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Пятница (11/22/13)

/dev/pts/3
13:47:32
#make menuconfig
                                                                  Press 'h' for help.
                                                                  Add-ons (See README-addons.txt)
                                                                  Applications
                                                                  Bridging Modules
                                                                  Call Detail Recording
                                                                  Channel Event Logging
                                                                  Channel Drivers
                                                                  Codec Translators
                                                                  Format Interpreters
                                                                  Dialplan Functions
...
                                                                  Compiler Flags
                                                                  Voicemail Build Options
                                                                  Utilities
                                                                  AGI Samples
                                                            --->  Module Embedding
                                                                  Core Sound Packages
                                                                  Music On Hold File Packages
                                                                  Extras Sound Packages
menuselect changes saved!
make[1]: Leaving directory `/usr/src/asterisk-11.6.0'
13:48:30
#make install
CC="cc" CXX="" LD="" AR="" RANLIB="" CFLAGS="" LDFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" makeopts
make[1]: Entering directory `/usr/src/asterisk-11.6.0/menuselect'
make[1]: `makeopts' is up to date.
make[1]: Leaving directory `/usr/src/asterisk-11.6.0/menuselect'
Generating embedded module rules ...
   [LD] abstract_jb.o acl.o adsi.o alaw.o aoc.o app.o ast_expr2.o ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o audiohook.o autochan.o autoservice.o bridging.o callerid.o ccss.o cdr.o cel.o channel.o channel_internal_api.o chanvars.o cli.o config.o config_options.o data.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o event.o features.o file.o fixedjitterbuf.o format.o forma
Installing modules from channels...
Installing modules from pbx...
Installing modules from apps...
Installing modules from codecs...
...
 +-----------------  or ---------------------+
 +                                           +
 + You can go ahead and install the asterisk +
 + program documentation now or later run:   +
 +                                           +
 +               make progdocs               +
 +                                           +
 + **Note** This requires that you have      +
 + doxygen installed on your local system    +
 +-------------------------------------------+
13:48:50
#make menuconfig
                                                                  Press 'h' for help.
                                                            --->  Add-ons (See README-addons.txt)
                                                                  Applications
                                                                  Bridging Modules
                                                                  Call Detail Recording
                                                                  Channel Event Logging
                                                                  Channel Drivers
                                                                  Codec Translators
                                                                  Format Interpreters
                                                                  Dialplan Functions
...
                                                                  Compiler Flags
                                                                  Voicemail Build Options
                                                                  Utilities
                                                                  AGI Samples
                                                                  Module Embedding
                                                                  Core Sound Packages
                                                                  Music On Hold File Packages
                                                                  Extras Sound Packages
menuselect changes saved!
make[1]: Leaving directory `/usr/src/asterisk-11.6.0'
13:50:22
#/etc/init.d/asterisk start
[ ok ] Starting Asterisk PBX: asterisk.
13:50:33
#joe /etc/asterisk/meetme.conf
                                                                                                                 7
  How many seconds before the scheduled end of the conference should
 schedule=yes
  the participants be warned?
 endalert=120
  Update realtime when members login/out of the conference
 logmembercount=no
[rooms]
  How much earlier than the start time should we allow participants to
  Usage is conf => confno[,pin][,adminpin]
...
conf => 101,1234,0987
File /etc/asterisk/meetme.conf not changed so no update needed
;
  Usage is conf => confno[,pin][,adminpin]
  Note that once a participant has called the conference, a change to the pin
  number done in this file will not take effect until there are no more users
  in the conference and it goes away.  When it is created again, it will have
  the new pin number.
 conf => 1234
         2345,9938
прошло 32 минуты
/dev/pts/1
14:23:18
#asterisk -rvvv
Asterisk 11.6.0, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 11.6.0 currently running on debian (pid = 21926)
  == Using SIP RTP CoS mark 5
    -- Executing [3110@bank:1] MeetMe("SIP/3101-00000001", "101") in new stack
...
    -- Executing [3110@vip:1] MeetMe("SIP/montreal-00000002", "101") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
[Nov 22 15:24:31] WARNING[22140][C-00000003]: app_meetme.c:1296 build_conf: Unable to open DAHDI pseudo device
  == Spawn extension (vip, 3110, 1) exited non-zero on 'SIP/montreal-00000002'
[Nov 22 15:24:38] NOTICE[21979]: chan_sip.c:15071 sip_reg_timeout:    -- Registration for 'ottawa@192.168.15.254' timed out, trying again (Attempt #8)
debian*CLI> lsmod dahdi
No such command 'lsmod dahdi' (type 'core show help lsmod dahdi' for other possible commands)
debian*CLI> exit
Asterisk cleanly ending (0).
Executing last minute cleanups
/dev/pts/3
14:23:21
#make menuconfig
                                                                  Press 'h' for help.
                                                                  Add-ons (See README-addons.txt)
                                                                  Applications
                                                                  Bridging Modules
                                                                  Call Detail Recording
                                                                  Channel Event Logging
                                                            --->  Channel Drivers
                                                                  Codec Translators
                                                                  Format Interpreters
                                                                  Dialplan Functions
...
                                                                  Compiler Flags
                                                                  Voicemail Build Options
                                                                  Utilities
                                                                  AGI Samples
                                                                  Module Embedding
                                                                  Core Sound Packages
                                                                  Music On Hold File Packages
                                                                  Extras Sound Packages
menuselect changes NOT saved!
make[1]: Leaving directory `/usr/src/asterisk-11.6.0'
14:23:39
#pwd
/usr/src/asterisk-11.6.0
/dev/pts/1
14:24:44
#lsmod dahdi
Usage: lsmod
14:24:49
#lsmod | grep dahdi

14:25:08
#lsmod | greep dahdi
l3script: greep: command not found
14:25:13
#lsmod | grep dahdi

14:25:15
#lsmod | grep dahdi

14:27:32
#lsmod | grep dahdi

/dev/pts/3
14:29:02
#make
CC="cc" CXX="" LD="" AR="" RANLIB="" CFLAGS="" LDFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" makeopts
make[1]: Entering directory `/usr/src/asterisk-11.6.0/menuselect'
make[1]: `makeopts' is up to date.
make[1]: Leaving directory `/usr/src/asterisk-11.6.0/menuselect'
Generating embedded module rules ...
   [CC] astcanary.c -> astcanary.o
   [LD] astcanary.o -> astcanary
   [CC] astdb2sqlite3.c -> astdb2sqlite3.o
   [CC] hash/hash.c -> hash/hash.o
   [CC] hash/hash_bigkey.c -> hash/hash_bigkey.o
...
   [CC] app_zapateller.c -> app_zapateller.o
   [LD] app_zapateller.o -> app_zapateller.so
   [CC] src/add.c -> src/add.o
   [CC] src/code.c -> src/code.o
   [CC] src/debug.c -> src/debug.o
   [CC] src/decode.c -> src/decode.o
^Cmake[2]: *** Deleting file `src/decode.o'
make[2]: *** [src/decode.o] Interrupt
make[1]: *** [gsm] Interrupt
make: *** [codecs] Interrupt
14:30:52
#make install
CC="cc" CXX="" LD="" AR="" RANLIB="" CFLAGS="" LDFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" makeopts
make[1]: Entering directory `/usr/src/asterisk-11.6.0/menuselect'
make[1]: `makeopts' is up to date.
make[1]: Leaving directory `/usr/src/asterisk-11.6.0/menuselect'
Installing modules from channels...
Installing modules from pbx...
Installing modules from apps...
Installing modules from codecs...
Installing modules from formats...
Installing modules from cdr...
...
 +-----------------  or ---------------------+
 +                                           +
 + You can go ahead and install the asterisk +
 + program documentation now or later run:   +
 +                                           +
 +               make progdocs               +
 +                                           +
 + **Note** This requires that you have      +
 + doxygen installed on your local system    +
 +-------------------------------------------+
14:31:08
#modprobe dahid
FATAL: Module dahid not found.
/dev/pts/1
14:31:14
#apt-get
apt 0.9.7.9 for amd64 compiled on Jun  4 2013 09:30:58
Usage: apt-get [options] command
       apt-get [options] install|remove pkg1 [pkg2 ...]
       apt-get [options] source pkg1 [pkg2 ...]
apt-get is a simple command line interface for downloading and
installing packages. The most frequently used commands are update
and install.
Commands:
   update - Retrieve new lists of packages
   upgrade - Perform an upgrade
...
  -f  Attempt to correct a system with broken dependencies in place
  -m  Attempt to continue if archives are unlocatable
  -u  Show a list of upgraded packages as well
  -b  Build the source package after fetching it
  -V  Show verbose version numbers
  -c=? Read this configuration file
  -o=? Set an arbitrary configuration option, eg -o dir::cache=/tmp
See the apt-get(8), sources.list(5) and apt.conf(5) manual
pages for more information and options.
                       This APT has Super Cow Powers.
/dev/pts/3
14:31:28
#depmod -a

14:31:47
#modprobe dahid
FATAL: Module dahid not found.
14:31:54
#modprobe dahid
FATAL: Module dahid not found.
14:32:16
#depmod -a

14:33:02
#./configure --with-srtp --with-ssl --with-dahdi --with-pri --with-tonezone | grep dahdi
checking for DAHDI_RESET_COUNTERS in dahdi/user.h... yes
checking for DAHDI_POLICY_HALF_FULL in dahdi/user.h... yes
checking for enhanced dahdi vmwi support... yes
checking if "int foo = DAHDI_ECHOCANCEL_FAX_MODE" compiles using dahdi/user.h... yes
checking dahdi/tonezone.h usability... yes
checking dahdi/tonezone.h presence... yes
checking for dahdi/tonezone.h... yes
14:34:00
#make clean all
rm -f defaults.h
rm -f include/asterisk/build.h
rm -f main/version.c
rm -f doc/core-en_US.xml
rm -f doc/full-en_US.xml
make[1]: Entering directory `/usr/src/asterisk-11.6.0/menuselect'
rm -f menuselect cmenuselect gmenuselect nmenuselect menuselect.o strcompat.o menuselect_stub.o menuselect_curses.o
make[2]: Entering directory `/usr/src/asterisk-11.6.0/menuselect/mxml'
/bin/rm -f mxmldoc.o testmxml.o mxml-attr.o mxml-entity.o mxml-file.o mxml-index.o mxml-node.o mxml-search.o mxml-set.o mxml-private.o mxml-string.o libmxml.a mxmldoc testmxml mxml.xml
/bin/rm -f mxmldoc-static libmxml.a
...
creating makelist
creating config.h
   [LD] abstract_jb.o acl.o adsi.o alaw.o aoc.o app.o ast_expr2.o ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o audiohook.o autochan.o autoservice.o bridging.o callerid.o ccss.o cdr.o cel.o channel.o channel_internal_api.o chanvars.o cli.o config.o config_options.o data.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o event.o features.o file.o fixedjitterbuf.o format.o forma
Building Documentation For: channels pbx apps codecs formats cdr cel bridges funcs tests main res addons
 +--------- Asterisk Build Complete ---------+
 + Asterisk has successfully been built, and +
 + can be installed by running:              +
 +                                           +
 +                make install               +
 +-------------------------------------------+
/dev/pts/1
14:35:05
#apt-get remove asterisk
Reading package lists... Done
Building dependency tree
Reading state information... Done
Package 'asterisk' is not installed, so not removed
0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded.
14:35:15
#asterisk -rvvvv
Asterisk 11.6.0, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 11.6.0 currently running on debian (pid = 21926)
  == Using SIP RTP CoS mark 5
[Nov 22 15:36:14] NOTICE[21979][C-0000000b]: chan_sip.c:25399 handle_request_invite: Call from 'ottawa' (192.168.15.8:5060) to extension '2191' rejected because extension not found in context 'vip'.
...
[Nov 22 15:41:18] NOTICE[21979]: chan_sip.c:15071 sip_reg_timeout:    -- Registration for 'ottawa@192.168.15.254' timed out, trying again (Attempt #58)
[Nov 22 15:41:38] NOTICE[21979]: chan_sip.c:15071 sip_reg_timeout:    -- Registration for 'ottawa@192.168.15.254' timed out, trying again (Attempt #59)
[Nov 22 15:41:58] NOTICE[21979]: chan_sip.c:15071 sip_reg_timeout:    -- Registration for 'ottawa@192.168.15.254' timed out, trying again (Attempt #60)
[Nov 22 15:42:18] NOTICE[21979]: chan_sip.c:15071 sip_reg_timeout:    -- Registration for 'ottawa@192.168.15.254' timed out, trying again (Attempt #61)
[Nov 22 15:42:38] NOTICE[21979]: chan_sip.c:15071 sip_reg_timeout:    -- Registration for 'ottawa@192.168.15.254' timed out, trying again (Attempt #62)
[Nov 22 15:42:58] NOTICE[21979]: chan_sip.c:15071 sip_reg_timeout:    -- Registration for 'ottawa@192.168.15.254' timed out, trying again (Attempt #63)
debian*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
/dev/pts/5
14:36:27
#joe /etc/asterisk/extensions.conf
         s,n,WaitExten(3)
 bank]
 super]  s,n(main),Background(/var/tmp/asterisk/records/01)
;exten => _33XX,1,Dial(SIP/3101)
include => localtExten(3)
;exten => _inter1,Dial(SIP/31${EXTEN:2})
include => nationalound(/var/tmp/asterisk/records/02)
           national
include => international
include => international
...
;exten => _31xx,1,Dial(SIP/${EXTEN},2)
 local]   32XX,1,Answer
exten => 3101,1,Dial(SIP/3101,5,tT)${EXTEN})
exten => 3104,1,Dial(SIP/3102&SIP/3103)
         3102,1,Dial(SIP/3102)
         3110,1,MeetMe(101)
[national]
          32XX,1,Answer
exten => _32XX,n,Dial(SIP/montreal/${EXTEN})
File /etc/asterisk/extensions.conf not changed so no update needed
14:36:50
#joe /etc/asterisk/sip.conf
[montreal]
[3103]riend
[3103]
type=friend
type=friend
secret=1234c
username=ottawa
host=dynamic
context=vip
language=ru
...
[london]
type=friend
secret=1234
host=dynamic
username=ottawa
[kiev]
type=friend
secret=1234
username=ottawa
File /etc/asterisk/sip.conf not changed so no update needed
14:37:44
#joe /etc/asterisk/extensions.conf
exten => 3103,1,Dial(SIP/3103)
;exten => _31xx,1,Dial(SIP/${EXTEN},2)
exten => 3104,1,Dial(SIP/3102&SIP/3103)
exten => 3110,1,MeetMe(101)
  ational]                                                           87
[super]
include => local
include => national
include => international
exten => s,n,Hangup
...
exten => 2,n(invalid_order),Playback(/var/tmp/asterisk/records/09)
         s,n,WaitExten(3)
exten => s,n,Background(/var/tmp/asterisk/records/02)
exten => s,n,WaitExten(3)
exten => 3,1,Playback(/var/tmp/asterisk/records/10)
         s,n,Background(/var/tmp/asterisk/records/03)
         3,n,Dial(SIP/3103)
         s,n,Hangup
         3,n,Hangup
exten => 2,1,Read(ORDER,/var/tmp/asterisk/records/05
/dev/pts/3
14:39:54
#pwd
/home/alex
14:39:55
#cd /usr/src/
asterisk-11.6.0/                                              linux-headers-3.2.0-4-amd64/
asterisk-11.6.0.tar.gz                                        linux-headers-3.2.0-4-common/
dahdi-modules-3.2.0-4-amd64_2.6.1+dfsg2-1+3.2.51-1_amd64.deb  linux-kbuild-3.2/
dahdi.tar.bz2                                                 modules/
linux/
14:39:55
#cd /usr/src/

14:40:01
#rm -r asterisk-11.6.0/

14:40:11
#ls -la
total 40276
drwxr-xr-x  5 root root     4096 Nov 22 15:40 .
drwxr-xr-x 10 root root     4096 Nov 18 15:20 ..
-rw-r--r--  1 root root 34767605 Nov 22 13:23 asterisk-11.6.0.tar.gz
-rw-r--r--  1 root root  5605980 Nov 19 16:24 dahdi-modules-3.2.0-4-amd64_2.6.1+dfsg2-1+3.2.51-1_amd64.deb
-rw-r--r--  1 root root   843676 Jan 30  2013 dahdi.tar.bz2
lrwxrwxrwx  1 root root       27 Nov 19 16:23 linux -> linux-headers-3.2.0-4-amd64
drwxr-xr-x  4 root root     4096 Nov 19 16:22 linux-headers-3.2.0-4-amd64
drwxr-xr-x  4 root root     4096 Nov 19 16:22 linux-headers-3.2.0-4-common
lrwxrwxrwx  1 root root       23 Jun 24  2012 linux-kbuild-3.2 -> ../lib/linux-kbuild-3.2
drwxr-xr-x  3 root root     4096 Jan 30  2013 modules
14:40:14
#tar -xf asterisk-11.6.0.tar.gz

14:40:22
#cd asterisk-11.6.0
asterisk-11.6.0/        asterisk-11.6.0.tar.gz
14:40:22
#cd asterisk-11.6.0

14:40:25
#./configure --with-srtp --with-ssl --with-dahdi --with-pri --with-tonezone | grep dahdi
^C
14:40:44
#./configure --with-srtp --with-ssl --with-dahdi --with-pri --with-tonezone
checking build system type... x86_64-unknown-linux-gnu
checking host system type... x86_64-unknown-linux-gnu
checking for gcc... gcc
checking whether the C compiler works... yes
checking for C compiler default output file name... a.out
checking for suffix of executables...
checking whether we are cross compiling... no
checking for suffix of object files... o
checking whether we are using the GNU C compiler... yes
checking whether gcc accepts -g... yes
...
 $$$$$                        $$$
  $$$$7.                       $$  (TM)
   $$$$$$$.           .7$$$$$$  $$
     $$$$$$$$$$$$7$$$$$$$$$.$$$$$$
       $$$$$$$$$$$$$$$$.
configure: Package configured for:
configure: OS type  : linux-gnu
configure: Host CPU : x86_64
configure: build-cpu:vendor:os: x86_64 : unknown : linux-gnu :
configure: host-cpu:vendor:os: x86_64 : unknown : linux-gnu :
/dev/pts/1
14:43:13
#/etc/init.d/asterisk stop
[ ok ] Stopping Asterisk PBX: asterisk.
14:43:21
#htop
l3script: htop: command not found
14:43:29
#atop
l3script: atop: command not found
14:43:31
#top
top - 15:43:38 up 3 days,  5:24,  5 users,  load average: 6.47, 2.12, 1.11
Tasks: 265 total,  33 running, 232 sleeping,   0 stopped,   0 zombie
%Cpu(s): 57.8 us,  3.1 sy,  0.0 ni, 39.1 id,  0.0 wa,  0.0 hi,  0.0 si,  0.0 st
KiB Mem:  16357056 total,  3201148 used, 13155908 free,   220560 buffers
KiB Swap: 16662524 total,        0 used, 16662524 free,  1862032 cached
  PID USER      PR  NI  VIRT  RES  SHR S  %CPU %MEM    TIME+  COMMAND
16283 root      20   0  327m 302m 8068 R  98.1  1.9   0:16.20 cc1
 3163 alex      20   0  980m 126m  28m S   6.5  0.8   5:03.33 gnome-shell
    1 root      20   0 10644  820  688 S   0.0  0.0   0:02.77 init
    2 root      20   0     0    0    0 S   0.0  0.0   0:00.03 kthreadd
...
 3163 alex      20   0  980m 126m  28m S   2.3  0.8   5:03.55 gnome-shell
 3278 alex      20   0  374m  22m  11m S   1.3  0.1   2:06.23 gnome-terminal
19631 root      20   0 10636 1744  948 S   0.7  0.0   0:00.03 make
19718 root      20   0 17248 8720 1348 S   0.7  0.1   0:00.02 as
19783 root      20   0 16192 7664 1348 S   0.7  0.0   0:00.02 as
   35 root      20   0     0    0    0 S   0.3  0.0   0:00.06 ksoftirqd/7
  169 root      20   0     0    0    0 S   0.3  0.0   0:03.84 kworker/5:1
  183 root      20   0     0    0    0 S   0.3  0.0   0:02.71 kworker/u:3
 2274 root      20   0     0    0    0 S   0.3  0.0   0:01.92 flush-8:0
17122 root      20   0     0    0    0 S   0.3  0.0   0:02.30 kworker/7:2
14:43:46
#/etc/init.d/asterisk -rvv
Usage: /etc/init.d/asterisk {start|stop|restart|reload|force-reload|status}
/dev/pts/3
14:43:48
#make install
Installing modules from channels...
Installing modules from pbx...
Installing modules from apps...
Installing modules from codecs...
Installing modules from formats...
Installing modules from cdr...
Installing modules from cel...
Installing modules from bridges...
Installing modules from funcs...
Installing modules from tests...
...
 version of Asterisk. Please ensure that these
 modules are compatible with this version before
 attempting to run Asterisk.
    app_dahdibarge.so
    app_ivrdemo.so
    app_readfile.so
    app_saycounted.so
    app_setcallerid.so
    app_skel.so
 WARNING WARNING WARNING
14:44:18
#modprobe -a
Error: missing parameters. See -h.
14:44:35
#modprobe -a
Error: missing parameters. See -h.
14:44:37
#modprobe -a
Error: missing parameters. See -h.
14:45:46
#depmod -a

14:45:48
#modprobe dahdi

14:46:12
#lsmod
Module                  Size  Used by
dahdi                 191056  0
crc_ccitt              12347  1 dahdi
bnep                   17567  2
rfcomm                 33700  0
bluetooth             119455  10 rfcomm,bnep
binfmt_misc            12957  1
nls_utf8               12456  1
nls_cp437              16553  1
vfat                   17316  1
...
cdrom                  35401  1 sr_mod
crc_t10dif             12348  1 sd_mod
ahci                   24997  3
libahci                22860  1 ahci
libata                140630  2 libahci,ahci
scsi_mod              162269  4 libata,sd_mod,sr_mod,sg
ehci_hcd               40215  0
usbcore               128741  3 ehci_hcd,usbhid
usb_common             12354  1 usbcore
e1000e                120822  0
14:46:28
#lsmod | grep dahdi
dahdi                 191056  0
crc_ccitt              12347  1 dahdi
14:46:34
#joe /etc/asterisk/extensions.conf
exten => 8201,1,Queue(queue1)
exten => _83XX,1,AgentLogin(0${EXTEN:2})
exten => _32XX,n,Dial(SIP/montreal/${EXTEN})
[local]
exten => 3101,1,Dial(SIP/3101,5,tT)
exten => 3102,1,Dial(SIP/3102)ec/${EXTEN})
** Joe's Own Editor v3.7 ** (utf-8) ** Copyright © 2008 **
File /etc/asterisk/extensions.conf not changed so no update needed
/dev/pts/1
14:46:49
#/etc/init.d/asterisk start
[ ok ] Starting Asterisk PBX: asterisk.
прошло 20 минут
/dev/pts/3
15:07:09
#joe /etc/asterisk/sip.conf
callgroup=1
context=vip
[general]
callgroup=1
 ontext=default
pickupgroup=1
directmedia=no
register => ottawa:1234@192.168.15.8/montreal
[3103]
register => ottawa:1234@192.168.15.254/quebec
...
pickupgroup=1
[quebec]
pickupgroup=1
[montreal]ttawa
type=friend
secret=1234
host=dynamic
username=ottawa
context=vip
[quebec]
прошло 50 минут
15:57:49