13:02:23
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#screen
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| /l3/users/kronas/nt-voip/linux5.unix.nt/root :1 :2 :3 :4 :5 :6 :7 :8 :9 :10 |
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#nano meetme.conf
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#;
;
; Configuration file for MeetMe simple conference rooms for Asterisk of course.
;
; This configuration file is read every time you call app meetme()
; when feeding audio frames from non-DAHDI channels
; into the conference; larger numbers will allow
; for the conference to 'de-jitter' audio that arrives
[general] ; at different timing than the conference's timing
; source, but can also allow for latency in hearing
;audiobuffers=32 ; the audio from the speaker. Minimum value is 2,
...
; in the conference and it goes away. When it is created again, it will have
; the new pin number.
;
;conf => 1234
;conf => 2345,9938
44,1 Bot
conf =>
conf =>
conf =>
49,1 Bot
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#asterisk -r
Asterisk 1.6.2.9-2+squeeze10, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 1.6.2.9-2+squeeze10 currently running on linux5 (pid = 5926) linux5*CLI> meetme kick lock mute unlock unmute list ... == Using SIP RTP CoS mark 5 -- Executing [1234@user_group3:1] Dial("SIP/2101-00000002", "SIP/minsk/1234") in new stack == Using SIP RTP CoS mark 5 -- Called minsk/1234 -- SIP/minsk-00000003 answered SIP/2101-00000002 -- Packet2Packet bridging SIP/2101-00000002 and SIP/minsk-00000003 == Spawn extension (user_group3, 1234, 1) exited non-zero on 'SIP/2101-00000002' [Apr 10 14:07:50] NOTICE[5948]: chan_sip.c:18485 handle_response_peerpoke: Peer '2110' is now Lagged. (2062ms / 2000ms) [Apr 10 14:08:14] NOTICE[5948]: chan_sip.c:18485 handle_response_peerpoke: Peer '2110' is now Reachable. (607ms / 2000ms) linux5*CLI> exit |
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#asterisk -r
Asterisk 1.6.2.9-2+squeeze10, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 1.6.2.9-2+squeeze10 currently running on linux5 (pid = 5926) Verbosity is at least 3 linux5*CLI> module reload app_meetme.so ... == Using SIP RTP CoS mark 5 -- Executing [2102@user_group3:1] Answer("SIP/2101-00000015", "") in new stack -- Executing [2102@user_group3:2] Dial("SIP/2101-00000015", "SIP/2102,10,wWtT") in new stack == Using SIP RTP CoS mark 5 -- Called 2102 -- SIP/2102-00000016 is ringing -- SIP/2102-00000016 answered SIP/2101-00000015 -- Music class default requested but no musiconhold loaded. == Spawn extension (user_group3, 2102, 2) exited non-zero on 'SIP/2101-00000015' linux5*CLI> exit |
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#~
same => n,Playback(demo-congrats)
exten => _81XX,1,Record(/var/tmp/message/${EXTEN:2}:gsm,1)
exten => _81XX,n,Playback(/var/tmp/message/${EXTEN:2})
exten => _82XX,1,Playback(/var/tmp/message/${EXTEN:2})
exten => _8300,1,Voicemailmain()
[local_monitor]f" 72L, 1588C
exten => _21XX,1,Answer
same => n,Set(CALLFILENAME=${EXTEN}-${CALLERID(num)}-${EPOCH})
same => n,Monitor(wav,${CALLFILENAME},m)
same => n,Dial(SIP/${EXTEN})
...
[ ]
[xten => ]21XX,1,Answer
[ ]
same => n,Dial(SIP/${EXTEN},10,wWtT)
same => n,Voicemail(${EXTEN}@default)
[national]
exten => _22XX,1,Dial(SIP/minsk/${EXTEN})
exten => _23XX,1,Dial(SIP/minsk/${EXTEN})
exten => _23XX,1,Dial(SIP/minsk/${EXTEN})
-- INSERT -- 19,1 Top
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#apt-get install mpg123
Reading package lists... Done Building dependency tree Reading state information... Done The following packages were automatically installed and are no longer required: mysql-common libmysqlclient16 Use 'apt-get autoremove' to remove them. The following extra packages will be installed: libmpg123-0 Suggested packages: libartsc0 libaudio2 ... Selecting previously deselected package libmpg123-0. (Reading database ... 128690 files and directories currently installed.) Unpacking libmpg123-0 (from .../libmpg123-0_1.12.1-3_i386.deb) ... Selecting previously deselected package mpg123. Unpacking mpg123 (from .../mpg123_1.12.1-3_i386.deb) ... Processing triggers for man-db ... Setting up libmpg123-0 (1.12.1-3) ... Setting up mpg123 (1.12.1-3) ... update-alternatives: using /usr/bin/mpg123.bin to provide /usr/bin/mpg123 (mpg123) in auto mode. update-alternatives: using /usr/bin/mpg123.bin to provide /usr/bin/mp3-decoder (mp3-decoder) in auto mode. |
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#cd /home/user/
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#mpg123 -w moh_music.wav moh_music.mp3
High Performance MPEG 1.0/2.0/2.5 Audio Player for Layers 1, 2 and 3
version 1.12.1; written and copyright by Michael Hipp and others
free software (LGPL/GPL) without any warranty but with best wishes
Playing MPEG stream 1 of 1: moh_music.mp3 ...
Title: Шалай-валяй (zaycev.net)
Artist: Чайф
Comment: (zaycev.net)
MPEG 1.0 layer III, VBR, 44100 Hz stereo
[4:00] Decoding of moh_music.mp3 finished.
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#sox -t wav moh_music.wav -r 8000 -c1 -t gsm moh_music.gsm resample -ql
sox WARN sox: effect `resample' is deprecated; see sox(1) for an alternative sox WARN resample: resample clipped 57 samples; decrease volume? |
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#sox -t wav moh_music.wav -r 8000 -c1 -t gsm moh_music.gsm rate -ql
sox WARN rate: rate clipped 35 samples; decrease volume? |
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#cp moh_music.gsm /var/lib/asterisk/moh/
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#cd /var/lib/asterisk/moh/
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#ls -l
total 392 -rw-r--r-- 1 root root 396825 Apr 10 14:36 moh_music.gsm |
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#cd /etc/asterisk/
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#vim musiconhold.conf
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#; understand when it loads.
; /var/lib/asterisk. ; NOTE: ; If you are not using "autoload" in modules.conf, then you ; must ensure that the format modules for any formats you wish ; to use are loaded _before_ res_musiconhold. If you do not do ; this, res_musiconhold will skip the files it is not able to ; understand when it loads. ; [default] [ ] directory=moh ; ;[native-random] ;mode=files ;directory=moh ;digit=# ; If this option is set for a class, then when callers are mode=files 52,1 33% |
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#/etc/init.d/asterisk restart
Stopping Asterisk PBX: asterisk. Starting Asterisk PBX: asterisk. |
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#asterisk -r
Asterisk 1.6.2.9-2+squeeze10, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 1.6.2.9-2+squeeze10 currently running on linux5 (pid = 6717) linux5*CLI> moh show classes files ... in it jp lt mx my nl no nz ph pl pt ru se sg th tw uk us us-o ve za bg linux5*CLI> module reload indications [Apr 10 14:50:34] WARNING[6763]: config.c:1163 process_text_line: No '=' (equal sign) in line 736 of /etc/asterisk/indications.conf [Apr 10 14:50:39] NOTICE[6739]: chan_sip.c:18485 handle_response_peerpoke: Peer '2110' is now Reachable. (693ms / 2000ms) linux5*CLI> module reload indications linux5*CLI> dialplan reload Dialplan reloaded. [Apr 10 14:51:42] NOTICE[6739]: chan_sip.c:18485 handle_response_peerpoke: Peer '2110' is now Lagged. (2841ms / 2000ms) linux5*CLI> exit |
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#vim indications.conf
--- /tmp/l3-saved-6477.1086.3675 2013-04-10 14:48:57.000000000 +0300 +++ indications.conf 2013-04-10 14:49:30.000000000 +0300 @@ -729,3 +729,9 @@ record = 1400/500,0/10000 info = 950/330,1400/330,1800/330,0/330 stutter = !400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,400*33 + +[starwars](us) +description = Star Wars Theme Song +ring = 262/400,392/500,0/100,349/400,330/400,294/400,524/400,392/500,0/100,349/400, \ + 330/400,294/400,524/400,392/500,0/100,349/400,330/400,349/400,294/500,0/2000 + |
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#description = Star Wars Theme Song
exten => _8300,1,Voicemailmain() exten => _8301,1,Meetme(1234) exten => _8302,1,Meetme(1234,m) exten => _8303,1,Meetme(1234,t) exten => _8304,1,Meetme(1234,as) exten => _8305,1,Meetme(1234,s) exten => _8306,1,Meetme(1234,sq) exten => _8309,1,Meetme(9999) [local_monitor] exten => _21XX,1,Answer ... exten => _2121,1,Answer same => n,Dial(SIP/2101&SIP/2102) 2,0-1 Top [ ] [ ] [local] [ ] exten => _21XX,1,Answer same => n,Dial(SIP/${EXTEN},10,wWtT) same => n,Voicemail(${EXTEN}@default) 39,1 6% |
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#/etc/init.d/asterisk restart
Stopping Asterisk PBX: asterisk. Starting Asterisk PBX: asterisk. |
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#asterisk -r
Asterisk 1.6.2.9-2+squeeze10, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 1.6.2.9-2+squeeze10 currently running on linux5 (pid = 6894) linux5*CLI> module reload indications linux5*CLI> module reload indications ... linux5*CLI> module reload indications linux5*CLI> dialplan reload Dialplan reloaded. [Apr 10 14:54:58] NOTICE[6916]: chan_sip.c:18485 handle_response_peerpoke: Peer '2110' is now Reachable. (201ms / 2000ms) [Apr 10 14:56:02] NOTICE[6916]: chan_sip.c:23124 sip_poke_noanswer: Peer '2110' is now UNREACHABLE! Last qualify: 0 [Apr 10 14:56:43] NOTICE[6916]: chan_sip.c:18485 handle_response_peerpoke: Peer '2110' is now Reachable. (174ms / 2000ms) linux5*CLI> module reload indications linux5*CLI> core set verbose 3 Verbosity was 0 and is now 3 linux5*CLI> quit |
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#vim extensions.conf
--- /tmp/l3-saved-6477.29109.9241 2013-04-10 15:00:32.000000000 +0300
+++ extensions.conf 2013-04-10 15:01:43.000000000 +0300
@@ -35,6 +35,7 @@
same => n,Dial(SIP/2101&SIP/2102)
exten => _21XX,1,Answer
+ same => n,Set(CHANNEL(tonezone)=starwars)
same => n,Dial(SIP/${EXTEN},10,wWtT)
same => n,Voicemail(${EXTEN}@default)
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#vim indications.conf
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#vim queues.conf
--- /tmp/l3-saved-6477.22967.12696 2013-04-10 15:02:12.000000000 +0300
+++ queues.conf 2013-04-10 15:04:18.000000000 +0300
@@ -498,3 +498,7 @@
;member => Agent/:1,1 ; Any agent in group 1, wait for first
; available, but consider with penalty
+[queue_main]
+; General queue section
+member => SIP/2101
+member => SIP/2102
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#vim extensions.conf
--- /tmp/l3-saved-6477.6749.23014 2013-04-10 15:04:19.000000000 +0300
+++ extensions.conf 2013-04-10 15:05:56.000000000 +0300
@@ -24,6 +24,8 @@
exten => _8306,1,Meetme(1234,sq)
exten => _8309,1,Meetme(9999)
+exten => 2105,1,Queue(queue_main)
+
[local_monitor]
exten => _21XX,1,Answer
same => n,Set(CALLFILENAME=${EXTEN}-${CALLERID(num)}-${EPOCH})
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#vim sip.conf
--- /tmp/l3-saved-6477.1094.19528 2013-04-10 15:40:59.000000000 +0300 +++ sip.conf 2013-04-10 15:42:02.000000000 +0300 @@ -48,6 +48,8 @@ callgroup=2 pickupgroup=2 +qualify=no + [2111] context=user_group3 type=friend |
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#vim queues.conf
--- /tmp/l3-saved-6477.16888.9089 2013-04-10 15:42:57.000000000 +0300 +++ queues.conf 2013-04-10 15:52:13.000000000 +0300 @@ -500,5 +500,10 @@ [queue_main] ; General queue section -member => SIP/2101 -member => SIP/2102 +;member => SIP/2101 +;member => SIP/2102 + +member => Agent/1001 +member => Agent/1002 + +strategy = linear |
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#vim agents.conf
--- /tmp/l3-saved-6477.9026.25557 2013-04-10 15:52:16.000000000 +0300 +++ agents.conf 2013-04-10 15:54:25.000000000 +0300 @@ -109,5 +109,5 @@ ; ; agent => agentid,agentpassword,name ; -;agent => 1001,4321,Mark Spencer -;agent => 1002,4321,Will Meadows +agent => 1001,4321,Mark Spencer +agent => 1002,4321,Will Meadows |
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