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Пример - конфигурация Quasar/Quasar-mini для Asterisk.
Основная статья Отладка Ниже приведена следующая конфигурация: - Порт 1 настроен как LTE (сигнализация PRI_CPE).
Этот порт вы можете подключить к ГАТС. - Порт 2 настроен как NTE (сигнализация PRI_NET).
Этот порт вы можете подключить к вашей (внутренней) АТС.
Скачать конфигурационные файлы. Внимание! Конфигурационные файлы должны быть доступны на чтение процессу asterisk. #sudo chown asterisk.asterisk /etc/asterisk/* #sudo sudo chmod u+r,g+r /etc/asterisk/*
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/etc/dahdi/modulesquasar quasarm | /etc/dahdi/system.conf # Span Configuration # ^^^^^^^^^^^^^^^^^^ # First come the span definitions, in the format # # span=<span num>,<timing source>,<line build out (LBO)>,<framing>,<coding>[,yellow] # # All T1/E1/BRI spans generate a clock signal on their transmit side. The # <timing source> parameter determines whether the clock signal from the far # end of the T1/E1/BRI is used as the master source of clock timing. If it is, our # own clock will synchronise to it. T1/E1/BRI connected directly or indirectly to # a PSTN provider (telco) should generally be the first choice to sync to. The # PSTN will never be a slave to you. You must be a slave to it. # # Choose 1 to make the equipment at the far end of the E1/T1/BRI link the preferred # source of the master clock. Choose 2 to make it the second choice for the master # clock, if the first choice port fails (the far end dies, a cable breaks, or # whatever). Choose 3 to make a port the third choice, and so on. If you have, say, # 2 ports connected to the PSTN, mark those as 1 and 2. The number used for each # port should be different. # # If you choose 0, the port will never be used as a source of timing. This is # appropriate when you know the far end should always be a slave to you. If # the port is connected to a channel bank, for example, you should always be # its master. Likewise, BRI TE ports should always be configured as a slave. # Any number of ports can be marked as 0. # # Incorrect timing sync may cause clicks/noise in the audio, poor quality or failed # faxes, unreliable modem operation, and is a general all round bad thing.
# SPAN 1, timing source (E1 slave), HDB3, CRC4, CCS. span=1,1,0,ccs,hdb3,crc4
# SPAN 2, timing target (E1 master), HDB3, CRC4, CCS. span=2,0,0,ccs,hdb3,crc4
# SPAN 1, CCS(PRI/SS7/...), A-law, OSLEC echocanceller. bchan=1-15,17-31 dchan=16 alaw=1-15,17-31 echocanceller=oslec,1-15,17-31
# SPAN 2, CCS(PRI/SS7/...), A-law, OSLEC echocanceller. bchan=32-46,48-62 dchan=47 alaw=32-46,48-62 echocanceller=oslec,32-46,48-62
# Setting correct zone info (tone info) loadzone=ru defaultzone=ru
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/etc/asterisk/chan_dahdi.conf[channels] relaxdtmf=yes ; pavel - necessary on bad lines... rxflash=850
;== busy busydetect=yes busycount=3
;== pulse pulsedial=no pulse=no
;== Calls handling == callwaiting=yes ;If enabled, Asterisk will generate "call waiting pips" when you are already in a conversation ;{ threewaycalling=yes transfer=yes ;} cancallforward=yes immediate=no ;yes to let Asterisk handle 's' extention (passing control immediately to Asterisk)
;== Echo == echocancel = yes echocancelwhenbridged = yes echotraining=no
;== Caller ID == usecallerid=yes callerid=asreceived ;Unrecognized prilocaldialplan NPI modifier: s ;don't have a time to find the reason. disabling... hidecallerid=yes ;;callwaitingcallerid=yes useincomingcalleridonzaptransfer=yes ; cidsignalling=bell cidstart=ring restrictcid=no callreturn=yes ;dial *69 to have Asterisk read to you the caller ID of the last person to call pridialplan=national prilocaldialplan=national internationalprefix= nationalprefix= localprefix=
;==== FXO lines (external PBX) ==== group=0 context=pri0 signalling=pri_cpe switchtype=euroisdn channel => 1-15,17-31 group = context = default
group=1 context=pri1 signalling=pri_net switchtype=euroisdn channel => 32-46,48-62 group = context = default
| /etc/asterisk/extensions.conf [general] static=yes writeprotect=yes autofallthrough=yes clearglobalvars=no ;priorityjumping=no
[globals] CONSOLE=Console/dsp
[default] exten => s,1,Macro(mylog, "BUG: Unhandled context !!!") exten => s,n,Hangup() exten => _X.,1,Macro(mylog, "BUG: Unhandled context !!!") exten => _X.,n,Hangup()
[test_context] exten => _1.,1,Wait(0) ; PSTN can reject an outgoing call without caller id. ;exten => _1.,n,Set(CALLERID(all)=84950000000) exten => _1.,n, Dial(DAHDI/g0/${EXTEN:1}) exten => _1.,n,Hangup()
exten => _2.,1,Wait(0) ; Our PBX should send caller ID to the client PBX ;exten => _2.,n,Set(CALLERID(all)=84950000001) exten => _2.,n, Dial(DAHDI/g1/${EXTEN:1}) exten => _2.,n,Hangup()
exten => 310,1, Dial(SIP/sip1) exten => 320,1, Dial(SIP/sip2)
exten => 91,1, Answer exten => 91,n, Wait(3600)
exten => 92,1, Answer exten => 92,n, Echo()
exten => 93,1,Answer exten => 93,n,MP3Player(/etc/asterisk/1.mp3)
[pri_base] exten => _X.,1,Answer() exten => _X.,n,Verbose(Received callerID: ${CALLERID(all)}) exten => _X.,n,Echo() ;exten => _X.,n,MP3Player(/etc/asterisk/1.mp3) exten => _X.,n,Hangup()
exten => s,1,Answer() exten => s,n,Verbose(Received callerID: ${CALLERID(all)}) exten => s,n,Echo() ;exten => s,n,MP3Player(/etc/asterisk/1.mp3) exten => s,n,Hangup()
[pri0] include => pri_base
[pri1] include => pri_base
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/etc/asterisk/logger.conf; ; Logging Configuration ; ; In this file, you configure logging to files or to ; the syslog system. ; ; "logger reload" at the CLI will reload configuration ; of the logging system.
[general] ; ; Customize the display of debug message time stamps ; this example is the ISO 8601 date format (yyyy-mm-dd HH:MM:SS) ; ; see strftime(3) Linux manual for format specifiers. Note that there is also ; a fractional second parameter which may be used in this field. Use %1q ; for tenths, %2q for hundredths, etc. ; ;dateformat=%F %T ; ISO 8601 date format ;dateformat=%F %T.%3q ; with milliseconds ; ; This appends the hostname to the name of the log files. ;appendhostname = yes ; ; This determines whether or not we log queue events to a file ; (defaults to yes). ;queue_log = no ; ; Set the queue_log filename ; (defaults to queue_log) ;queue_log_name = queue_log ; ; Log rotation strategy: ; sequential: Rename archived logs in order, such that the newest ; has the highest sequence number [default]. ; rotate: Rotate all the old files, such that the oldest has the ; highest sequence number [this is the expected behavior ; for Unix administrators]. ; timestamp: Rename the logfiles using a timestamp instead of a ; sequence number when "logger rotate" is executed. ;rotatestrategy = rotate ; ; Run a system command after rotating the files. This is mainly ; useful for rotatestrategy=rotate. The example allows the last ; two archive files to remain uncompressed, but after that point, ; they are compressed on disk. ; ; exec_after_rotate=gzip -9 ${filename}.2 ; ; This determines whether or not we log generic events to a file ; (defaults to yes). ;event_log = no ; ; ; For each file, specify what to log. ; ; For console logging, you set options at start of ; Asterisk with -v for verbose and -d for debug ; See 'asterisk -h' for more information. ; ; Directory for log files is configures in asterisk.conf ; option astlogdir ; [logfiles] ; ; Format is "filename" and then "levels" of debugging to be included: ; debug ; notice ; warning ; error ; verbose ; dtmf ; ; Special filename "console" represents the system console ; ; Filenames can either be relative to the standard Asterisk log directory ; (see 'astlogdir' in asterisk.conf), or absolute paths that begin with ; '/'. ; ; We highly recommend that you DO NOT turn on debug mode if you are simply ; running a production system. Debug mode turns on a LOT of extra messages, ; most of which you are unlikely to understand without an understanding of ; the underlying code. Do NOT report debug messages as code issues, unless ; you have a specific issue that you are attempting to debug. They are ; messages for just that -- debugging -- and do not rise to the level of ; something that merit your attention as an Asterisk administrator. Debug ; messages are also very verbose and can and do fill up logfiles quickly; ; this is another reason not to have debug mode on a production system unless ; you are in the process of debugging a specific issue. ; ;debug => debug console => notice,warning,error ;console => notice,warning,error,debug messages => notice,warning,error full => notice,warning,error,debug,verbose
;syslog keyword : This special keyword logs to syslog facility ; ;syslog.local0 => notice,warning,error ;
| /etc/asterisk/sip.conf[general] jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a SIP jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP jbmaxsize=300 jbimpl = adaptive ; Jitterbuffer implementation, used on the receiving side of a SIP jbresyncthreshold=100 ;adaptive jbimpl had troubles without thresh... Asterisk 1.6.2.9. ;jblog = yes ; Enables jitterbuffer frame logging. Defaults to "no".
[sip1] type=friend 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=ulaw allow=alaw ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes secret=6213495309 context=test_context
[sip2] type=friend 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=ulaw allow=alaw ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes secret=6213495309 context=test_context
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