234
 host=dynamic
+callerid="Minsk Softphone 2203"
 
 [2204]
 type=friend
 secret=1234
 host=dynamic
+callerid="Minsk 2204"
 
 [bb]
 type=friend
11:02:55
#asterisk -rcv
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
    -- Executing [2203@default:1] Dial("SIP/2202-095bcee8", "SIP/2203") in new stack
    -- Called 2203
    -- SIP/2203-095be478 is ringing
    -- Got SIP response 486 "Busy Here" back from 192.168.112.2
    -- SIP/2203-095be478 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
  == Auto fallthrough, channel 'SIP/2202-095bcee8' status is 'BUSY'
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
11:03:57
#vi /etc/asterisk/sip.conf
прошло 14 минут
11:18:56
#asterisk -rcv
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
  == Spawn extension (default, 2204, 1) exited non-zero on 'SIP/2203-095ba3c8'
    -- Executing [2203@default:1] Dial("SIP/2204-095b6cc8", "SIP/2203") in new stack
    -- Called 2203
    -- SIP/2203-095b52c0 is ringing
    -- SIP/2203-095b52c0 answered SIP/2204-095b6cc8
    -- Native bridging SIP/2204-095b6cc8 and SIP/2203-095b52c0
  == Spawn extension (default, 2203, 1) exited non-zero on 'SIP/2204-095b6cc8'
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
11:21:30
#less
11:21:34
#less /etc/asterisk/extensions.conf
11:24:19
#asterisk -rsv
asterisk: invalid option -- s
11:28:07
#asterisk -rcv
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
br/mn                      192.168.110.1    D          5060     Unmonitored
bb/mn                      192.168.111.1    D          5060     Unmonitored
2204/2204                  192.168.112.11   D          5060     Unmonitored
2203/2203                  192.168.112.2    D          5060     Unmonitored
2202/2202                  192.168.112.2    D          1193     Unmonitored
2201/2201                  (Unspecified)    D          0        Unmonitored
7 sip peers [Monitored: 0 online, 0 offline Unmonitored: 6 online, 1 offline]
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
11:28:54
#ifconfig
eth0      Link encap:Ethernet  HWaddr 00:1b:fc:7d:bb:37
          inet addr:192.168.112.1  Bcast:192.168.112.255  Mask:255.255.255.0
          inet6 addr: fe80::21b:fcff:fe7d:bb37/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:26364 errors:0 dropped:0 overruns:0 frame:0
          TX packets:20975 errors:0 dropped:0 overruns:0 carrier:1
          collisions:0 txqueuelen:1000
          RX bytes:5018772 (4.7 MiB)  TX bytes:3613270 (3.4 MiB)
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:3503 errors:0 dropped:0 overruns:0 frame:0
          TX packets:3503 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:172334 (168.2 KiB)  TX bytes:172334 (168.2 KiB)
11:28:57
#asterisk -rcv
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
br/mn                      192.168.110.1    D          5060     Unmonitored
bb/mn                      192.168.111.1    D          5060     Unmonitored
2204/2204                  192.168.112.11   D          5060     Unmonitored
2203/2203                  192.168.112.2    D          5060     Unmonitored
2202/2202                  192.168.112.2    D          1193     Unmonitored
2201/2201                  (Unspecified)    D          0        Unmonitored
7 sip peers [Monitored: 0 online, 0 offline Unmonitored: 6 online, 1 offline]
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
11:32:13
#asterisk -rcv
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
br/mn                      192.168.110.1    D          5060     Unmonitored
bb/mn                      192.168.111.1    D          5060     Unmonitored
2204/2204                  192.168.112.11   D          5060     Unmonitored
2203/2203                  192.168.112.2    D          5060     Unmonitored
2202/2202                  192.168.112.2    D          1193     Unmonitored
2201/2201                  (Unspecified)    D          0        Unmonitored
7 sip peers [Monitored: 0 online, 0 offline Unmonitored: 6 online, 1 offline]
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
11:32:45
#vi /etc/asterisk/extensions.conf
--- /tmp/l3-saved-5131.2799.17595	2010-10-26 12:32:52.000000000 +0300
+++ /etc/asterisk/extensions.conf	2010-10-26 12:36:59.000000000 +0300
@@ -565,9 +565,9 @@
 exten =>2201,1,Dial(SIP/2201,1)
 exten =>2201,n,Dial(SIP/2204)
 
-exten =>2202,1,DIal(SIP/2202)
-exten =>2203,1,DIal(SIP/2203)
-exten =>2204,1,DIal(SIP/2204)
+;exten =>2202,1,Dial(SIP/2202)
+exten =>2203,1,Dial(SIP/2203)
+exten =>2204,1,Dial(SIP/2204)
 ;exten =>2201,1,Dial(SIP/2201)
 exten =>2298,1,Record(/tmp/warning1:gsm,,3)
 ;in Belorussia
@@ -578,4 +578,6 @@
 ;exten =>_21XX,1,Dial(SIP/ptr/${EXTEN})
 exten =>_XXXX,1,Dial(SIP/bb/${EXTEN})
 
+[group1]
+exten =>2202,1,Dial(SIP/2202)
 
/dev/pts/0
11:37:48
#vi /etc/asterisk/sip.conf
--- /tmp/l3-saved-5705.18323.18026	2010-10-26 12:37:55.000000000 +0300
+++ /etc/asterisk/sip.conf	2010-10-26 12:38:42.000000000 +0300
@@ -22,6 +22,7 @@
 secret=1234
 host=dynamic
 callerid="Minsk Softphone 2202"
+context=group1                 ; group1 context for incoming calls
 
 [2203]
 type=friend
11:38:42
#vi /etc/asterisk/extensions.conf
11:39:26
#exten =>2202,1,Dial(SIP/2202)
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
    -- Executing [2202@default:1] Dial("SIP/2204-095b52c0", "SIP/2202") in new stack
    -- Called 2202
    -- SIP/2202-095be090 is ringing
    -- Got SIP response 486 "Busy Here" back from 192.168.112.2
    -- SIP/2202-095be090 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
  == Auto fallthrough, channel 'SIP/2204-095b52c0' status is 'BUSY'
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
11:40:01
#vi /etc/asterisk/sip.conf
11:47:29
#vi /etc/asterisk/sip.conf
11:47:37
#[2202]
;exten =>2202,1,Dial(SIP/2202)
exten =>2203,1,Dial(SIP/2203)
exten =>2204,1,Dial(SIP/2204)
;exten =>2201,1,Dial(SIP/2201)
exten =>2298,1,Record(/tmp/warning1:gsm,,3)
;in Belorussia
;exten =>_19XX,1,Dial(SIP/gm/${EXTEN})
;exten =>_20XX,1,Dial(SIP/br/${EXTEN})
;exten =>_21XX,1,Dial(SIP/bb/${EXTEN})
;in other locations
;exten =>_21XX,1,Dial(SIP/ptr/${EXTEN})
;exten =>_XXXX,1,Dial(SIP/bb/${EXTEN})
[local]
exten =>_22XX,1,Dial(SIP/mn/${EXTEN})
[national]
exten =>_19XX,1,Dial(SIP/gm/${EXTEN})
exten =>_20XX,1,Dial(SIP/br/${EXTEN})
exten =>_21XX,1,Dial(SIP/bb/${EXTEN})
[international]
11:49:16
#{EXTEN})
E325: ATTENTION
Found a swap file by the name "/etc/asterisk/.sip.conf.swp"
          owned by: root   dated: Tue Oct 26 12:37:23 2010
         file name: /etc/asterisk/sip.conf
          modified: YES
         user name: root   host name: linux12.unix.nt
        process ID: 5639
While opening file "/etc/asterisk/sip.conf"
             dated: Tue Oct 26 12:38:42 2010
      NEWER than swap file!
...
[2204]ynamic
[2204]
secret=1234i
host=dynamic
callerid="Minsk 2204"
context=group1
[bb]
[bb]=friend
secret=1234
"/etc/asterisk/sip.conf" 61L, 1141C written
/dev/pts/2
11:53:27
#vi /etc/asterisk/extensions.conf
--- /tmp/l3-saved-5980.12795.17794	2010-10-26 12:53:36.000000000 +0300
+++ /etc/asterisk/extensions.conf	2010-10-26 12:58:43.000000000 +0300
@@ -588,3 +588,15 @@
 
 [international]
 exten =>_XXXX,1,Dial(SIP/bb/${EXTEN})
+
+[group1]
+include => local
+include => national
+include => international
+
+[group2]
+include => local
+include => national
+
+[group3]
+include =>local
11:58:43
#vi /etc/asterisk/extensions.conf
/dev/pts/0
12:00:41
#asterisk -rcv
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
br/mn                      192.168.110.1    D          5060     Unmonitored
bb/mn                      192.168.111.1    D          5060     Unmonitored
2204/2204                  192.168.112.11   D          5060     Unmonitored
2203/2203                  192.168.112.2    D          5060     Unmonitored
2202/2202                  192.168.112.2    D          1193     Unmonitored
2201/2201                  (Unspecified)    D          0        Unmonitored
7 sip peers [Monitored: 0 online, 0 offline Unmonitored: 6 online, 1 offline]
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
прошло 38 минут
12:38:59
#cat /etc/asterisk/extensions.conf
; extensions.conf - the Asterisk dial plan
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "dialplan reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI
;
...
exten =>_XXXX,1,Dial(SIP/bb/${EXTEN})
[group1]
include => local
include => national
include => international
[group2]
include => local
include => national
[group3]
include =>local
12:39:06
#cat /etc/asterisk/sip.conf
[general]
context=default                 ; Default context for incoming calls
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
register=>mn:1234@192.168.111.1/bb
register=>mn:1234@192.168.110.1/br
register=>mn:1234@192.168.109.1/gm
[authentication]
...
[br]
type=friend
secret=1234
host=dynamic
username=mn
[gm]
type=friend
secret=1234
host=dynamic
username=mn
12:39:18
#asterisk -rcv
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
br/mn                      192.168.110.1    D          5060     Unmonitored
bb/mn                      192.168.111.1    D          5060     Unmonitored
2204/2204                  192.168.112.11   D          5060     Unmonitored
2203/2203                  192.168.112.2    D          5060     Unmonitored
2202/2202                  192.168.112.2    D          1695     Unmonitored
2201/2201                  (Unspecified)    D          0        Unmonitored
7 sip peers [Monitored: 0 online, 0 offline Unmonitored: 6 online, 1 offline]
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
прошло 13 минут
/dev/pts/0
12:53:01
#vi /etc/asterisk/extensions.conf
--- /tmp/l3-saved-6274.580.5719	2010-10-26 13:53:09.000000000 +0300
+++ /etc/asterisk/extensions.conf	2010-10-26 13:57:43.000000000 +0300
@@ -579,7 +579,7 @@
 ;exten =>_XXXX,1,Dial(SIP/bb/${EXTEN})
 
 [local]
-exten =>_22XX,1,Dial(SIP/mn/${EXTEN})
+exten =>_22XX,1,Dial(SIP/${EXTEN})
 
 [national]
 exten =>_19XX,1,Dial(SIP/gm/${EXTEN})
@@ -599,4 +599,4 @@
 include => national
 
 [group3]
-include =>local
+include => local
/dev/pts/2
12:53:51
#vi /etc/asterisk/sip.conf
/dev/pts/0
12:57:43
#asterisk -rcv
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
    -- Executing [2001@group1:1] Dial("SIP/2204-095ba3c8", "SIP/br/2001") in new stack
    -- Called br/2001
    -- SIP/br-095c1e88 is ringing
    -- SIP/br-095c1e88 answered SIP/2204-095ba3c8
    -- Native bridging SIP/2204-095ba3c8 and SIP/br-095c1e88
  == Spawn extension (group1, 2001, 1) exited non-zero on 'SIP/2204-095ba3c8'
[Oct 26 14:01:35] NOTICE[2616]: chan_sip.c:14668 handle_request_invite: Unable to create/find SIP channel for this INVITE
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
13:02:28
#less /etc/asterisk/extensions.conf
/dev/pts/2
13:10:59
#[2202]
dbus-1/              default/             dhcp3/
debconf.conf         defoma/              dictionaries-common/
debian_version       deluser.conf         dpkg/
13:10:59
#ls /etc/dhcp3/dhc
dhclient.conf           dhclient-exit-hooks.d/
dhclient-enter-hooks.d/ dhcpd.conf
13:10:59
#ls /etc/dhcp3/dhc
dhclient.conf           dhclient-exit-hooks.d/
dhclient-enter-hooks.d/ dhcpd.conf
13:10:59
#less /etc/dhcp3/dhcpd.conf
13:11:42
#less /etc/asterisk/extensions.conf
прошло 10 минут
/dev/pts/0
13:21:56
#vi /etc/asterisk/extensions.conf
--- /tmp/l3-saved-6274.20962.17902	2010-10-26 14:22:16.000000000 +0300
+++ /etc/asterisk/extensions.conf	2010-10-26 14:22:58.000000000 +0300
@@ -569,7 +569,7 @@
 exten =>2203,1,Dial(SIP/2203)
 exten =>2204,1,Dial(SIP/2204)
 ;exten =>2201,1,Dial(SIP/2201)
-exten =>2298,1,Record(/tmp/warning1:gsm,,3)
+;exten =>2298,1,Record(/tmp/warning1:gsm,,3)
 ;in Belorussia
 ;exten =>_19XX,1,Dial(SIP/gm/${EXTEN})
 ;exten =>_20XX,1,Dial(SIP/br/${EXTEN})
@@ -580,6 +580,7 @@
 
 [local]
 exten =>_22XX,1,Dial(SIP/${EXTEN})
+exten =>2298,1,Record(/tmp/warning1:gsm,,3)
 
 [national]
 exten =>_19XX,1,Dial(SIP/gm/${EXTEN})
13:24:36
#vi /etc/asterisk/extensions.conf
--- /tmp/l3-saved-6274.27689.14722	2010-10-26 14:24:37.000000000 +0300
+++ /etc/asterisk/extensions.conf	2010-10-26 14:25:25.000000000 +0300
@@ -580,7 +580,7 @@
 
 [local]
 exten =>_22XX,1,Dial(SIP/${EXTEN})
-exten =>2298,1,Record(/tmp/warning1:gsm,,3)
+;exten =>2298,1,Record(/tmp/warning1:gsm,,3)
 
 [national]
 exten =>_19XX,1,Dial(SIP/gm/${EXTEN})
13:25:25
#vi /etc/asterisk/extensions.conf
--- /tmp/l3-saved-6274.26118.31391	2010-10-26 14:26:15.000000000 +0300
+++ /etc/asterisk/extensions.conf	2010-10-26 14:27:08.000000000 +0300
@@ -569,7 +569,7 @@
 exten =>2203,1,Dial(SIP/2203)
 exten =>2204,1,Dial(SIP/2204)
 ;exten =>2201,1,Dial(SIP/2201)
-;exten =>2298,1,Record(/tmp/warning1:gsm,,3)
+exten =>2298,1,Record(/tmp/warning1:gsm,,3)
 ;in Belorussia
 ;exten =>_19XX,1,Dial(SIP/gm/${EXTEN})
 ;exten =>_20XX,1,Dial(SIP/br/${EXTEN})
13:27:08
#vi /etc/asterisk/extensions.conf
13:29:22
#include => local
SoX(1)                                                                Sound eXchange                                                               SoX(1)
NAME
       SoX - Sound eXchange, the Swiss Army knife of audio manipulation
SYNOPSIS
       sox [global-options] [format-options] infile1
           [[format-options] infile2] ... [format-options] outfile
           [effect [effect-options]] ...
       play [global-options] [format-options] infile1
           [[format-options] infile2] ... [format-options]
           [effect [effect-options]] ...
...
            rec -M take1.aiff take1-dub.aiff
       records a new track in a multi-track recording.
       Further examples are included throughout this manual; more-detailed examples can be found in soxexam(7).
   File Formats
       There are two types of audio file format that SoX can work with.  The first is ‘self-describing’; these formats include a header  that  completely
       describes  the  characteristics  of  the  audio  data  that  follows.   The  second  type  is  ‘headerless’  (or ‘raw data’); here, the audio data
       characteristics must be described using the SoX command line.
       The following four characteristics are sufficient to describe the format of audio data such that it can be processed with SoX:
       sample rate
              The sample rate in samples per second (‘Hertz’ or ‘Hz’).  For example, digital telephony  traditionally  uses  a  sample  rate  of  8000 Hz
13:30:27
#man play
13:30:34
#ls /tmp/
.ICE-unix/                l3-saved-2913.28927.9583  lost+found/               .X11-unix/
l3-saved-2758.1533.12585  l3-saved-5131.5837.12387  warning1.gsm
13:30:34
#play /tmp/warning1.gsm
play soxio: Can't open input file `/tmp/warning1.gsm': unknown file type `gsm'
13:30:57
#vi /etc/asterisk/extensions.conf
--- /tmp/l3-saved-6274.31999.15031	2010-10-26 14:31:42.000000000 +0300
+++ /etc/asterisk/extensions.conf	2010-10-26 14:32:21.000000000 +0300
@@ -569,7 +569,7 @@
 exten =>2203,1,Dial(SIP/2203)
 exten =>2204,1,Dial(SIP/2204)
 ;exten =>2201,1,Dial(SIP/2201)
-exten =>2298,1,Record(/tmp/warning1:gsm,,3)
+exten => 2298,1,Record(/tmp/warning1:gsm,,3)
 ;in Belorussia
 ;exten =>_19XX,1,Dial(SIP/gm/${EXTEN})
 ;exten =>_20XX,1,Dial(SIP/br/${EXTEN})
@@ -580,7 +580,7 @@
 
 [local]
 exten =>_22XX,1,Dial(SIP/${EXTEN})
-;exten =>2298,1,Record(/tmp/warning1:gsm,,3)
+;exten => 2298,1,Record(/tmp/warning1:gsm,,3)
 
 [national]
 exten =>_19XX,1,Dial(SIP/gm/${EXTEN})
13:32:21
#asterisk -rcx "dialplan reload"
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
[ Booting...
[ Reading Master Configuration ]
[ Initializing Custom Configuration Options ]
Connected to Asterisk 1.4.21.2~dfsg-3+lenny1 currently running on linux12 (pid = 2552)
Dialplan reloaded.
13:34:55
#asterisk -rcx "dialplan reload"
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
[ Booting...
[ Reading Master Configuration ]
[ Initializing Custom Configuration Options ]
Connected to Asterisk 1.4.21.2~dfsg-3+lenny1 currently running on linux12 (pid = 2552)
Dialplan reloaded.
13:38:24
#uptime
 14:39:00 up  5:00,  2 users,  load average: 0.01, 0.01, 0.00
прошло 11 минут
/dev/pts/2
13:50:22
#vi /etc/asterisk/extensions.conf
--- /tmp/l3-saved-7213.24362.11361	2010-10-26 14:50:37.000000000 +0300
+++ /etc/asterisk/extensions.conf	2010-10-26 14:54:41.000000000 +0300
@@ -581,7 +581,7 @@
 ;exten =>_XXXX,1,Dial(SIP/bb/${EXTEN})
 
 [local]
-exten =>_22XX,1,Dial(SIP/${EXTEN})
+;exten =>_22XX,1,Dial(SIP/${EXTEN})
 exten => 2298,1,Record(/tmp/warning1:gsm,,3)
 exten => 2297,1,Playback(/tmp/warning1)
 
13:54:41
#asterisk -rx "dialplan reload"
Dialplan reloaded.
13:55:53
#include => local
;
; 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
...
[international]
exten =>_XXXX,1,Dial(SIP/bb/${EXTEN})
[group1]
include => local
include => national
include => international
[group2]
include => local
include => national
[group3]
13:56:10
#include => local
[general]
Found a swap file by the name "/etc/asterisk/.sip.conf.swp"
context=default                 ; Default context for incoming calls
          owned by: root   dated: Tue Oct 26 12:37:23 2010
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
         file name: /etc/asterisk/sip.conf
bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)
          modified: YES
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
         user name: root   host name: linux12.unix.nt
...
callerid="Minsk Softphone 2203"
cont
[2204]
[2204]riend
secret=1234
callerid="Minsk 2204"
context=group1
[bb]
type=friend
secret=1234                                                                                                                                42,0-1        Top
13:56:47
#[bb]
exten => 2298,1,Record(/tmp/warning1:gsm,,3)
exten => 2297,1,Playback(/tmp/warning1)
[national]
exten =>_19XX,1,Dial(SIP/gm/${EXTEN})
exten =>_20XX,1,Dial(SIP/br/${EXTEN})
exten =>_21XX,1,Dial(SIP/bb/${EXTEN})
[international]
exten =>_XXXX,1,Dial(SIP/bb/${EXTEN})
[group1]
include => local
...
;exten =>2299,n,Playback(demo-thanks)
exten =>2299,n,Hangup
;exten => 2297,1,Playback(/tmp/warning1)
exten => _22XX,1,Dial(SIP/${EXTEN})
exten => _22XXX,1,Dial(SIP/1${EXTEN:1:4})
exten => _68XX,1,Wait(2)
include => local
[local]
[        ]
"/etc/asterisk/extensions.conf" 607L, 20983C written
/dev/pts/4
14:02:27
#asterisk -rcv
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
    -- Executing [2001@group1:1] Dial("SIP/2204-095c07d0", "SIP/br/2001") in new stack
    -- Called br/2001
    -- SIP/br-095c8920 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/2204-095c07d0' status is 'CONGESTION'
linux12*CLI>
linux12*CLI>
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
14:03:13
#vi /etc/asterisk/extensions.conf
--- /tmp/l3-saved-7470.11739.18461	2010-10-26 15:03:22.000000000 +0300
+++ /etc/asterisk/extensions.conf	2010-10-26 15:03:41.000000000 +0300
@@ -581,7 +581,7 @@
 ;exten =>_XXXX,1,Dial(SIP/bb/${EXTEN})
 
 [local]
-;exten =>_22XX,1,Dial(SIP/${EXTEN})
+exten =>_22XX,1,Dial(SIP/${EXTEN})
 exten => 2298,1,Record(/tmp/warning1:gsm,,3)
 exten => 2297,1,Playback(/tmp/warning1)
 
/dev/pts/2
14:05:52
#[group2]
exten =>2201,1,Dial(SIP/2201,1)
exten =>2201,n,Dial(SIP/2204)
;exten =>2202,1,Dial(SIP/2202)
exten =>2203,1,Dial(SIP/2203)
exten =>2204,1,Dial(SIP/2204)
;exten =>2201,1,Dial(SIP/2201)
;exten => 2298,1,Record(/tmp/warning1:gsm,,3)
;in Belorussia
;exten =>_19XX,1,Dial(SIP/gm/${EXTEN})
;exten =>_20XX,1,Dial(SIP/br/${EXTEN})
...
[        ]
exten =>_20XX,1,Dial(SIP/br/${EXTEN})
exten =>_21XX,1,Dial(SIP/bb/${EXTEN})
[international]
[             ]
[group1]
[group1]
include => national
include => international
include => localDial(SIP/bb/${EXTEN})                                                                                                      601,0-1       98%
прошло 13 минут
14:19:33
#apt-get install tcpdump
Reading package lists... Done
Building dependency tree
Reading state information... Done
The following NEW packages will be installed:
  tcpdump
0 upgraded, 1 newly installed, 0 to remove and 0 not upgraded.
Need to get 306kB of archives.
After this operation, 631kB of additional disk space will be used.
Get:1 http://ftp.ua.debian.org lenny/main tcpdump 3.9.8-4 [306kB]
Fetched 306kB in 2s (115kB/s)
Selecting previously deselected package tcpdump.
(Reading database ... 44806 files and directories currently installed.)
Unpacking tcpdump (from .../tcpdump_3.9.8-4_i386.deb) ...
Processing triggers for man-db ...
Setting up tcpdump (3.9.8-4) ...
14:19:48
#vi /etc/asterisk/extensions.conf
14:22:09
#[ ]
         file name: /etc/asterisk/sip.conf
bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)
          modified: YES
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
         user name: root   host name: linux12.unix.nt
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
[       process]ID: 5639
[              ]
[2201]opening file "/etc/asterisk/sip.conf"
[2201]er=>mn:1234@192.168.111.1/bb
...
-- INSERT --                                                                                                                               54,1          56%
[gm]
type=friend-                                                                                                                               55,5          62%
secret=1234
-- INSERT --                                                                                                                               56,12         68%
host=dynamic
-- INSERT --                                                                                                                               57,12         75%
username=mn
-- INSERT --                                                                                                                               58,13         81%
"/etc/asterisk/sip.conf" 61L, 1149C written
14:23:09
#asterisk -rx "sip reload"

прошло 12 минут
14:36:05
#asterisk -rx "dialplan reload"
Dialplan reloaded.
14:42:06
#asterisk -rx "sip reload"

прошло 11 минут
14:53:24
#asterisk -rcv
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
    -- Called 2203
    -- SIP/2203-095cb530 is ringing
  == Spawn extension (default, 2203, 1) exited non-zero on 'SIP/mn-095c07d0'
    -- Executing [2204@default:1] Dial("SIP/mn-095c07d0", "SIP/2204") in new stack
    -- Called 2204
    -- SIP/2204-095cb530 is ringing
  == Spawn extension (default, 2204, 1) exited non-zero on 'SIP/mn-095c07d0'
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
14:59:28
#ifconfig
eth0      Link encap:Ethernet  HWaddr 00:1b:fc:7d:bb:37
          inet addr:192.168.112.1  Bcast:192.168.112.255  Mask:255.255.255.0
          inet6 addr: fe80::21b:fcff:fe7d:bb37/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:53537 errors:0 dropped:0 overruns:0 frame:0
          TX packets:44252 errors:0 dropped:0 overruns:0 carrier:1
          collisions:0 txqueuelen:1000
          RX bytes:10377859 (9.8 MiB)  TX bytes:9210247 (8.7 MiB)
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:3527 errors:0 dropped:0 overruns:0 frame:0
          TX packets:3527 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:179798 (175.5 KiB)  TX bytes:179798 (175.5 KiB)
14:59:33
#fg
register=>mn:1234@192.168.111.1/bb
register=>mn:1234@192.168.110.1/br
register=>mn:1234@192.168.109.1/gm
[authentication]
[2201]
type=friend
secret=1234
host=dynamic
callerid="Andrey_2201"
context=group1                 ; group1 context for incoming calls
...
host=dynamic
callerid="Andrey_2201"
context=group1
[2005]
type=friend
secret=1234
host=dynamic
callerid="Andrey_2201"
context=group1
[bb]
15:00:19
#type=friend
Asterisk 1.4.21.2~dfsg-3+lenny1, Copyright (C) 1999 - 2008 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.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
...
2005                       (Unspecified)    D          0        Unmonitored
2204/2204                  192.168.112.11   D          5060     Unmonitored
2203/2203                  192.168.112.2    D          5060     Unmonitored
2202/2202                  192.168.112.2    D          1695     Unmonitored
2201/2201                  (Unspecified)    D          0        Unmonitored
8 sip peers [Monitored: 0 online, 0 offline Unmonitored: 6 online, 2 offline]
[Oct 26 16:03:03] NOTICE[2616]: chan_sip.c:15642 handle_request_register: Registration from 'sip:2205@192.168.112.1' failed for '192.168.112.11' - No matching peer found
linux12*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
прошло 10 минут
15:11:01
#vi /etc/asterisk/sip.conf
15:13:19
#vi /etc/asterisk/sip.conf
15:17:39
#ifconfig
eth0      Link encap:Ethernet  HWaddr 00:1b:fc:7d:bb:37
          inet addr:192.168.112.1  Bcast:192.168.112.255  Mask:255.255.255.0
          inet6 addr: fe80::21b:fcff:fe7d:bb37/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:56677 errors:0 dropped:0 overruns:0 frame:0
          TX packets:46966 errors:0 dropped:0 overruns:0 carrier:1
          collisions:0 txqueuelen:1000
          RX bytes:10957145 (10.4 MiB)  TX bytes:9871922 (9.4 MiB)
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:3529 errors:0 dropped:0 overruns:0 frame:0
          TX packets:3529 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:180510 (176.2 KiB)  TX bytes:180510 (176.2 KiB)
15:17:41
#vi /etc/asterisk/sip.conf
15:19:01
#vi /etc/asterisk/extensions.conf
--- /tmp/l3-saved-7213.14836.28204	2010-10-26 16:19:18.000000000 +0300
+++ /etc/asterisk/extensions.conf	2010-10-26 16:20:20.000000000 +0300
@@ -564,8 +564,10 @@
 exten => _68XX,n,Dial(SIP/22${EXTEN:2})
 
 
-exten =>2201,1,Dial(SIP/2201,1)
-exten =>2201,n,Dial(SIP/2204)
+exten =>2201,1,Dial(SIP/2201,5)
+exten =>2204,n,Dial(SIP/2204,5)
+exten =>2205,n,Dial(SIP/2205)
+
 
 ;exten =>2202,1,Dial(SIP/2202)
 exten =>2203,1,Dial(SIP/2203)
15:21:24
#vi /etc/asterisk/sip.conf
15:22:59
#vi /etc/asterisk/extensions.conf
--- /tmp/l3-saved-7213.12068.19896	2010-10-26 16:23:01.000000000 +0300
+++ /etc/asterisk/extensions.conf	2010-10-26 16:23:30.000000000 +0300
@@ -564,8 +564,8 @@
 exten => _68XX,n,Dial(SIP/22${EXTEN:2})
 
 
-exten =>2201,1,Dial(SIP/2201,5)
-exten =>2204,n,Dial(SIP/2204,5)
+exten =>2201,1,Dial(SIP/2201,1)
+exten =>2204,n,Dial(SIP/2204,1)
 exten =>2205,n,Dial(SIP/2205)
 
 
15:25:04
#vi /etc/asterisk/extensions.conf
15:26:09
#[ ]
Reading package lists... Done
Building dependency tree
Reading state information... Done
Package bind is not available, but is referred to by another package.
This may mean that the package is missing, has been obsoleted, or
is only available from another source
However the following packages replace it:
  manpages dnsutils bind9
E: Package bind has no installation candidate

Файлы

  • /etc/asterisk/extensions.conf
  • /etc/asterisk/sip.conf
  • /etc/asterisk/extensions.conf
    >
    ; extensions.conf - the Asterisk dial plan
    ;
    ; Static extension configuration file, used by
    ; the pbx_config module. This is where you configure all your
    ; inbound and outbound calls in Asterisk.
    ;
    ; This configuration file is reloaded
    ; - With the "dialplan reload" command in the CLI
    ; - With the "reload" command (that reloads everything) in the CLI
    ;
    ; The "General" category is for certain variables.
    ;
    [general]
    ;
    ; If static is set to no, or omitted, then the pbx_config will rewrite
    ; this file when extensions are modified.  Remember that all comments
    ; made in the file will be lost when that happens.
    ;
    ; XXX Not yet implemented XXX
    ;
    static=yes
    ;
    ; if static=yes and writeprotect=no, you can save dialplan by
    ; CLI command "dialplan save" too
    ;
    writeprotect=no
    ;
    ; If autofallthrough is set, then if an extension runs out of
    ; things to do, it will terminate the call with BUSY, CONGESTION
    ; or HANGUP depending on Asterisk's best guess. This is the default.
    ;
    ; If autofallthrough is not set, then if an extension runs out of
    ; things to do, Asterisk will wait for a new extension to be dialed
    ; (this is the original behavior of Asterisk 1.0 and earlier).
    ;
    ;autofallthrough=no
    ;
    ; If clearglobalvars is set, global variables will be cleared
    ; and reparsed on an extensions 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"
    ; 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=Zap/1
    ;CONSOLE=Phone/phone0
    IAXINFO=guest                                   ; IAXtel username/password
    ;IAXINFO=myuser:mypass
    TRUNK=Zap/G2                                    ; Trunk interface
    ;
    ; Note the 'G2' in the TRUNK variable above. It specifies which group (defined
    ; in zapata.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 Zap channel
    ;    (aka. ascending sequential hunt group).
    ; G: select the highest-numbered non-busy Zap 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
    ;
    ; 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
    ; parenthesis after their name which can be used in goto situations
    ;
    ; Contexts contain several lines, one for each step of each
    ; extension, which can take one of two forms as listed below,
    ; with the first form being preferred.
    ;
    ;[context]
    ;exten => someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,...)
    ;exten => someexten,{priority|label{+|-}offset}[(alias)],application,arg1|arg2...
    ;
    ; Included Contexts
    ;
    ; 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.
    ; The reason a context would include other contexts is for their
    ; extensions.
    ; The algorithm to find an extension is recursive, and works in this
    ; fashion:
    ;        first, given a stack on which to store context references,
    ;           push the context to find the extension onto the stack...
    ;    a) Try to find a matching extension in the context at the top of
    ;       the stack, and, if found, begin executing the priorities
    ;       there in sequence.
    ;    b) If not found, Search the switches, if any declared, in
    ;       sequence.
    ;    c) If still not found, for each include, push that context onto
    ;       the top of the context stack, and recurse to a).
    ;    d) If still not found, pop the entry from the top of the stack;
    ;       if the stack is empty, the search has failed. If it's not,
    ;       continue with the next context in c).
    ; This is a depth-first traversal, and stops with the first context
    ; that provides a matching extension. As usual, if more than one
    ; pattern in a context will match, the 'best' match will win.
    ; Please note that that extensions found in an included context are
    ; treated as if they were in the context from which the search began.
    ; The PBX's notion of the "current context" is not changed.
    ; Please note that in a context, it does not matter where an include
    ; directive occurs. Whether at the top, or near the bottom, the effect
    ; will be the same. The only thing that matters is that if there is
    ; more than one include directive, they will be searched for extensions
    ; in order, first to last.
    ; Also please note that pattern matches (like _9XX) are not treated
    ; any differently than exact matches (like 987). Also note that the
    ; order of extensions in a context have no affect on the outcome.
    ;
    ; Timing list for includes is
    ;
    ;   <time range>|<days of week>|<days of month>|<months>
    ;
    ; 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.
    ;
    ;
    ; Sample entries for extensions.conf
    ;
    ;
    [dundi-e164-canonical]
    ;
    ; List canonical entries here
    ;
    ;exten => 12564286000,1,Macro(stdexten,6000,IAX2/foo)
    ;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(Zap/G2/${EXTEN:7}) ; Expose all of 256-428
    ;exten => _1256325XXXX,1,Dial(Zap/G2/${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(${GLOBAL(TRUNK)}/${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}
    [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
    [macro-stdexten];
    ;
    ; Standard extension macro:
    ;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
    ;   ${ARG2} - Device(s) to ring
    ;
    exten => s,1,Dial(${ARG2},20)                   ; Ring the interface, 20 seconds maximum
    exten => s,2,Goto(s-${DIALSTATUS},1)            ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
    exten => s-NOANSWER,1,Voicemail(${ARG1},u)      ; If unavailable, send to voicemail w/ unavail announce
    exten => s-NOANSWER,2,Goto(default,s,1)         ; If they press #, return to start
    exten => s-BUSY,1,Voicemail(${ARG1},b)          ; If busy, send to voicemail w/ busy announce
    exten => s-BUSY,2,Goto(default,s,1)             ; If they press #, return to start
    exten => _s-.,1,Goto(s-NOANSWER,1)              ; Treat anything else as no answer
    exten => a,1,VoicemailMain(${ARG1})             ; If they press *, send the user into VoicemailMain
    [macro-stdPrivacyexten];
    ;
    ; Standard extension macro:
    ;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
    ;   ${ARG2} - Device(s) to ring
    ;   ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 extension-priority)
    ;   ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 extension-priority)`
    ;
    exten => s,1,Dial(${ARG2},20|p)                 ; Ring the interface, 20 seconds maximum, call screening
                                                    ; option (or use P for databased call screening)
    exten => s,2,Goto(s-${DIALSTATUS},1)            ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
    exten => s-NOANSWER,1,Voicemail(${ARG1},u)      ; If unavailable, send to voicemail w/ unavail announce
    exten => s-NOANSWER,2,Goto(default,s,1)         ; If they press #, return to start
    exten => s-BUSY,1,Voicemail(${ARG1},b)          ; If busy, send to voicemail w/ busy announce
    exten => s-BUSY,2,Goto(default,s,1)             ; If they press #, return to start
    exten => s-DONTCALL,1,Goto(${ARG3},s,1)         ; Callee chose to send this call to a polite "Don't call again" script.
    exten => s-TORTURE,1,Goto(${ARG4},s,1)          ; Callee chose to send this call to a telemarketer torture script.
    exten => _s-.,1,Goto(s-NOANSWER,1)              ; Treat anything else as no answer
    exten => a,1,VoicemailMain(${ARG1})             ; If they press *, send the user into VoicemailMain
    [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}|js)                    ; j is for Jump and s is for ANY call
    exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail)
    exten => s,n(autoanswer),Set(_ALERT_INFO="RA")                  ; This is for the PolyComs
    exten => s,n,SIPAddHeader(Call-Info: Answer-After=0)    ; This is for the Grandstream, Snoms, and Others
    exten => s,n,NoOp()                                     ; Add others here and Post on the Wiki!!!!
    exten => s,n,Dial(${ARG1}||)
    exten => s,n(fail),Hangup
    [demo]
    ;
    ; 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,Macro(stdexten,1234,${GLOBAL(CONSOLE)})
    exten => 1235,1,Voicemail(1234,u)               ; Right to voicemail
    exten => 1236,1,Dial(Console/dsp)               ; Ring forever
    exten => 1236,n,Voicemail(1234,b)               ; Unless busy
    ;
    ; # for when they're done with the demo
    ;
    exten => #,1,Playback(demo-thanks)      ; "Thanks for trying the demo"
    exten => #,n,Hangup                     ; Hang them up.
    ;
    ; A timeout and "invalid extension rule"
    ;
    exten => t,1,Goto(#,1)                  ; If they take too long, give up
    exten => i,1,Playback(invalid)          ; "That's not valid, try again"
    ;
    ; Create an extension, 500, for dialing the
    ; Asterisk demo.
    ;
    exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
    exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default)        ; Call the Asterisk demo
    exten => 500,n,Playback(demo-nogo)      ; Couldn't connect to the demo site
    exten => 500,n,Goto(s,6)                ; Return to the start over message.
    ;
    ; Create an extension, 600, for evaluating echo latency.
    ;
    exten => 600,1,Playback(demo-echotest)  ; Let them know what's going on
    exten => 600,n,Echo                     ; Do the echo test
    exten => 600,n,Playback(demo-echodone)  ; Let them know it's over
    exten => 600,n,Goto(s,6)                ; Start over
    ;
    ;       You can use the Macro Page to intercom a individual user
    exten => 76245,1,Macro(page,SIP/Grandstream1)
    ; or if your peernames are the same as extensions
    exten => _7XXX,1,Macro(page,SIP/${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/${EXTEN})
    ;[mainmenu]
    ;
    ; Example "main menu" context with submenu
    ;
    ;exten => s,1,Answer
    ;exten => s,n,Background(thanks)                ; "Thanks for calling press 1 for sales, 2 for support, ..."
    ;exten => s,n,WaitExten
    ;exten => 1,1,Goto(submenu,s,1)
    ;exten => 2,1,Hangup
    ;include => default
    ;
    ;[submenu]
    ;exten => s,1,Ringing                                   ; Make them comfortable with 2 seconds of ringback
    ;exten => s,n,Wait,2
    ;exten => s,n,Background(submenuopts)   ; "Thanks for calling the sales department.  Press 1 for steve, 2 for..."
    ;exten => s,n,WaitExten
    ;exten => 1,1,Goto(default,steve,1)
    ;exten => 2,1,Goto(default,mark,2)
    [default]
    exten =>2299,1,Answer
    exten =>2299,n,Wait(1)
    exten =>2299,n,Playback(demo-thanks)
    exten =>2299,n,Hangup
    exten => _22XX,1,Dial(SIP/${EXTEN})
    exten => _22XXX,1,Dial(SIP/1${EXTEN:1:4})
    exten => _68XX,1,Wait(2)
    exten => _68XX,n,Playback(/tmp/warning1)
    exten => _68XX,n,Dial(SIP/22${EXTEN:2})
    exten =>2201,1,Dial(SIP/2201,1)
    exten =>2201,n,Dial(SIP/2204)
    ;exten =>2202,1,Dial(SIP/2202)
    exten =>2203,1,Dial(SIP/2203)
    exten =>2204,1,Dial(SIP/2204)
    ;exten =>2201,1,Dial(SIP/2201)
    exten =>2298,1,Record(/tmp/warning1:gsm,,3)
    ;in Belorussia
    ;exten =>_19XX,1,Dial(SIP/gm/${EXTEN})
    ;exten =>_20XX,1,Dial(SIP/br/${EXTEN})
    ;exten =>_21XX,1,Dial(SIP/bb/${EXTEN})
    ;in other locations
    ;exten =>_21XX,1,Dial(SIP/ptr/${EXTEN})
    ;exten =>_XXXX,1,Dial(SIP/bb/${EXTEN})
    [local]
    exten =>_22XX,1,Dial(SIP/mn/${EXTEN})
    [national]
    exten =>_19XX,1,Dial(SIP/gm/${EXTEN})
    exten =>_20XX,1,Dial(SIP/br/${EXTEN})
    exten =>_21XX,1,Dial(SIP/bb/${EXTEN})
    [international]
    exten =>_XXXX,1,Dial(SIP/bb/${EXTEN})
    [group1]
    include => local
    include => national
    include => international
    [group2]
    include => local
    include => national
    [group3]
    include =>local
    
    /etc/asterisk/sip.conf
    >
    [general]
    context=default                 ; Default context for incoming calls
    allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
    bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)
    bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
    srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
    register=>mn:1234@192.168.111.1/bb
    register=>mn:1234@192.168.110.1/br
    register=>mn:1234@192.168.109.1/gm
    [authentication]
    [2201]
    type=friend
    secret=1234
    host=dynamic
    callerid="Minsk 2201"
    context=group1                 ; group1 context for incoming calls
    [2202]
    type=friend
    secret=1234
    host=dynamic
    callerid="Minsk Softphone 2202"
    context=group3               ; group1 context for incoming calls
    [2203]
    type=friend
    secret=1234
    host=dynamic
    callerid="Minsk Softphone 2203"
    context=group2
    [2204]
    type=friend
    secret=1234
    host=dynamic
    callerid="Minsk 2204"
    context=group1
    [bb]
    type=friend
    secret=1234
    host=dynamic
    username=mn
    [br]
    type=friend
    secret=1234
    host=dynamic
    username=mn
    [gm]
    type=friend
    secret=1234
    host=dynamic
    username=mn
    

    Статистика

    Время первой команды журнала09:36:54 2010-10-26
    Время последней команды журнала15:26:09 2010-10-26
    Количество командных строк в журнале101
    Процент команд с ненулевым кодом завершения, % 4.95
    Процент синтаксически неверно набранных команд, % 0.99
    Суммарное время работы с терминалом *, час 5.18
    Количество командных строк в единицу времени, команда/мин 0.32
    Частота использования команд
    vi33|===============================| 31.43%
    asterisk27|=========================| 25.71%
    less5|====| 4.76%
    ifconfig4|===| 3.81%
    cat4|===| 3.81%
    ping3|==| 2.86%
    nmap3|==| 2.86%
    local3|==| 2.86%
    ls3|==| 2.86%
    include3|==| 2.86%
    [2202]2|=| 1.90%
    [2|=| 1.90%
    {EXTEN})1|| 0.95%
    type=friend1|| 0.95%
    fg1|| 0.95%
    man1|| 0.95%
    [bb]1|| 0.95%
    [group2]1|| 0.95%
    play1|| 0.95%
    rm1|| 0.95%
    uptime1|| 0.95%
    namp1|| 0.95%
    apt-get1|| 0.95%
    2202)1|| 0.95%
    exten1|| 0.95%
    ____
    *) Интервалы неактивности длительностью 30 минут и более не учитываются

    Справка

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

    2. Для того чтобы убедиться, что журнал на текущем терминале ведётся, и команды записываются, дайте команду w. В поле WHAT, соответствующем текущему терминалу, должна быть указана программа script.

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

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

    5. Команды, ход выполнения которых был прерван пользователем, выделяются цветом.
      $ find / -name abc
      find: /home/devi-orig/.gnome2: Keine Berechtigung
      find: /home/devi-orig/.gnome2_private: Keine Berechtigung
      find: /home/devi-orig/.nautilus/metafiles: Keine Berechtigung
      find: /home/devi-orig/.metacity: Keine Berechtigung
      find: /home/devi-orig/.inkscape: Keine Berechtigung
      ^C
      

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

    7. Изменения, внесённые в текстовый файл с помощью редактора, запоминаются и показываются в журнале в формате ed. Строки, начинающиеся символом "<", удалены, а строки, начинающиеся символом ">" -- добавлены.
      $ vi ~/.bashrc
      2a3,5
      >    if [ -f /usr/local/etc/bash_completion ]; then
      >         . /usr/local/etc/bash_completion
      >        fi
      

    8. Для того чтобы изменить файл в соответствии с показанными в диффшоте изменениями, можно воспользоваться командой patch. Нужно скопировать изменения, запустить программу patch, указав в качестве её аргумента файл, к которому применяются изменения, и всавить скопированный текст:
      $ patch ~/.bashrc
      В данном случае изменения применяются к файлу ~/.bashrc

    9. Для того чтобы получить краткую справочную информацию о команде, нужно подвести к ней мышь. Во всплывающей подсказке появится краткое описание команды.

      Если справочная информация о команде есть, команда выделяется голубым фоном, например: vi. Если справочная информация отсутствует, команда выделяется розовым фоном, например: notepad.exe. Справочная информация может отсутствовать в том случае, если (1) команда введена неверно; (2) если распознавание команды LiLaLo выполнено неверно; (3) если информация о команде неизвестна LiLaLo. Последнее возможно для редких команд.

    10. Большие, в особенности многострочные, всплывающие подсказки лучше всего показываются браузерами KDE Konqueror, Apple Safari и Microsoft Internet Explorer. В браузерах Mozilla и Firefox они отображаются не полностью, а вместо перевода строки выводится специальный символ.

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

    12. Имя терминала, на котором была введена команда, показано в специальном блоке. Этот блок показывается только в том случае, если терминал текущей команды отличается от терминала предыдущей.

    13. Вывод не интересующих вас в настоящий момент элементов журнала, таких как время, имя терминала и других, можно отключить. Для этого нужно воспользоваться формой управления журналом вверху страницы.

    14. Небольшие комментарии к командам можно вставлять прямо из командной строки. Комментарий вводится прямо в командную строку, после символов #^ или #v. Символы ^ и v показывают направление выбора команды, к которой относится комментарий: ^ - к предыдущей, v - к следующей. Например, если в командной строке было введено:

      $ whoami
      
      user
      
      $ #^ Интересно, кто я?
      
      в журнале это будет выглядеть так:
      $ whoami
      
      user
      
      Интересно, кто я?

    15. Если комментарий содержит несколько строк, его можно вставить в журнал следующим образом:

      $ whoami
      
      user
      
      $ cat > /dev/null #^ Интересно, кто я?
      
      Программа whoami выводит имя пользователя, под которым 
      мы зарегистрировались в системе.
      -
      Она не может ответить на вопрос о нашем назначении 
      в этом мире.
      
      В журнале это будет выглядеть так:
      $ whoami
      user
      
      Интересно, кто я?
      Программа whoami выводит имя пользователя, под которым
      мы зарегистрировались в системе.

      Она не может ответить на вопрос о нашем назначении
      в этом мире.
      Для разделения нескольких абзацев между собой используйте символ "-", один в строке.

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

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

    О программе

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

    $Id$