Журнал лабораторных работ

# and how to activate them. For more information, see interfaces(5).
# The loopback network interface
auto lo
iface l address 192.168.10
        netmask 255.255.255.0
        network 192.168.101.0
        broadcast 192.168.15.255
# The p gateway 192.168.101.253
        # dns-* options are implemented by the resolvconf package, if installed
allow-h dns-nameservers 192.168.101.253
...
~
~
~
~
~
~
~
~
~
"/etc/network/interfaces" 18L, 522C записано

--- /tmp/l3-saved-8268.37.2992	2011-06-20 13:08:50.000000000 +0300
+++ /etc/network/interfaces	2011-06-20 13:40:27.000000000 +0300
@@ -8,11 +8,11 @@
 # The primary network interface
 allow-hotplug eth0
 iface eth0 inet static
-	address 192.168.10
+	address 192.168.109.1
 	netmask 255.255.255.0
-	network 192.168.101.0
-	broadcast 192.168.15.255
-	gateway 192.168.101.253
+	network 192.168.109.0
+	broadcast 192.168.109.255
+	gateway 192.168.109.253
 	# dns-* options are implemented by the resolvconf package, if installed
 	dns-nameservers 192.168.101.253
 	dns-search unix.nt

bash: etc/netwok/interfaces: Нет такого файла или каталога

traceroute to 192.168.101.1 (192.168.101.1), 30 hops max, 60 byte packets
 1  192.168.109.253 (192.168.109.253)  0.421 ms  0.406 ms  0.397 ms
 2  192.168.101.1 (192.168.101.1)  0.625 ms  0.624 ms  0.617 ms

# This file describes the network interfaces available on your system
# and how to activate them. For more information, see interfaces(5).
# The loopback network interface
auto lo
iface l address 192.168.109.1
        netmask 255.255.255.0
        network 192.168.109.0
        broadcast 192.168.109.255
# The p gateway 192.168.109.253
        # dns-* options are implemented by the resolvconf package, if installed
allow-h dns-nameservers 192.168.101.253
        dns-search unix.nt
iface eth0 inet static

eth0      Link encap:Ethernet  HWaddr 00:1d:60:ec:c2:4c
          inet addr:192.168.15.9  Bcast:192.168.15.255  Mask:255.255.255.0
          inet6 addr: fe80::21d:60ff:feec:c24c/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:38246 errors:0 dropped:0 overruns:0 frame:0
          TX packets:12953 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:35306064 (33.6 MiB)  TX bytes:937698 (915.7 KiB)
          Interrupt:17
eth1      Link encap:Ethernet  HWaddr 00:07:e9:18:d7:9b
...
          collisions:0 txqueuelen:1000
          RX bytes:0 (0.0 B)  TX bytes:0 (0.0 B)
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:22 errors:0 dropped:0 overruns:0 frame:0
          TX packets:22 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:0
          RX bytes:2071 (2.0 KiB)  TX bytes:2071 (2.0 KiB)

eth0      Link encap:Ethernet  HWaddr 00:1d:60:ec:c2:4c
          inet addr:192.168.15.9  Bcast:192.168.15.255  Mask:255.255.255.0
          inet6 addr: fe80::21d:60ff:feec:c24c/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:38247 errors:0 dropped:0 overruns:0 frame:0
          TX packets:12953 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:35306433 (33.6 MiB)  TX bytes:937698 (915.7 KiB)
          Interrupt:17

SIOCDELRT: No such process

bash: get: команда не найдена

eth0      Link encap:Ethernet  HWaddr 00:1d:60:ec:c2:4c
          inet addr:192.168.109.1  Bcast:192.168.109.255  Mask:255.255.255.0
          inet6 addr: fe80::21d:60ff:feec:c24c/64 Scope:Link
          UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
          RX packets:38273 errors:0 dropped:0 overruns:0 frame:0
          TX packets:12985 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:35312090 (33.6 MiB)  TX bytes:941763 (919.6 KiB)
          Interrupt:17

bash: bash/get: Нет такого файла или каталога

Чтение списков пакетов... Готово
Построение дерева зависимостей
Чтение информации о состоянии... Готово
Будут установлены следующие дополнительные пакеты:
  isc-dhcp-server
Предлагаемые пакеты:
  isc-dhcp-server-ldap
НОВЫЕ пакеты, которые будут установлены:
  dhcp3-server isc-dhcp-server
обновлено 0, установлено 2 новых пакетов, для удаления отмечено 0 пакетов, и 0 пакетов не обновлено.
...
Распаковывается пакет isc-dhcp-server (из файла .../isc-dhcp-server_4.1.1-P1-15+squeeze2_amd64.deb)...
Выбор ранее не выбранного пакета dhcp3-server.
Распаковывается пакет dhcp3-server (из файла .../dhcp3-server_4.1.1-P1-15+squeeze2_all.deb)...
Обрабатываются триггеры для man-db ...
Настраивается пакет isc-dhcp-server (4.1.1-P1-15+squeeze2) ...
Generating /etc/default/isc-dhcp-server...
Starting ISC DHCP server: dhcpdcheck syslog for diagnostics. ... failed!
 failed!
invoke-rc.d: initscript isc-dhcp-server, action "start" failed.
Настраивается пакет dhcp3-server (4.1.1-P1-15+squeeze2) ...

# Defaults for dhcp initscript
# sourced by /etc/init.d/dhcp
# installed at /etc/default/isc-dhcp-server by the maintainer scripts
#
# This is a POSIX shell fragment
#
# On what interfaces should the DHCP server (dhcpd) serve DHCP requests?
#       Separate multiple interfaces with spaces, e.g. "eth0 eth1".
INTERFACES=""

--- /tmp/l3-saved-9276.29881.19785	2011-06-20 14:33:45.000000000 +0300
+++ /etc/dhcp/dhcpd.conf	2011-06-20 14:36:17.000000000 +0300
@@ -32,8 +32,8 @@
 
 # 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;
+#subnet 192.168.109.0 netmask 255.255.255.0 {
+#  range 192.168.109.200 192.168.109.250;
 #  option routers rtr-239-0-1.example.org, rtr-239-0-2.example.org;
 #}
 

Stopping ISC DHCP server: dhcpd failed!
Starting ISC DHCP server: dhcpdcheck syslog for diagnostics. ... failed!
 failed!

--- /tmp/l3-saved-9710.20544.10182	2011-06-20 14:43:08.000000000 +0300
+++ /etc/dhcp/dhcpd.conf	2011-06-20 14:43:53.000000000 +0300
@@ -32,10 +32,10 @@
 
 # This is a very basic subnet declaration.
 
-#subnet 192.168.109.0 netmask 255.255.255.0 { 
-#  range 192.168.109.200 192.168.109.250;
+subnet 192.168.109.0 netmask 255.255.255.0 { 
+  range 192.168.109.200 192.168.109.250;
 #  option routers rtr-239-0-1.example.org, rtr-239-0-2.example.org;
-#}
+}
 
 # This declaration allows BOOTP clients to get dynamic addresses,
 # which we don't really recommend.

ihu - Qt VoIP softphone with an own, encrypted protocol
libexosip2-4 - eXtended OSIP library
liblivemedia-dev - multimedia RTSP streaming library
livemedia-utils - multimedia RTSP streaming tools
pidgin-sipe - Pidgin protocol plugin to connect to MS Office Communicator
sflphone-daemon - SIP and IAX2 compatible VoIP phone - core daemon
sflphone-data - SIP and IAX2 compatible VoIP phone - common data
sflphone-gnome - SIP and IAX2 compatible VoIP phone - GNOME client
sip-tester - a performance testing tool for the SIP protocol
yate-qt4 - YATE-based universal telephony client

asterisk-mobile - bluetooth mobile devices support for Asterisk
asterisk-mp3 - MP3 format support (format_mp3) for the Asterisk PBX
asterisk-mysql - MySQL support for the Asterisk PBX (cdr mainly)
asterisk-ooh323c - H.323 protocol support for Asterisk (ooh323c stack)
asterisk-chan-capi - Common ISDN API 2.0 implementation for Asterisk
asterisk-core-sounds-en-g722 - asterisk PBX sound files - English/g722
asterisk-core-sounds-en-gsm - asterisk PBX sound files - English/gsm
asterisk-core-sounds-en-wav - asterisk PBX sound files - English/wav
asterisk-core-sounds-es-g722 - asterisk PBX sound files - Spanish/g722
asterisk-core-sounds-es-gsm - asterisk PBX sound files - Spanish/gsm
...
op-panel - switchboard type application for the Asterisk PBX
libopenr2-3 - MFC/R2 (telephony) call setup library
python-asterisk - управление Asterisk из сценариев Python
asterisk-config - Configuration files for Asterisk
asterisk-dbg - Debugging symbols for Asterisk
asterisk-dev - Development files for Asterisk
asterisk-doc - Source code documentation for Asterisk
asterisk-h323 - H.323 protocol support for Asterisk
asterisk-sounds-main - Core Sound files for Asterisk (English)
asterisk - Open Source Private Branch Exchange (PBX)

Чтение списков пакетов... Готово
Построение дерева зависимостей
Чтение информации о состоянии... Готово
Будут установлены следующие дополнительные пакеты:
  asterisk autopoint dahdi dahdi-linux debhelper freetds-common fxload gettext git html2text intltool-debian
  libc-client2007e libcorosync4 libcurl3 liberror-perl libgmime-2.0-2a libiksemel3 libmail-sendmail-perl libopenais3
  libopenh323-1.18.0 libopenr2-3 libpq5 libpri1.4 libpt-1.10.10 libpt-1.10.10-plugins-alsa libpt-1.10.10-plugins-v4l
  libradiusclient-ng2 libresample1 libsox-fmt-alsa libsox-fmt-base libsox1b libspandsp2 libsqlite0 libss7-1 libssh2-1
  libsybdb5 libsys-hostname-long-perl libtonezone2.0 libunistring0 libvpb0 mlock module-assistant odbcinst odbcinst1debian2
  po-debconf rsync sox unixodbc vpb-driver-source
...
Настраивается пакет asterisk (1:1.6.2.9-2+squeeze2) ...
Adding system user for Asterisk
Добавляется пользователь «asterisk» в группу «dialout» ...
Добавление пользователя asterisk в группу dialout
Готово.
Добавляется пользователь «asterisk» в группу «audio» ...
Добавление пользователя asterisk в группу audio
Готово.
Starting Asterisk PBX: asterisk.
Настраивается пакет asterisk-h323 (1:1.6.2.9-2+squeeze2) ...

adsi.conf                chan_dahdi.conf         festival.conf     misdn.conf              say.conf
adtranvofr.conf          cli_aliases.conf        followme.conf     modules.conf            sip.conf
agents.conf              cli.conf                func_odbc.conf    musiconhold.conf        sip_notify.conf
ais.conf                 cli_permissions.conf    gtalk.conf        muted.conf              skinny.conf
alarmreceiver.conf       codecs.conf             h323.conf         osp.conf                sla.conf
alsa.conf                console.conf            http.conf         oss.conf                smdi.conf
amd.conf                 dbsep.conf              iax.conf          phone.conf              telcordia-1.adsi
asterisk.adsi            dnsmgr.conf             iaxprov.conf      phoneprov.conf          udptl.conf
asterisk.conf            dsp.conf                indications.conf  queuerules.conf         unistim.conf
cdr_adaptive_odbc.conf   dundi.conf              jabber.conf       queues.conf             usbradio.conf
cdr.conf                 enum.conf               jingle.conf       res_config_sqlite.conf  users.conf
cdr_custom.conf          extconfig.conf          logger.conf       res_ldap.conf           voicemail.conf
cdr_manager.conf         extensions.ael          manager.conf      res_odbc.conf           vpb.conf
cdr_odbc.conf            extensions.conf         manager.d         res_pgsql.conf
cdr_pgsql.conf           extensions.lua          meetme.conf       res_snmp.conf
cdr_sqlite3_custom.conf  extensions_minivm.conf  mgcp.conf         rpt.conf
cdr_tds.conf             features.conf           minivm.conf       rtp.conf

/etc/asterisk/sip.conf

/etc/asterisk/sip.conf

-rw-r----- 1 asterisk asterisk 63882 Апр 23 18:04 /etc/asterisk/sip.conf

/etc/asterisk/sip.conf

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

Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux9 (pid = 11296)
...
[Jun 20 16:34:44] NOTICE[11333]: chan_sip.c:20281 handle_request_invite: Call from '' to extension '0972681570' rejected because extension not found in context 'default'.
linux9*CLI>
  == Using SIP RTP CoS mark 5
[Jun 20 16:44:16] NOTICE[11333]: chan_sip.c:20281 handle_request_invite: Call from '3101' to extension '3102' rejected because extension not found in context 'default'.
    -- Remote UNIX connection
Executing last minute cleanups
  == Destroying musiconhold processes
linux9*CLI>
Disconnected from Asterisk server
Executing last minute cleanups

Чтение списков пакетов... Готово
Построение дерева зависимостей
Чтение информации о состоянии... Готово
Уже установлена самая новая версия gnome.
обновлено 0, установлено 0 новых пакетов, для удаления отмечено 0 пакетов, и 0 пакетов не обновлено.

--- /tmp/l3-saved-8268.20230.12546	2011-06-20 15:54:40.000000000 +0300
+++ /etc/network/interfaces	2011-06-20 15:59:45.000000000 +0300
@@ -8,11 +8,11 @@
 # The primary network interface
 allow-hotplug eth0
 iface eth0 inet static
-	address 192.168.15.9
+	address 192.168.101.1
 	netmask 255.255.255.0
-	network 192.168.15.0
+	network 192.168.101.0
 	broadcast 192.168.15.255
-	gateway 192.168.15.253
+	gateway 192.168.101.253
 	# dns-* options are implemented by the resolvconf package, if installed
-	dns-nameservers 192.168.15.253
+	dns-nameservers 192.168.101.253
 	dns-search unix.nt

# and how to activate them. For more information, see interfaces(5).
# The loopback network interface
auto lo
iface l address 192.168.101.1
        netmask 255.255.255.0
        network 192.168.101.0
        broadcast 192.168.15.255
# The p gateway 192.168.101.253
        # dns-* options are implemented by the resolvconf package, if installed
allow-h dns-nameservers 192.168.101.253
...
~
~
~
~
~
~
~
~
~
"/etc/network/interfaces" 18L, 525C записано

--- /tmp/l3-saved-11423.25944.30559	2011-06-20 16:16:55.000000000 +0300
+++ /etc/asterisk/sip.conf	2011-06-20 16:21:54.000000000 +0300
@@ -12,57 +12,32 @@
 host=dynamic
 user=3101 
 
+[general]
+context=default
+allowoverlap=no
+udpbindaddr=0.0.0.0
+tcpenable=no
+tcpbindaddr=0.0.0.0
+srvlookup=yes
+
+[3102]
+type=friend
+secre=password
+host=dynamic
+user=3102 
+
+
+
+
+
+
+
+
+
+
 
-				; unauthenticated sessions that will be allowed
-                                ; to connect at any given time. (default: 100)
 
-srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
-                                ; Note: Asterisk only uses the first host 
-                                ; in SRV records
-                                ; Disabling DNS SRV lookups disables the 
-                                ; ability to place SIP calls based on domain 
-                                ; names to some other SIP users on the Internet
-                                ; Specifying a port in a SIP peer definition or
-                                ; when dialing outbound calls will supress SRV
-                                ; lookups for that peer or call.
-
-;pedantic=yes                   ; Enable checking of tags in headers, 
-                                ; international character conversions in URIs
-                                ; and multiline formatted headers for strict
-                                ; SIP compatibility (defaults to "no")
-
-; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
-;tos_sip=cs3                    ; Sets TOS for SIP packets.
-;tos_audio=ef                   ; Sets TOS for RTP audio packets.
-;tos_video=af41                 ; Sets TOS for RTP video packets.
-;tos_text=af41                  ; Sets TOS for RTP text packets.
-
-;cos_sip=3                      ; Sets 802.1p priority for SIP packets.
-;cos_audio=5                    ; Sets 802.1p priority for RTP audio packets.
-;cos_video=4                    ; Sets 802.1p priority for RTP video packets.
-;cos_text=3                     ; Sets 802.1p priority for RTP text packets.
-
-;maxexpiry=3600                 ; Maximum allowed time of incoming registrations
-                                ; and subscriptions (seconds)
-;minexpiry=60                   ; Minimum length of registrations/subscriptions (default 60)
-;defaultexpiry=120              ; Default length of incoming/outgoing registration
-;mwiexpiry=3600                 ; Expiry time for outgoing MWI subscriptions
-;qualifyfreq=60                 ; Qualification: How often to check for the 
-                                ; host to be up in seconds
-                                ; Set to low value if you use low timeout for
-                                ; NAT of UDP sessions
-;qualifygap=100			; Number of milliseconds between each group of peers being qualified
-;qualifypeers=1			; Number of peers in a group to be qualified at the same time
-;notifymimetype=text/plain      ; Allow overriding of mime type in MWI NOTIFY
-;buggymwi=no                    ; Cisco SIP firmware doesn't support the MWI RFC
-                                ; fully. Enable this option to not get error messages
-                                ; when sending MWI to phones with this bug.
-;vmexten=voicemail              ; dialplan extension to reach mailbox sets the 
-                                ; Message-Account in the MWI notify message 
-                                ; defaults to "asterisk"
-;disallow=all                   ; First disallow all codecs
-;allow=ulaw                     ; Allow codecs in order of preference
-;allow=ilbc                     ; see doc/rtp-packetization for framing options
+              ; 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

--- /tmp/l3-saved-11423.13282.24921	2011-06-20 16:22:19.000000000 +0300
+++ /etc/asterisk/sip.conf	2011-06-20 16:24:02.000000000 +0300
@@ -37,47 +37,6 @@
 
 
 
-              ; see doc/rtp-packetization for framing options
-;
-; This option specifies a preference for which music on hold class this channel
-; should listen to when put on hold if the music class has not been set on the
-; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
-; channel putting this one on hold did not suggest a music class.
-;
-; This option may be specified globally, or on a per-user or per-peer basis.
-;
-;mohinterpret=default
-;
-; This option specifies which music on hold class to suggest to the peer channel
-; when this channel places the peer on hold. It may be specified globally or on
-; a per-user or per-peer basis.
-;
-;mohsuggest=default
-;
-;parkinglot=plaza               ; Sets the default parking lot for call parking
-                                ; This may also be set for individual users/peers
-                                ; Parkinglots are configured in features.conf
-;language=en                    ; Default language setting for all users/peers
-                                ; This may also be set for individual users/peers
-;relaxdtmf=yes                  ; Relax dtmf handling
-;trustrpid = no                 ; If Remote-Party-ID should be trusted
-;sendrpid = yes                 ; If Remote-Party-ID should be sent
-;prematuremedia=no		; Some ISDN links send empty media frames before 
-				; the call is in ringing or progress state. The SIP 
-				; channel will then send 183 indicating early media
-				; which will be empty - thus users get no ring signal.
-				; Setting this to "no" will stop any media before we have
-				; call progress. Default is "yes".
-
-;progressinband=never           ; If we should generate in-band ringing always
-                                ; use 'never' to never use in-band signalling, even in cases
-                                ; where some buggy devices might not render it
-                                ; Valid values: yes, no, never Default: never
-;useragent=Asterisk PBX         ; Allows you to change the user agent string
-                                ; The default user agent string also contains the Asterisk
-                                ; version. If you don't want to expose this, change the
-                                ; useragent string.
-;sdpsession=Asterisk PBX        ; Allows you to change the SDP session name string, (s=)
                                 ; Like the useragent parameter, the default user agent string
                                 ; also contains the Asterisk version.
 ;sdpowner=root                  ; Allows you to change the username field in the SDP owner string, (o=)

--- /tmp/l3-saved-11423.12721.7147	2011-06-20 16:36:17.000000000 +0300
+++ /etc/asterisk/sip.conf	2011-06-20 16:37:09.000000000 +0300
@@ -12,6 +12,9 @@
 host=dynamic
 user=3101 
 
+
+
+
 [general]
 context=default
 allowoverlap=no
@@ -37,18 +40,7 @@
 
 
 
-                                ; Like the useragent parameter, the default user agent string
-                                ; also contains the Asterisk version.
-;sdpowner=root                  ; Allows you to change the username field in the SDP owner string, (o=)
-                                ; This field MUST NOT contain spaces
-;promiscredir = no              ; If yes, allows 302 or REDIR to non-local SIP address
-                                ; Note that promiscredir when redirects are made to the
-                                ; local system will cause loops since Asterisk is incapable
-                                ; of performing a "hairpin" call.
-;usereqphone = no               ; If yes, ";user=phone" is added to uri that contains
-                                ; a valid phone number
-;dtmfmode = rfc2833             ; Set default dtmfmode for sending DTMF. Default: rfc2833
-                                ; Other options: 
+ ; Other options: 
                                 ; info : SIP INFO messages (application/dtmf-relay)
                                 ; shortinfo : SIP INFO messages (application/dtmf)
                                 ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)

; 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
"/etc/asterisk/extensions.conf" 846L, 31344C
;
...
;
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
; variable on top of an existing variable, and its value will revert to itshe                              239,1         29%

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

Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux9 (pid = 11956)
...
  == Spawn extension (default, 3102, 1) exited non-zero on 'SIP/3101-0000000e'
[Jun 20 17:19:36] NOTICE[11993]: chan_sip.c:21599 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 3102
[Jun 20 17:21:37] NOTICE[11993]: chan_sip.c:21599 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 3102
[Jun 20 17:24:37] NOTICE[11993]: chan_sip.c:21599 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 3102
    -- Remote UNIX connection
Executing last minute cleanups
  == Destroying musiconhold processes
linux9*CLI>
Disconnected from Asterisk server
Executing last minute cleanups

No such command 'restart' (type 'core show help restart' for other possible commands)

--- /tmp/l3-saved-11423.13696.28624	2011-06-20 16:56:17.000000000 +0300
+++ /etc/asterisk/extensions.conf	2011-06-20 16:56:42.000000000 +0300
@@ -2,8 +2,8 @@
 exten => 8000,1,Wait(1)
 exten => 8000,n,Playback(demo-congrats)
 
-exten => 8xxx,1,Wait(1)
-exten => 8xxx,n,Playback(demo-congrats)
+exten => _8xxx,1,Wait(1)
+exten => _8xxx,n,Playback(demo-congrats)
 
 
 

;
[page]
exten => _X.,1,Macro(page,SIP/${EXTEN})
;[mainmenu]
;
; Example "main menu" context with submenu
;
;exten => s,1,Answer
;exten => s,n,Background(thanks)                ; "Thanks for calling press 1 for sales, 2 for support, ..."
;exten => s,n,WaitExten
...
exten => 8000,1,Wait(1)
exten => 8000,n,Playback(demo-congrats)
exten => _8xxx,1,Wait(1)
exten => _8xxx,n,Say($exten)
exten => _31xx,1,Dial(SIP/${EXTEN})
"/etc/asterisk/extensions.conf" 801L, 27724C записано
;
; User context is where entries from users.conf are registered.  The
; default value is 'default'
;

--- /tmp/l3-saved-11423.18981.29881	2011-06-20 17:34:51.000000000 +0300
+++ /etc/asterisk/extensions.conf	2011-06-20 17:38:21.000000000 +0300
@@ -6,7 +6,7 @@
 exten => _8xxx,n,Say($exten)
 
 exten => _31xx,1,Dial(SIP/${EXTEN},10)
-exten => _3103,1,Dial(SIP/3101$SIP/3102,10)
+exten => _3103,1,Dial(SIP/3101&SIP/3102,10)
 exten => _31xx,n,Playback(demo-congrats)
 
 

--- /tmp/l3-saved-9710.30147.23685	2011-06-20 17:36:47.000000000 +0300
+++ /etc/asterisk/sip.conf	2011-06-20 17:37:28.000000000 +0300
@@ -29,22 +29,6 @@
 host=dynamic
 user=3102 
 
-#
-
-
-[general]
-context=default
-allowoverlap=no
-udpbindaddr=0.0.0.0
-tcpenable=no
-tcpbindaddr=0.0.0.0
-srvlookup=yes
-
-[3103]
-type=friend
-secre=password
-host=dynamic
-user=3103
 
 
 

Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux9 (pid = 1683)
...
  == Parsing '/etc/asterisk/sip.conf':   == Found
[Jun 21 09:17:21] WARNING[1806]: config.c:1041 process_text_line: Unknown directive '#' at line 15 of /etc/asterisk/sip.conf
  == Parsing '/etc/asterisk/users.conf':   == Found
  == Parsing '/etc/asterisk/sip_notify.conf':   == Found
    -- Remote UNIX connection
Executing last minute cleanups
  == Destroying musiconhold processes
linux9*CLI>
Disconnected from Asterisk server
Executing last minute cleanups

#ssh user@192.168.15.253

root@192.168.108.1's password:
Linux linux8 2.6.32-5-amd64 #1 SMP Wed May 18 23:13:22 UTC 2011 x86_64
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: Mon Jun 20 15:17:46 2011 from 192.168.108.253
l3-agent is already running: pid=8475; pidfile=/root/.lilalo/l3-agent.pid

user@192.168.15.253's password:
root@192.168.108.1's password:
Linux linux8 2.6.32-5-amd64 #1 SMP Wed May 18 23:13:22 UTC 2011 x86_64
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: Mon Jun 20 15:17:46 2011 from 192.168.108.253
l3-agent is already running: pid=8475; pidfile=/root/.lilalo/l3-agent.pid

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

                    /etc/init.d/asterisk restart
Stopping Asterisk PBX: asterisk.
Starting Asterisk PBX: asterisk.

Asterisk 1.6.2.9-2+squeeze2, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.2.9-2+squeeze2 currently running on linux9 (pid = 3398)
...
[Jun 21 09:36:25] NOTICE[3435]: chan_sip.c:21768 handle_request_register: Registration from '<sip:lion@192.168.109.1>' failed for '192.168.103.1' - Wrong password
[Jun 21 09:36:31] NOTICE[3435]: chan_sip.c:21768 handle_request_register: Registration from '<sip:lion@192.168.109.1>' failed for '192.168.103.1' - Wrong password
  == Using SIP RTP CoS mark 5
[Jun 21 09:36:52] NOTICE[3435]: chan_sip.c:20281 handle_request_invite: Call from '3101' to extension '1301' rejected because extension not found in context 'default'.
    -- Remote UNIX connection
Executing last minute cleanups
  == Destroying musiconhold processes
linux9*CLI>
Disconnected from Asterisk server
Executing last minute cleanups

[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
"/etc/asterisk/sip.conf" 831L, 42478C
srvlookup=yes
[3101]
type=friendord
...
tcpbindaddr=0.0.0.0
            lion:password@192.168.103.1/paris
register => paris:password@192.168.103.1/lion
type=friendord
host=dynamic
callerid="VoIP Phone <3101>"
user=3101
#
[general]
"/etc/asterisk/sip.conf" 831L, 42478C записано