| /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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