Результат MediaWiki API

{
    "batchcomplete": "",
    "continue": {
        "gapcontinue": "Registration",
        "continue": "gapcontinue||"
    },
    "warnings": {
        "main": {
            "*": "Subscribe to the mediawiki-api-announce mailing list at <https://lists.wikimedia.org/postorius/lists/mediawiki-api-announce.lists.wikimedia.org/> for notice of API deprecations and breaking changes."
        },
        "revisions": {
            "*": "Because \"rvslots\" was not specified, a legacy format has been used for the output. This format is deprecated, and in the future the new format will always be used."
        }
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    "query": {
        "pages": {
            "2748": {
                "pageid": 2748,
                "ns": 0,
                "title": "Recommendations for configuring the Asterisk system",
                "revisions": [
                    {
                        "contentformat": "text/x-wiki",
                        "contentmodel": "wikitext",
                        "*": "[[Recommendations_for_configuring_the_Asterisk_system|en]] | [[\u0420\u0435\u043a\u043e\u043c\u0435\u043d\u0434\u0430\u0446\u0438\u0438_\u043f\u043e_\u043d\u0430\u0441\u0442\u0440\u043e\u0439\u043a\u0435_\u0441\u0438\u0441\u0442\u0435\u043c\u044b_Asterisk|ru]]\n\n'''Basic Asterisk settings that the [[Usm_asterisk_EN|usm_asterisk]] module is designed for'''\n\nThis article provides a typical basic contact center configuration, which allows you to receive incoming calls from a SIP line and transfer them to a queue to which users are connected.\n\nFollow the instructions for installing the telephone system: http://blog.denisbondar.com/post/asterisk11-chan_dongle_e1550-ubuntu14\n\n== Input data for the demo configuration ==\n\n* one external SIP line [SIPOPOPERATOR], from the side of which incoming phone calls are received. Phone number 0551234567, password qwerty123456, user server IP address 11.22.33.44\n* one serial search group (call queue)\n* four internal SIP lines on which contact centre users work - all of them are included in one queue\n\n== sip.conf ==\n\nContents of the file responsible for SIP lines\n\n ; general settings\n [general]\n language = ru\n bindport=5060\n callcounter = yes\n limitonpeers = yes\n dtmf=rfc2833\n compensate=yes\n defaultexpirey=3600\n disallow=all\n allow=alaw\n registertimeout=3600\n ; here you need to list all external sip lines in the specified format\n register=0551234567:qwerty123456@11.22.33.44/0551234567\n \n ; next, the settings for the external user that provides the phone number\n [SIPOPERATOR]\n type=friend\n host=11.22.33.44\n port=5060\n defaultip=here_ip_address_on_interface_to_the_user_side\n fromdomain=here_domain_name_of_the_user_softswitch_name\n fromuser=0551234567\n username=0551234567\n secret=qwerty123456\n registertimeout=3600\n defaultexpirey=3600\n nat=no\n canreinvite=no\n disallow=all\n allow=alaw\n qualify=yes\n dtmfmode=inband\n insecure=port,invite\n context=from-external-sip ; this context will be used in routing when calling from an external sip line\n \n ; here are the general settings for all internal sip lines\n [CallCenter](!)\n call-limit=1 ; Limit simultaneous calls (users receive only one call)\n qualify = no\n dtmfmode=rfc2833\n canreinvite=no\n pickupgroup=1\n callgroup=1\n host=dynamic\n type=friend\n port=5060\n qualify=yes\n deny=0.0.0.0/0.0.0.0\n permit=0.0.0.0.0/0.0.0.0.0 ; allowed to connect from anywhere. Change if needed otherwise\n callcounter=yes\n faxdetect=no\n disallow=all\n allow=alaw\n allow=gsm\n context=from-internal-sip ; this context will be used in routing for calls from contact centre user lines\n \n The 4 internal telephone lines with common settings are described below\n [500](CallCenter)\n username=500\n nat=no\n secret=here_password\n callerid=user 500 <500>\n \n [501](CallCenter)\n username=501\n nat=no\n secret=here_password\n callerid=user 501 <501>\n \n [502](CallCenter)\n username=502\n nat=no\n secret=here_password\n callerid=user 502 <502>\n \n [503](CallCenter)\n username=503\n nat=no\n secret=here_password\n callerid=user 503 <503>\n\n== queues.conf ==\n\nThe group in the configuration file with the settings of serial search groups (queues) is discussed below\n\n [callcenter-queue]\n monitor-type = MixMonitor\n music = file_name ; name of the file without extension, which will be played to the user in the user's answer waiting mode.\n strategy = ringall ; serial search strategy - call all users at once\n timeout = 30\n retry = 2\n joinempty = yes ; a call can join the queue even if no user is connected to it\n ringinuse = no ; do not accept calls to people who are already on a call\n leavewhenempty = yes\n announce-frequency = 30\n announce-holdtime = no\n ; the members of the group are then listed\n member => SIP/500\n member => SIP/501\n member => SIP/502\n member => SIP/503\n\n== extensions.conf ==\n\nThe following file configures routing. A fragment that routes incoming calls to the group and outgoing calls through the external SIP line is considered\n\n ; incoming calls are those that come from the external sip line towards the contact centre\n [from-external-sip]\n exten => s,1,Ringing() ; if required, send a ringing signal to the caller (if not required, delete the line).\n exten => s,n,Wait(2) ; wait two seconds for the caller to hear the first beep after dialling the number, if you answer immediately without beeping - the caller gets scared :)\n exten => s,n,Answer(300) ; answer the call\n exten => s,n,Playback(hello) ; play the hello file (file name without extension).\n exten => s,n,Queue(callcenter-queue) ; direct the call to the \"callcenter-queue\" queue \n exten => s,n,Hangup() ; direct the caller to the hangup queue\n \n ; outgoing calls - those that originate from the contact centre users to the PSTN\n [from-internal-sip]\n exten => _X.,1,Set(CALLERID(num)=380551234567) ; Change the Caller_ID before sending the call to the PSTN to the outside line number\n exten => _X.,n,Dial(SIP/SIPOPOPERATOR/${EXTEN},300) ; Direct the call towards SIPOPERATOR\n\n'''Do not make any other settings if you do not know what they are for.'''\n\nIf you configure the system through various web-interfaces (trixbox, elastix, freepbx, etc.), then it is impossible to guarantee the work of usm_asterisk module, because the logic of Asterisk operation after such configuration is often unpredictable and it is impossible to trace the call passing through the standard module."
                    }
                ]
            },
            "3477": {
                "pageid": 3477,
                "ns": 0,
                "title": "Register of works",
                "revisions": [
                    {
                        "contentformat": "text/x-wiki",
                        "contentmodel": "wikitext",
                        "*": "[[Register_of_works|en]] | [[\u0412\u0435\u0434\u043e\u043c\u043e\u0441\u0442\u0438_\u0440\u0430\u0431\u043e\u0442|ru]]\n\nRegister of works is a somewhat collected term. This interface allows you to get information about completed [[Work_schedule|works]] both for individual employees and for the whole company. At the same time, the total cost of the work is calculated, taking into account [[Settings - Tasks|billing rates]].\n\n[[File:2024-01-29 19-54.png|thumb|800px|center]]\n\nThe display filter allows you to set the required date range, or customise the selection for a specific employee.\n\n[[File:2024-01-29 19-54 1.png|thumb|800px|center]]\n\nFor a more detailed display, you need to add in the \"In detail\" filter.\n\n[[File:2024-01-29 19-56.png|thumb|800px|center]]"
                    }
                ]
            }
        }
    }
}