/l3/users/sergs/asterisk/linux5.unix.nt/root :1 :2 :3 :4 :5 :6
[ править ]

Журнал лабораторных работ

Содержание

Журнал

Вторник (05/25/10)

/dev/tty2
11:07:51
#w
 12:08:36 up 55 min,  2 users,  load average: 0.00, 0.00, 0.00
USER     TTY      FROM              LOGIN@   IDLE   JCPU   PCPU WHAT
root     tty1     -                11:26   28:21   1.48s  1.34s ssh 192.168.15.
root     tty2     -                12:07    0.00s  0.06s  0.00s /bin/login --
11:08:36
#free
             total       used       free     shared    buffers     cached
Mem:       3099904     273700    2826204          0      25612     201752
-/+ buffers/cache:      46336    3053568
Swap:      2650684          0    2650684
test comment
11:17:37
#vi /etc/network/interfaces
--- /tmp/l3-saved-4631.1030.5468	2010-05-25 12:21:45.000000000 +0300
+++ /etc/network/interfaces	2010-05-25 12:23:22.000000000 +0300
@@ -8,11 +8,11 @@
 # The primary network interface
 allow-hotplug eth0
 iface eth0 inet static
-	address 192.168.15.5
+	address 192.168.5.1
 	netmask 255.255.255.0
-	network 192.168.15.0
-	broadcast 192.168.15.255
-	gateway 192.168.15.254
+	network 192.168.5.0
+	broadcast 192.168.5.255
+	gateway 192.168.5.254
 	# dns-* options are implemented by the resolvconf package, if installed
 	dns-nameservers 10.0.35.1
 	dns-search unix.nt
прошло 14 минут
11:32:09
#ls -al
total 32
drwxr-xr-x  7 root root 4096 2010-05-25 12:23 .
drwxr-xr-x 60 root root 4096 2010-05-25 12:04 ..
drwxr-xr-x  2 root root 4096 2008-05-05 23:33 if-down.d
drwxr-xr-x  2 root root 4096 2008-05-05 23:33 if-post-down.d
drwxr-xr-x  2 root root 4096 2008-05-05 23:33 if-pre-up.d
drwxr-xr-x  2 root root 4096 2010-05-25 11:48 if-up.d
-rw-r--r--  1 root root  512 2010-05-25 12:23 interfaces
drwxr-xr-x  2 root root 4096 2010-05-25 11:13 run
11:32:11
#vi /etc/network/interfaces
11:33:31
#./if- 
if-down.d/      if-post-down.d/ if-pre-up.d/    if-up.d/
11:33:31
#ls
if-down.d  if-post-down.d  if-pre-up.d  if-up.d  interfaces  run
11:33:58
#./interfaces 
bash: ./interfaces: Permission denied
11:34:03
#ifocnfig 
bash: ifocnfig: command not found
11:34:28
#ifconfig
eth0      Link encap:Ethernet  HWaddr 00:1b:fc:7d:bc:f0
          inet addr:192.168.15.5  Bcast:192.168.15.255  Mask:255.255.255.0
          inet6 addr: fe80::21b:fcff:fe7d:bcf0/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:25163 errors:0 dropped:0 overruns:0 frame:0
          TX packets:15540 errors:0 dropped:0 overruns:0 carrier:1
          collisions:0 txqueuelen:1000
          RX bytes:31456733 (29.9 MiB)  TX bytes:1056711 (1.0 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:14 errors:0 dropped:0 overruns:0 frame:0
          TX packets:14 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:1072 (1.0 KiB)  TX bytes:1072 (1.0 KiB)
11:34:33
#ifconfig down eth0 
^C
11:34:54
#ifconfig down eth0 
^C
11:34:57
#/etc/init.d/networking restart
Reconfiguring network interfaces...SIOCDELRT: No such process
done.
11:35:15
#/etc/init.d/networking restart
Reconfiguring network interfaces...done.
11:35:18
#ifconfig
eth0      Link encap:Ethernet  HWaddr 00:1b:fc:7d:bc:f0
          inet addr:192.168.15.5  Bcast:192.168.15.255  Mask:255.255.255.0
          inet6 addr: fe80::21b:fcff:fe7d:bcf0/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:25163 errors:0 dropped:0 overruns:0 frame:0
          TX packets:15560 errors:0 dropped:0 overruns:0 carrier:1
          collisions:0 txqueuelen:1000
          RX bytes:31456733 (29.9 MiB)  TX bytes:1057911 (1.0 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:23 errors:0 dropped:0 overruns:0 frame:0
          TX packets:23 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:1826 (1.7 KiB)  TX bytes:1826 (1.7 KiB)
11:35:21
#if 
if        ifconfig  ifdown    ifup
11:35:21
#ifdown eth0
ifdown: interface eth0 not configured
11:35:45
#ifup eth0










11:35:53




#ifconfig


eth0      Link encap:Ethernet  HWaddr 00:1b:fc:7d:bc:f0
          inet addr:192.168.5.1  Bcast:192.168.5.255  Mask:255.255.255.0
          inet6 addr: fe80::21b:fcff:fe7d:bcf0/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:25164 errors:0 dropped:0 overruns:0 frame:0
          TX packets:15575 errors:0 dropped:0 overruns:0 carrier:1
          collisions:0 txqueuelen:1000
          RX bytes:31456793 (29.9 MiB)  TX bytes:1058832 (1.0 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:28 errors:0 dropped:0 overruns:0 frame:0
          TX packets:28 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:2250 (2.1 KiB)  TX bytes:2250 (2.1 KiB)











11:35:55




#w


 12:36:07 up  1:22,  2 users,  load average: 0.00, 0.00, 0.00
USER     TTY      FROM              LOGIN@   IDLE   JCPU   PCPU WHAT
root     tty1     -                11:26    2.00s  2.08s  1.94s ssh 192.168.15.
root     tty2     -                12:07    0.00s  0.20s  0.00s /bin/login --











11:36:07




#ps -ax


Warning: bad ps syntax, perhaps a bogus '-'? See http://procps.sf.net/faq.html
  PID TTY      STAT   TIME COMMAND
    1 ?        Ss     0:01 init [2]
    2 ?        S<     0:00 [kthreadd]
    3 ?        S<     0:00 [migration/0]
    4 ?        S<     0:00 [ksoftirqd/0]
    5 ?        S<     0:00 [watchdog/0]
    6 ?        S<     0:00 [migration/1]
    7 ?        S<     0:00 [ksoftirqd/1]
    8 ?        S<     0:00 [watchdog/1]
...
 4581 ?        Ss     0:09 l3-agent
 4586 tty2     Ss+    0:00 /bin/login --
 4593 tty2     S+     0:00 script -f -c bash -q /root/.lilalo//16534126012696426
 4630 tty2     S+     0:00 script -f -c bash -q /root/.lilalo//16534126012696426
 4631 pts/0    Ss     0:00 bash
 4860 pts/0    S      0:00 /bin/sh /etc/network/if-up.d/ntpdate
 4871 pts/0    S      0:00 lockfile-touch /var/lock/ntpdate
 4872 pts/0    S      0:00 /usr/sbin/ntpdate -s -b 0.debian.pool.ntp.org 1.debia
 4876 ?        Ss     0:00 /usr/sbin/sshd
 4900 pts/0    R+     0:00 ps -ax











11:36:35




#kill -9 2674












11:36:59




#ping ya.ru


PING ya.ru (93.158.134.3) 56(84) bytes of data.
64 bytes from www.yandex.ru (93.158.134.3): icmp_seq=1 ttl=48 time=49.5 ms
64 bytes from www.yandex.ru (93.158.134.3): icmp_seq=2 ttl=48 time=52.7 ms
^C
--- ya.ru ping statistics ---
2 packets transmitted, 2 received, 0% packet loss, time 1003ms
rtt min/avg/max/mdev = 49.512/51.142/52.772/1.630 ms











11:38:04




#apt-get install gpm


Reading package lists... Done
Building dependency tree
Reading state information... Done
The following NEW packages will be installed:
  gpm
0 upgraded, 1 newly installed, 0 to remove and 0 not upgraded.
Need to get 213kB of archives.
After this operation, 455kB of additional disk space will be used.
Get:1 http://10.0.35.1 lenny/main gpm 1.20.4-3.1 [213kB]
Fetched 213kB in 0s (4761kB/s)
Preconfiguring packages ...
Selecting previously deselected package gpm.
(Reading database ... 21823 files and directories currently installed.)
Unpacking gpm (from .../gpm_1.20.4-3.1_i386.deb) ...
Processing triggers for man-db ...
Setting up gpm (1.20.4-3.1) ...
Creating config file /etc/gpm.conf with new version
Stopping mouse interface server: gpmO0o.oops(): [daemon/check_kill.c(38)]: Could not open /var/run/gpm.pid.
 failed!
Starting mouse interface server: gpm.











11:39:56




#ping 192.168.4.1


PING 192.168.4.1 (192.168.4.1) 56(84) bytes of data.
64 bytes from 192.168.4.1: icmp_seq=1 ttl=63 time=0.226 ms
64 bytes from 192.168.4.1: icmp_seq=2 ttl=63 time=0.224 ms
64 bytes from 192.168.4.1: icmp_seq=3 ttl=63 time=0.217 ms
^C
--- 192.168.4.1 ping statistics ---
3 packets transmitted, 3 received, 0% packet loss, time 1999ms
rtt min/avg/max/mdev = 0.217/0.222/0.226/0.012 ms











11:40:25




#apt-cache search dhcp server | less










/dev/tty1
11:41:58




#ssh 192.168.4.1


The authenticity of host '192.168.4.1 (192.168.4.1)' can't be established.
RSA key fingerprint is a8:91:5d:22:21:72:02:03:a6:3e:1e:6b:70:33:b2:c8.
Are you sure you want to continue connecting (yes/no)? yes
Warning: Permanently added '192.168.4.1' (RSA) to the list of known hosts.
root@192.168.4.1's password:
Linux linux4.unix.nt 2.6.26-2-686 #1 SMP Wed May 12 21:56:10 UTC 2010 i686
The programs included with the Debian GNU/Linux system are free software;
the exact distribution terms for each program are described in the
individual files in /usr/share/doc/*/copyright.
Debian GNU/Linux comes with ABSOLUTELY NO WARRANTY, to the extent
permitted by applicable law.
Last login: Tue May 25 12:40:50 2010 from 192.168.15.100
l3-agent is already running: pid=4496; pidfile=/root/.lilalo/l3-agent.pid











11:42:07




#screen -x










/dev/tty2
11:43:26




#apt-get install dhcp3-server


Package configuration
 ┌──────────────────────────────┤ DHCP Server ├──────────────────────────────┐
 │                                                                           │
 │ Non-authoritative version of DHCP server                                  │
 │                                                                           │
 │ The version 3 DHCP server is non-authoritative by default.                │
 │                                                                           │
 │ This means that if a client requests an address that the server knows     │
 │ nothing about and the address is incorrect for that network segment, the  │
 │ server will _not_ send a DHCPNAK (which tells the client it should stop   │
...
 └───────────────────────────────────────────────────────────────────────────┘
Selecting previously deselected package dhcp3-server.
(Reading database ... 21858 files and directories currently installed.)
Unpacking dhcp3-server (from .../dhcp3-server_3.1.1-6+lenny4_i386.deb) ...
Processing triggers for man-db ...
Setting up dhcp3-server (3.1.1-6+lenny4) ...
Generating /etc/default/dhcp3-server...
Starting DHCP server: dhcpd3check syslog for diagnostics. failed!
 failed!
invoke-rc.d: initscript dhcp3-server, action "start" failed.











11:44:22




#cd /etc/dhcp3/












11:45:00




#vi /etc/default/dhcp3-server


--- /tmp/l3-saved-4631.11626.16744	2010-05-25 12:45:58.000000000 +0300
+++ /etc/default/dhcp3-server	2010-05-25 12:46:26.000000000 +0300
@@ -8,4 +8,4 @@
 
 # On what interfaces should the DHCP server (dhcpd) serve DHCP requests?
 #	Separate multiple interfaces with spaces, e.g. "eth0 eth1".
-INTERFACES=""
+INTERFACES="eth0"









11:46:26




#vi dhcpd.conf


--- /tmp/l3-saved-4631.24149.3452	2010-05-25 12:47:46.000000000 +0300
+++ dhcpd.conf	2010-05-25 12:51:41.000000000 +0300
@@ -11,8 +11,8 @@
 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 "unix.nt";
+option domain-name-servers 10.0.35.1;
 
 default-lease-time 600;
 max-lease-time 7200;
@@ -33,10 +33,10 @@
 
 # This is a very basic subnet declaration.
 
-#subnet 10.254.239.0 netmask 255.255.255.224 {
-#  range 10.254.239.10 10.254.239.20;
-#  option routers rtr-239-0-1.example.org, rtr-239-0-2.example.org;
-#}
+subnet 192.168.5.0 netmask 255.255.255.0 {
+  range 192.168.5.10 192.168.5.50;
+  option routers 192.168.5.254;
+}
 
 # This declaration allows BOOTP clients to get dynamic addresses,
 # which we don't really recommend.









11:51:41




#/etc/init.d/dhcp3-server restart


Stopping DHCP server: dhcpd3 failed!
Starting DHCP server: dhcpd3.











11:51:55




#/etc/init.d/dhcp3-server restart


Stopping DHCP server: dhcpd3.
Starting DHCP server: dhcpd3.











11:52:01




#/etc/init.d/dhcp3-server status


Status of DHCP server: dhcpd3 is running.











11:52:23




#tail /var/lib/dhcp3/dhc





dhclient.leases  dhcpd.leases     dhcpd.leases~











11:52:23




#tail /var/lib/dhcp3/dhc





dhclient.leases  dhcpd.leases     dhcpd.leases~











11:52:23




#tail /var/lib/dhcp3/dhcpd.leases


lease 192.168.5.10 {
  starts 2 2010/05/25 09:53:31;
  ends 2 2010/05/25 10:03:31;
  cltt 2 2010/05/25 09:53:31;
  binding state active;
  next binding state free;
  hardware ethernet 00:1d:60:ec:c0:9e;
  uid "\001\000\035`\354\300\236";
  client-hostname "UC-10";
}











11:56:01




#apt-get install vlock


Reading package lists... Done
Building dependency tree
Reading state information... Done
The following NEW packages will be installed:
  vlock
0 upgraded, 1 newly installed, 0 to remove and 0 not upgraded.
Need to get 38.1kB of archives.
After this operation, 221kB of additional disk space will be used.
Get:1 http://10.0.35.1 lenny/main vlock 2.2.2-3 [38.1kB]
Fetched 38.1kB in 0s (945kB/s)
Selecting previously deselected package vlock.
(Reading database ... 21871 files and directories currently installed.)
Unpacking vlock (from .../vlock_2.2.2-3_i386.deb) ...
Processing triggers for man-db ...
Setting up vlock (2.2.2-3) ...











11:57:55




#vl





bash: vl: command not found











11:57:57




#vlock


This TTY is now locked.
Please press [ENTER] to unlock.
root's Password:











11:58:10




#man vlock










11:58:21




#vlock -a





vlock: this terminal is not a virtual console











11:58:25




#man vlock










11:58:46




#vlock --all





vlock: this terminal is not a virtual console











11:58:55




#vl





bash: vl: command not found











11:59:24




#vlock


This TTY is now locked.
Please press [ENTER] to unlock.
root's Password:









прошло 77 минут




13:16:54




#man vlock










13:18:02




#vlock -a





vlock: this terminal is not a virtual console











13:18:06




#vl





bash: vl: command not found











13:18:14




#tail /var/lib/dhcp3/dhcpd.leases


lease 192.168.5.10 {
  starts 2 2010/05/25 11:18:30;
  ends 2 2010/05/25 11:28:30;
  cltt 2 2010/05/25 11:18:30;
  binding state active;
  next binding state free;
  hardware ethernet 00:1d:60:ec:c0:9e;
  uid "\001\000\035`\354\300\236";
  client-hostname "UC-10";
}











13:20:48




#tail /var/lib/dhcp3/dhcpd.leases


lease 192.168.5.11 {
  starts 2 2010/05/25 11:21:07;
  ends 2 2010/05/25 11:31:07;
  cltt 2 2010/05/25 11:21:07;
  binding state active;
  next binding state free;
  hardware ethernet 00:0e:08:d2:2e:07;
  uid "\001\000\016\010\322.\007";
  client-hostname "ph200";
}











13:21:39




#tail /var/lib/dhcp3/dhcpd.leases -n 30


lease 192.168.5.10 {
  starts 2 2010/05/25 11:13:30;
  ends 2 2010/05/25 11:23:30;
  cltt 2 2010/05/25 11:13:30;
  binding state active;
  next binding state free;
  hardware ethernet 00:1d:60:ec:c0:9e;
  uid "\001\000\035`\354\300\236";
  client-hostname "UC-10";
}
...
lease 192.168.5.11 {
  starts 2 2010/05/25 11:21:07;
  ends 2 2010/05/25 11:31:07;
  cltt 2 2010/05/25 11:21:07;
  binding state active;
  next binding state free;
  hardware ethernet 00:0e:08:d2:2e:07;
  uid "\001\000\016\010\322.\007";
  client-hostname "ph200";
}











13:26:45




#apt-get install asterisk-sounds-





asterisk-sounds-extra  asterisk-sounds-main











13:26:45




#apt-get install asterisk-sounds-extra


Reading package lists... Done
Building dependency tree
Reading state information... Done
The following NEW packages will be installed:
  asterisk-sounds-extra
0 upgraded, 1 newly installed, 0 to remove and 0 not upgraded.
Need to get 3224kB of archives.
After this operation, 6291kB of additional disk space will be used.
Get:1 http://10.0.35.1 lenny/main asterisk-sounds-extra 1.4.7-1 [3224kB]
Fetched 3224kB in 0s (11.6MB/s)
Selecting previously deselected package asterisk-sounds-extra.
(Reading database ... 25234 files and directories currently installed.)
Unpacking asterisk-sounds-extra (from .../asterisk-sounds-extra_1.4.7-1_all.deb) ...
Setting up asterisk-sounds-extra (1.4.7-1) ...











13:26:54




#dpkg -l | grep aster


ii  asterisk                          1:1.4.21.2~dfsg-3+lenny1 Open Source Private Branch Exchange (PBX)
ii  asterisk-config                   1:1.4.21.2~dfsg-3+lenny1 Configuration files for Asterisk
ii  asterisk-sounds-extra             1.4.7-1                  Additional sound files for the Asterisk PBX
ii  asterisk-sounds-main              1:1.4.21.2~dfsg-3+lenny1 Core Sound files for Asterisk (English)
ii  base-passwd                       3.5.20                   Debian base system master password and group files











13:27:34




#cd /etc/asterisk/












13:27:51




#ls -al


total 436
drwxr-xr-x  3 asterisk asterisk  4096 2010-05-25 14:26 .
drwxr-xr-x 66 root     root      4096 2010-05-25 14:26 ..
-rw-r-----  1 asterisk asterisk   140 2009-12-14 21:08 adsi.conf
-rw-r-----  1 asterisk asterisk   840 2009-12-14 21:08 adtranvofr.conf
-rw-r-----  1 asterisk asterisk  2724 2009-12-14 21:08 agents.conf
-rw-r-----  1 asterisk asterisk  2227 2009-12-14 21:08 alarmreceiver.conf
-rw-r-----  1 asterisk asterisk  2675 2009-12-14 21:08 alsa.conf
-rw-r-----  1 asterisk asterisk   767 2009-12-14 21:08 amd.conf
-rw-r-----  1 asterisk asterisk  3260 2009-12-14 21:08 asterisk.adsi
...
-rw-r-----  1 asterisk asterisk  4044 2009-12-14 21:08 skinny.conf
-rw-r-----  1 asterisk asterisk  6691 2009-12-14 21:08 sla.conf
-rw-r-----  1 asterisk asterisk  2665 2009-12-14 21:08 smdi.conf
-rw-r-----  1 asterisk asterisk  1384 2009-12-14 21:08 telcordia-1.adsi
-rw-r-----  1 asterisk asterisk   598 2009-12-14 21:08 udptl.conf
-rw-r-----  1 asterisk asterisk  1804 2009-12-14 21:08 users.conf
-rw-r-----  1 asterisk asterisk 11723 2009-12-14 21:08 voicemail.conf
-rw-r-----  1 asterisk asterisk  2772 2009-12-14 21:08 vpb.conf
-rw-r-----  1 asterisk asterisk   393 2009-12-14 21:08 watchdog.conf
-rw-r-----  1 asterisk asterisk 24193 2009-12-14 21:08 zapata.conf











13:27:54




#ls


adsi.conf           enum.conf         manager.conf      rtp.conf
adtranvofr.conf     esel.conf         manager.d         say.conf
agents.conf         extconfig.conf    meetme.conf       sip.conf
alarmreceiver.conf  extensions.ael    mgcp.conf         sip_notify.conf
alsa.conf           extensions.conf   misdn.conf        skinny.conf
amd.conf            features.conf     modules.conf      sla.conf
asterisk.adsi       festival.conf     musiconhold.conf  smdi.conf
asterisk.conf       followme.conf     muted.conf        telcordia-1.adsi
cdr.conf            func_odbc.conf    osp.conf          udptl.conf
cdr_custom.conf     gtalk.conf        oss.conf          users.conf
cdr_manager.conf    h323.conf         phone.conf        voicemail.conf
cdr_odbc.conf       http.conf         privacy.conf      vpb.conf
cdr_pgsql.conf      iax.conf          queues.conf       watchdog.conf
cdr_tds.conf        iaxprov.conf      res_odbc.conf     zapata.conf
codecs.conf         indications.conf  res_pgsql.conf
dnsmgr.conf         jabber.conf       res_snmp.conf
dundi.conf          logger.conf       rpt.conf











13:27:58




#dp





dpkg                 dpkg-gencontrol      dpkg-scansources
dpkg-architecture    dpkg-gensymbols      dpkg-shlibdeps
dpkg-buildpackage    dpkg-name            dpkg-source
dpkg-checkbuilddeps  dpkg-parsechangelog  dpkg-split
dpkg-deb             dpkg-preconfigure    dpkg-statoverride
dpkg-distaddfile     dpkg-query           dpkg-trigger
dpkg-divert          dpkg-reconfigure     dprofpp
dpkg-genchanges      dpkg-scanpackages











13:27:58




#dpkg -L asterisk | less










13:28:59




#tail /var/lib/dhcp3/dhcpd.leases -n 30


lease 192.168.5.11 {
  starts 2 2010/05/25 11:23:37;
  ends 2 2010/05/25 11:33:37;
  cltt 2 2010/05/25 11:23:37;
  binding state active;
  next binding state free;
  hardware ethernet 00:0e:08:d2:2e:07;
  uid "\001\000\016\010\322.\007";
  client-hostname "SipuraSPA";
}
...
lease 192.168.5.10 {
  starts 2 2010/05/25 11:28:30;
  ends 2 2010/05/25 11:38:30;
  cltt 2 2010/05/25 11:28:30;
  binding state active;
  next binding state free;
  hardware ethernet 00:1d:60:ec:c0:9e;
  uid "\001\000\035`\354\300\236";
  client-hostname "UC-10";
}











13:29:15




#less /var/lib/dhcp3/dhcpd.leases










13:29:55




#less /etc/default/asterisk








прошло 34 минуты




14:04:07




#cd /etc/asterisk/










прошло 45 минут




14:49:20




#cp sip.conf sip.conf.old












14:50:13




#vi sip.conf


--- /tmp/l3-saved-4631.18656.27922	2010-05-25 15:58:47.000000000 +0300
+++ sip.conf	2010-05-25 16:00:12.000000000 +0300
@@ -1,669 +0,0 @@
-;
-; SIP Configuration example for Asterisk
-;
-; Syntax for specifying a SIP device in extensions.conf is
-; SIP/devicename where devicename is defined in a section below.
-;
-; You may also use 
-; SIP/username@domain to call any SIP user on the Internet
-; (Don't forget to enable DNS SRV records if you want to use this)
-; 
-; 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 
-; 
-; Useful CLI commands to check peers/users:
-;   sip show peers		Show all SIP peers (including friends)
-;   sip show users		Show all SIP users (including friends)
-;   sip show registry		Show status of hosts we register with
-;
-;   sip debug			Show all SIP messages
-;
-;   reload chan_sip.so		Reload configuration file
-;				Active SIP peers will not be reconfigured
-;
-
-[general]
-context=default			; Default context for incoming calls
-;allowguest=no			; Allow or reject guest calls (default is yes)
-allowoverlap=no			; Disable overlap dialing support. (Default is yes)
-;allowtransfer=no		; Disable all transfers (unless enabled in peers or users)
-				; Default is enabled
-;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
-bindport=5060			; UDP Port to bind to (SIP standard port is 5060)
-				; bindport is the local UDP port that Asterisk will listen on
-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
-				; 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
-				
-;domain=mydomain.tld		; Set default domain for this host
-				; If configured, Asterisk will only allow
-				; INVITE and REFER to non-local domains
-				; Use "sip show domains" to list local domains
-;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 doc/ip-tos.txt 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.
-
-;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
-;t1min=100			; Minimum roundtrip time for messages to monitored hosts
-				; Defaults to 100 ms
-;notifymimetype=text/plain	; Allow overriding of mime type in MWI NOTIFY
-;checkmwi=10			; Default time between mailbox checks for peers
-;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
-;
-;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
-;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
-;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
-				; 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 the 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.
-;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)
-;alwaysauthreject = yes		; When an incoming INVITE or REGISTER is to be rejected,
- 		    		; for any reason, always reject with '401 Unauthorized'
- 				; instead of letting the requester know whether there was
- 				; a matching user or peer for their request
-
-;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 :-(
-
-;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.
-
-;
-; 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
-;
-;--------------------------- 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 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 limit set
-; for a device. When the call limit is filled, we will indicate busy. Note that
-; you need at least 2 in order to be able to do attended transfers.
-;
-; 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 = yes		; Notify subscriptions on RINGING state (default: no)
-;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.
-;limitonpeers = yes		; Apply call limits on peers only. This will improve 
-				; status notification when you are using type=friend
-				; Inbound calls, that really apply to the user part
-				; of a friend will now be added to and compared with
-				; the peer limit instead of applying two call limits,
-				; one for the peer and one for the user.
-				; "sip show inuse" will only show active calls on 
-				; the peer side of a "type=friend" object if this
-				; setting is turned on.
-
-;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
-;
-; This setting is available in the [general] section as well as in device configurations.
-; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
-; both parties have T38 support enabled in their Asterisk configuration 
-; This has to be enabled in the general section for all devices to work. You can then
-; disable it on a per device basis. 
-;
-; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.
-;
-; t38pt_udptl = yes            ; Default false
-;
-;----------------------------------------- OUTBOUND SIP REGISTRATIONS  ------------------------
-; Asterisk can register as a SIP user agent to a SIP proxy (provider)
-; Format for the register statement is:
-;       register => user[:secret[:authuser]]@host[:port][/extension]
-;
-; 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).
-;
-; host is either a host name defined in DNS or the name of a section defined
-; below.
-;
-; 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 type=peer and type=user sections for SIP providers
-;           (instead of type=friend) if you have calls in both directions
-  
-;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
-
-;----------------------------------------- NAT SUPPORT ------------------------
-; The externip, externhost and localnet settings are used if you use Asterisk
-; behind a NAT device to communicate with services on the outside.
-
-;externip = 200.201.202.203	; Address that we're going to put in outbound SIP
-				; messages if we're behind a NAT
-
-				; The externip and localnet is used
-				; when registering and communicating with other proxies
-				; that we're registered with
-;externhost=foo.dyndns.net	; Alternatively you can specify an 
-				; external host, and Asterisk will 
-				; perform DNS queries periodically.  Not
-				; recommended for production 
-				; environments!  Use externip instead
-;externrefresh=10		; How often to refresh externhost if 
-				; used
-				; You may add multiple local networks.  A reasonable 
-				; set of defaults are:
-;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
-;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 nat= setting is used when Asterisk is on a public IP, communicating with
-; devices hidden behind a NAT device (broadband router).  If you have one-way
-; audio problems, you usually have problems with your NAT configuration or your
-; firewall's support of SIP+RTP ports.  You configure Asterisk choice of RTP
-; ports for incoming audio in rtp.conf
-;
-;nat=no				; Global NAT settings  (Affects all peers and users)
-                                ; yes = Always ignore info and assume NAT
-                                ; no = Use NAT mode only according to RFC3581 (;rport)
-                                ; never = Never attempt NAT mode or RFC3581 support
-				; route = Assume NAT, don't send rport 
-				; (work around more UNIDEN bugs)
-
-;----------------------------------- MEDIA HANDLING --------------------------------
-; By default, Asterisk tries to re-invite the audio 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 with in the case where Asterisk is outside and have
-; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
-;
-;canreinvite=yes		; 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).
-				; The default setting is YES. If you have all clients
-				; behind a NAT, or for some other reason wants Asterisk to
-				; stay in the audio path, you may want to turn this off.
-
-				; In Asterisk 1.4 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 canreinvite is enabled when
-				; the device is actually behind NAT.
-
-;canreinvite=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).
-
-;canreinvite=update		; Yet a third option... use UPDATE for media path redirection,
-				; instead of INVITE. This can be combined with 'nonat', as
-				; 'canreinvite=update,nonat'. It implies 'yes'.
-
-;----------------------------------------- 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'.
-;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.
-; 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.
-
-; 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
-
-;------------------------------------------------------------------------------
-; Users and peers have different settings available. Friends have all settings,
-; since a friend is both a peer and a user
-;
-; User config options:        Peer configuration:
-; --------------------        -------------------
-; context                     context
-; callingpres		      callingpres
-; permit                      permit
-; deny                        deny
-; secret                      secret
-; md5secret                   md5secret
-; dtmfmode                    dtmfmode
-; canreinvite                 canreinvite
-; nat                         nat
-; callgroup                   callgroup
-; pickupgroup                 pickupgroup
-; language                    language
-; allow                       allow
-; disallow                    disallow
-; insecure                    insecure
-; trustrpid                   trustrpid
-; progressinband              progressinband
-; promiscredir                promiscredir
-; useclientcode               useclientcode
-; accountcode                 accountcode
-; setvar                      setvar
-; callerid		      callerid
-; amaflags		      amaflags
-; call-limit		      call-limit
-; allowoverlap		      allowoverlap
-; allowsubscribe	      allowsubscribe
-; allowtransfer	      	      allowtransfer
-; subscribecontext	      subscribecontext
-; videosupport		      videosupport
-; maxcallbitrate	      maxcallbitrate
-; rfc2833compensate           mailbox
-; t38pt_usertpsource          username
-;                             template
-;                             fromdomain
-;                             regexten
-;                             fromuser
-;                             host
-;                             port
-;                             qualify
-;                             defaultip
-;                             rtptimeout
-;                             rtpholdtimeout
-;                             sendrpid
-;                             outboundproxy
-;                             rfc2833compensate
-;                             t38pt_usertpsource
-
-;[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
-;secret=guessit
-;username=yourusername			; Authentication user for outbound proxies
-;fromuser=yourusername			; Many SIP providers require this!
-;fromdomain=provider.sip.domain	
-;host=box.provider.com
-;usereqphone=yes			; This provider requires ";user=phone" on URI
-;call-limit=5				; permit only 5 simultaneous outgoing calls to this peer
-;outboundproxy=proxy.provider.domain	; send outbound signaling to this proxy, not directly to the peer
-					; Call-limits will not be enforced on real-time peers,
-					; since they are not stored in-memory
-;port=80				; The port number we want to connect to on the remote side
-					; Also used as "defaultport" in combination with "defaultip" settings
-
-;------------------------------------------------------------------------------
-; Definitions of locally connected SIP devices
-;
-; type = user	a device that authenticates to us by "from" field to place calls
-; type = peer	a device we place calls to or that calls us and we match by host
-; type = friend two configurations (peer+user) in one
-;
-; 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
-
-;[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
-;canreinvite=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
-				; 1 for the explicit peer, 1 for the explicit user,
-				; remember that a friend equals 1 peer and 1 user in
-				; memory
-				; This will affect your subscriptions as well.
-				; 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 doc/callingpres.txt 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
-;canreinvite=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
-
-
-;[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
-;username=polly			; Username to use in INVITE until peer registers
-				; 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
-;
-; 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
-
-;[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
-;canreinvite=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
-;username=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 this device
-
-;[pre14-asterisk]
-;type=friend
-;secret=digium
-;host=dynamic
-;rfc2833compensate=yes		; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
-				; You must have this turned on or DTMF reception will work improperly.
-;t38pt_usertpsource=yes         ; Use the source IP address of RTP as the destination IP address for UDPTL packets
-                                ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
-                                ; external IP address of the remote device. If port forwarding is done at the client side
-                                ; then UDPTL will flow to the remote device.









/dev/tty1
14:51:24




#exit


Connection to 192.168.4.1 closed.











14:51:36




#ssh 192.168.15.6


install_l3bashrc_for_this_users=${users:-"root user"}  # users who will use l3ag
lilalo_context="/users/${lilalo_user}/${lab}/${hostname}"
#
###############################################################################
"install" 185 lines, 5285 characters
lilalo_rc=.l3rc
lilalo_home=.lilalo
url_lilalo="http://xgu.ru/lilalo"
url_l3bashrc="${url_lilalo}"/l3bashrc
url_l3agent="${url_lilalo}"/l3-agent
...
#xterm*|rxvt*)
#    PROMPT_COMMAND='echo -ne "\033]0;${USER}@${HOSTNAME}: ${PWD}\007"'
#    ;;
#*)
#    ;;
#esac
# enable bash completion in interactive shells
fi
#if [ -f /etc/bash_completion ]; then
"/etc/bash.bashrc" 47 lines, 1450 characters written











/dev/tty2
15:00:12




#vi sip.conf


--- /tmp/l3-saved-4631.24952.3065	2010-05-25 16:00:14.000000000 +0300
+++ sip.conf	2010-05-25 16:05:19.000000000 +0300
@@ -0,0 +1,7 @@
+context=default
+
+[501]
+type=friend
+secret=password
+host=dynamic
+









15:05:19




#/etc/init.d/asterisk restart


Stopping Asterisk PBX: asterisk.
Starting Asterisk PBX: asterisk.











15:08:03




#c


[501]
type=friend
~
~
~
~
~
~
~
~
~
~
~
~
~
~
~
"sip.conf" 8L, 75C written











15:08:17




#/etc/init.d/asterisk restart


Stopping Asterisk PBX: asterisk.
Starting Asterisk PBX: asterisk.











15:14:11




#mv extensions.conf extensions.conf.SAVE












15:14:26




#vi extensions.conf













15:14:46




#~


exten => 8000,1,Answer
exten => 8000,n,Wait(1)
exten => 8000,n,Playback(demo-thanks)
~
~
~
~
~
~
~
...
~
~
~
~
~
~
~
~
~
"extensions.conf" [New] 6L, 97C written











15:19:36




#asterisk -rx 'dialplan reload'


Dialplan reloaded.











15:29:20




#vim sip.conf


--- /tmp/l3-saved-4631.1238.29186	2010-05-25 16:29:27.000000000 +0300
+++ sip.conf	2010-05-25 16:29:48.000000000 +0300
@@ -6,3 +6,8 @@
 secret=password
 host=dynamic
 
+[502]
+type=friend
+secret=password
+host=dynamic
+









15:36:18




#asterisk -vvvr


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, 501, 1) exited non-zero on 'SIP/502-081d07d0'
    -- Executing [501@default:1] Dial("SIP/502-081d07d0", "SIP/501") in new stack
    -- Called 501
    -- SIP/501-081d2108 is ringing
    -- SIP/501-081d2108 answered SIP/502-081d07d0
    -- Native bridging SIP/502-081d07d0 and SIP/501-081d2108
  == Spawn extension (default, 501, 1) exited non-zero on 'SIP/502-081d07d0'
[May 25 16:37:53] NOTICE[10684]: chan_sip.c:15500 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 502
linux5*CLI> quit
Executing last minute cleanups











15:38:03




#vim extensions.conf


--- /tmp/l3-saved-4631.19569.26894	2010-05-25 16:38:05.000000000 +0300
+++ extensions.conf	2010-05-25 16:38:49.000000000 +0300
@@ -4,4 +4,4 @@
 exten => 8000,n,Wait(1)
 exten => 8000,n,Playback(demo-thanks)
 
-exten => 501,1,Dial(SIP/501)
+exten => _5XX,1,Dial(SIP/${EXTEN})









15:38:49




#asterisk -vvvr


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 '8000' priority 3 to default
    -- Added extension '_5XX' priority 1 to default
  == Parsing '/etc/asterisk/users.conf': Found
    -- Executing [502@default:1] Dial("SIP/501-081d07d0", "SIP/502") in new stack
    -- Called 502
    -- SIP/502-081d2108 is ringing
  == Spawn extension (default, 502, 1) exited non-zero on 'SIP/501-081d07d0'
[May 25 16:40:54] NOTICE[10684]: chan_sip.c:15500 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 502
linux5*CLI> quit
Executing last minute cleanups











15:42:03




#asterisk -vvvr


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 501
    -- SIP/501-081d2108 is ringing
  == Spawn extension (default, 501, 1) exited non-zero on 'SIP/502-081d07d0'
    -- Executing [502@default:1] Dial("SIP/501-081d07d0", "SIP/502") in new stack
    -- Called 502
    -- SIP/502-b5900500 is ringing
  == Spawn extension (default, 502, 1) exited non-zero on 'SIP/501-081d07d0'
[May 25 16:43:54] NOTICE[10684]: chan_sip.c:15500 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 502
linux5*CLI> quit
Executing last minute cleanups











15:44:32




#ping 192.168.4.1


PING 192.168.4.1 (192.168.4.1) 56(84) bytes of data.
64 bytes from 192.168.4.1: icmp_seq=1 ttl=63 time=0.228 ms
64 bytes from 192.168.4.1: icmp_seq=2 ttl=63 time=0.227 ms
^C
--- 192.168.4.1 ping statistics ---
2 packets transmitted, 2 received, 0% packet loss, time 1005ms
rtt min/avg/max/mdev = 0.227/0.227/0.228/0.015 ms











15:44:39




#ping 192.168.3.1


PING 192.168.3.1 (192.168.3.1) 56(84) bytes of data.
64 bytes from 192.168.3.1: icmp_seq=1 ttl=63 time=0.221 ms
^C
--- 192.168.3.1 ping statistics ---
1 packets transmitted, 1 received, 0% packet loss, time 0ms
rtt min/avg/max/mdev = 0.221/0.221/0.221/0.000 ms











15:44:42




#ping 192.168.2.1


PING 192.168.2.1 (192.168.2.1) 56(84) bytes of data.
64 bytes from 192.168.2.1: icmp_seq=1 ttl=63 time=0.226 ms
64 bytes from 192.168.2.1: icmp_seq=2 ttl=63 time=0.225 ms
^C
--- 192.168.2.1 ping statistics ---
2 packets transmitted, 2 received, 0% packet loss, time 999ms
rtt min/avg/max/mdev = 0.225/0.225/0.226/0.015 ms











15:44:47




#asterisk -vvvr


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
=========================================================================
...
Connected to Asterisk 1.4.21.2~dfsg-3+lenny1 currently running on linux5 (pid = 10673)
Verbosity is at least 3
[May 25 16:46:54] NOTICE[10684]: chan_sip.c:15500 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 502
    -- Executing [502@default:1] Dial("SIP/501-081d07d0", "SIP/502") in new stack
    -- Called 502
    -- SIP/502-081d2108 is ringing
  == Spawn extension (default, 502, 1) exited non-zero on 'SIP/501-081d07d0'
[May 25 16:49:54] NOTICE[10684]: chan_sip.c:15500 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 502
linux5*CLI> quit
Executing last minute cleanups











15:52:49




#vim sip.conf










15:53:17




#date


Tue May 25 16:53:37 EEST 2010











15:53:37




#vim sip.conf


--- /tmp/l3-saved-4631.18032.17300	2010-05-25 16:55:20.000000000 +0300
+++ sip.conf	2010-05-25 17:01:04.000000000 +0300
@@ -1,5 +1,6 @@
 [general]
 context=default
+register => crimea:password@192.168.4.1/crimea
 
 [501]
 type=friend
@@ -13,3 +14,5 @@
 host=dynamic
 callerid="X-Lite soft phone"
 
+
+









16:01:04




#asterisk -vvvr


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
=========================================================================
...
Verbosity is at least 3
linux5*CLI> sip reload
 Reloading SIP
  == Parsing '/etc/asterisk/sip.conf': Found
  == Parsing '/etc/asterisk/users.conf': Found
  == Parsing '/etc/asterisk/sip_notify.conf': Found
[May 25 17:01:29] WARNING[10684]: chan_sip.c:12620 handle_response_register: Forbidden - wrong password on authentication for REGISTER for 'crimea' to '192.168.4.1'
[May 25 17:01:55] NOTICE[10684]: chan_sip.c:15500 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 502
linux5*CLI> quit
Executing last minute cleanups











16:02:23




#vim sip.conf










16:02:59




#callerid="X-Lite soft phone"


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
=========================================================================
...
Connected to Asterisk 1.4.21.2~dfsg-3+lenny1 currently running on linux5 (pid = 10673)
Verbosity is at least 3
linux5*CLI> sip reload
 Reloading SIP
  == Parsing '/etc/asterisk/sip.conf': Found
  == Parsing '/etc/asterisk/users.conf': Found
  == Parsing '/etc/asterisk/sip_notify.conf': Found
[May 25 17:03:03] WARNING[10684]: chan_sip.c:12620 handle_response_register: Forbidden - wrong password on authentication for REGISTER for 'crimea' to '192.168.4.1'
linux5*CLI> quit
Executing last minute cleanups











16:03:22




#asterisk -vvvr


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
=========================================================================
...
501/501                    192.168.5.12     D          5060     Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
[May 25 17:04:55] NOTICE[10684]: chan_sip.c:15500 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 502
    -- Executing [502@default:1] Dial("SIP/501-081d2108", "SIP/502") in new stack
    -- Called 502
    -- SIP/502-081d07d0 is ringing
  == Spawn extension (default, 502, 1) exited non-zero on 'SIP/501-081d2108'
[May 25 17:05:36] NOTICE[10684]: chan_sip.c:15642 handle_request_register: Registration from '<sip:dnepr@192.168.5.1>' failed for '192.168.4.1' - No matching peer found
linux5*CLI> quit
Executing last minute cleanups











16:05:47




#vim sip.conf


--- /tmp/l3-saved-4631.6644.1934	2010-05-25 17:05:49.000000000 +0300
+++ sip.conf	2010-05-25 17:07:44.000000000 +0300
@@ -14,5 +14,10 @@
 host=dynamic
 callerid="X-Lite soft phone"
 
+[kharkov]
+type=friend
+secret=password
+host=192.168.4.1
+
 
 









16:07:44




#asterisk -vvvr


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
=========================================================================
...
  == Parsing '/etc/asterisk/users.conf': Found
  == Parsing '/etc/asterisk/sip_notify.conf': Found
linux5*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
kharkov                    192.168.4.1                 5060     Unmonitored
502/502                    192.168.5.10     D          37094    Unmonitored
501/501                    192.168.5.12     D          5060     Unmonitored
3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline]
linux5*CLI> quit
Executing last minute cleanups











16:10:05




#vim sip.conf


--- /tmp/l3-saved-4631.15922.20870	2010-05-25 17:10:08.000000000 +0300
+++ sip.conf	2010-05-25 17:11:34.000000000 +0300
@@ -1,6 +1,10 @@
 [general]
 context=default
 register => crimea:password@192.168.4.1/crimea
+register => crimea:password@192.168.3.1/crimea
+register => crimea:password@192.168.2.1/crimea
+register => crimea:password@192.168.1.1/crimea
+
 
 [501]
 type=friend
@@ -19,5 +23,20 @@
 secret=password
 host=192.168.4.1
 
+[odessa]
+type=friend
+secret=password
+host=192.168.3.1
+
+[kiev]
+type=friend
+secret=password
+host=192.168.2.1
+
+[dnepr]
+type=friend
+secret=password
+host=192.168.1.1
+
 
 









16:11:34




#asterisk -vvvr


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
=========================================================================
...
Name/username              Host            Dyn Nat ACL Port     Status
dnepr                      192.168.1.1                 5060     Unmonitored
kiev                       192.168.2.1                 5060     Unmonitored
odessa                     192.168.3.1                 5060     Unmonitored
kharkov                    192.168.4.1                 5060     Unmonitored
502/502                    192.168.5.10     D          37094    Unmonitored
501/501                    192.168.5.12     D          5060     Unmonitored
6 sip peers [Monitored: 0 online, 0 offline Unmonitored: 6 online, 0 offline]
linux5*CLI> quit
Executing last minute cleanups











16:13:51




#asterisk -vvvr


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
=========================================================================
...
[May 25 17:13:55] NOTICE[10684]: chan_sip.c:15500 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 502
linux5*CLI> sip reload
 Reloading SIP
  == Parsing '/etc/asterisk/sip.conf': Found
  == Parsing '/etc/asterisk/users.conf': Found
  == Parsing '/etc/asterisk/sip_notify.conf': Found
[May 25 17:13:58] WARNING[10684]: chan_sip.c:12627 handle_response_register: Got 404 Not found on SIP register to service crimea@192.168.1.1, giving up
[May 25 17:13:58] WARNING[10684]: chan_sip.c:12627 handle_response_register: Got 404 Not found on SIP register to service crimea@192.168.2.1, giving up
linux5*CLI> quit
Executing last minute cleanups











16:14:09




#vim extensions.conf


--- /tmp/l3-saved-4631.32548.20143	2010-05-25 17:14:17.000000000 +0300
+++ extensions.conf	2010-05-25 17:15:00.000000000 +0300
@@ -5,3 +5,8 @@
 exten => 8000,n,Playback(demo-thanks)
 
 exten => _5XX,1,Dial(SIP/${EXTEN})
+exten => _4XX,1,Dial(SIP/kharkov/${EXTEN})
+exten => _3XX,1,Dial(SIP/odessa/${EXTEN})
+exten => _2XX,1,Dial(SIP/kiev/${EXTEN})
+exten => _1XX,1,Dial(SIP/dnepr/${EXTEN})
+







Статистика
Время первой команды журнала11:07:51 2010- 5-25
Время последней команды журнала16:14:09 2010- 5-25
Количество командных строк в журнале99
Процент команд с ненулевым кодом завершения, % 5.05
Процент синтаксически неверно набранных команд, % 4.04
Суммарное время работы с терминалом *, час 2.49
Количество командных строк в единицу времени, команда/мин 0.66
Частота использования команд
asterisk10|=========|  9.71%
vim8|=======|  7.77%
tail7|======|  6.80%
vi7|======|  6.80%
ifconfig5|====|  4.85%
apt-get5|====|  4.85%
vlock5|====|  4.85%
ping5|====|  4.85%
ls4|===|  3.88%
less4|===|  3.88%
man3|==|  2.91%
cd3|==|  2.91%
/etc/init.d/dhcp3-server3|==|  2.91%
vl3|==|  2.91%
dpkg2|=|  1.94%
/etc/init.d/asterisk2|=|  1.94%
/etc/init.d/networking2|=|  1.94%
w2|=|  1.94%
ssh2|=|  1.94%
screen1||  0.97%
ifocnfig1||  0.97%
cp1||  0.97%
free1||  0.97%
#^1||  0.97%
interfaces1||  0.97%
~1||  0.97%
mv1||  0.97%
dp1||  0.97%
if-1||  0.97%
ps1||  0.97%
c1||  0.97%
ifup1||  0.97%
callerid="X-Lite1||  0.97%
ifdown1||  0.97%
grep1||  0.97%
if1||  0.97%
kill1||  0.97%
exit1||  0.97%
date1||  0.97%
apt-cache1||  0.97%
____
*) 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. 
    
  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. 
    
  3. 
    Команды, при наборе которых были допущены синтаксические ошибки, 
    выводятся перечёркнутым текстом:

    

    


    $ l s-l
    

    bash: l: command not found

    
    		

    

    

    
    
    4. 
    
  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. 
    
  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. 
    
  6. 
    Команды, выполненные с привилегиями суперпользователя,
    выделяются слева красной чертой.

    

    


    # id
    

    uid=0(root) gid=0(root) Gruppen=0(root)

    
    		

    

    
    
    
    
    7. 
    
  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. 
    
  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. 
    
  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. 
    
  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. 
    
  11. 
    Время ввода команды, показанное в журнале, соответствует времени 
    начала ввода командной строки, которое равно тому моменту, 
    когда на терминале появилось приглашение интерпретатора
    
    12. 
    
  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. 
    
  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. 
    
  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. 
    
  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. 
    
  16. 
    Комментарии, не относящиеся непосредственно ни к какой из команд, 
    добавляются точно таким же способом, только вместо симолов #^ или #v 
    нужно использовать символы #=
    
    17. 

    
  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. 

    
    
    
  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. 
    
    

    
    
    
  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.
    
    20. 
    
    



О программе
    

    LiLaLo (L3) расшифровывается как Live Lab Log.

    Программа разработана для повышения эффективности обучения Unix/Linux-системам.

    (c) Игорь Чубин, 2004-2008

    

$Id$