MediaWiki API result

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        "gapcontinue": "Remote_Base:_Doug_Hall_RBI-1",
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    "warnings": {
        "main": {
            "*": "Subscribe to the mediawiki-api-announce mailing list at <https://lists.wikimedia.org/mailman/listinfo/mediawiki-api-announce> for notice of API deprecations and breaking changes."
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        "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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            "499": {
                "pageid": 499,
                "ns": 0,
                "title": "Recording Audio Files",
                "revisions": [
                    {
                        "contentformat": "text/x-wiki",
                        "contentmodel": "wikitext",
                        "*": "=Traditional Methods=\nThese [[Media:RecordingSoundFiles.pdf|Sound file recording instructions]] show how to record your own audio files for your AllStarLink node using [https://sourceforge.net/projects/audacity/ Audacity], a free program for Windows, macOS and Linux. \n\n\n=Alternate Methods=\n\nFor a more robust interface, to do the same as above within the diialplan, a sample dialplan example will be displayed here.\nIt makes use of this previously installed topic: https://wiki.allstarlink.org/wiki/Passing_DTMF_to_shell_scripts\n\nSome background.\nIn order to get a nice clean audio for your id's and tail message, you should do your recording from a quite environment, and a better radio with the best quality audio as possible.\nIt is suggested here to do this from a computer and iaxrpt: https://wiki.allstarlink.org/wiki/Setup/Download_IAXRpt or possibly a sip phone.\n\n==Install Alt Method==\n\nYou will find the need to keep your stuff organized as passing dtmf to shell script and the dialplan explodes the possibilities of the app_rpt software which limits dtmf and the use of star and pound by necessity.\n\nSo, the directory structure presented here is suggested for your initial setup and testing. Then make it what you want once you have it working.\nThis also reduces type errors. \n\n===1 Directory===\n\nYou need to create a directory to house separately your sound files. Our example here uses /etc/asterisk/msg so you will want to create that directory and set permissions for it.\n\nCreate a new directory if it does not already exist, /etc/asterisk/custom and check/set permissions for it. This is a standard directory for asterisk to add additional section of dialplan. We use it for our custom elements to keep these from exploding our dialplan without organization. So you can quickly view and edit those elements without searching the whole thing. Fewer fat finger errors without seeing them.\n\n===2 Setting file paths===\n\nedit rpt.conf to reflect the following\n\nidrecording = /etc/asterisk/msg/idmsg1\n\noptionally\nidtalkover= /etc/asterisk/msg/idmsg2   but it is advised here to keep this very short/quick or just use the cwid talkover. |N1CALL\n\ntailmessagelist= /etc/asterisk/msg/tmsg1, tmsg2, tmsg3 ...these rotate in order presented not random. So you could repeat some to play a few more than others.\n\nBe sure to set the timing interval on these if you have not run them before.\ntailmessagetime=1800000 ; 20 minutes between tail messages\ntailsquashedtime=300000 ; If squashed by another user, try again after 5 min\n\nSave your edit\n\n\n===3 The DTMF DialPlan Magic===\n\nAdd this line to your command list, and again, organization of your dialplan and commands is key here especially when you later debug.\n\n65=autopatchup,context=rsm,noct=1,farenddisconnect=1,dialtime=90000,quiet=1\n\nIt is suggested here that all these dialplan redirects stay organized in/near your phone patch commands starting with 6x and so you can see that they are non-conflicting listed together as we really are making a phone patch to ourselves in a call by 'CONTEXT', not by number. That is equated in the context when it is redirected to that context. Standard Asterisk stuff.\nIt is also here suggested that you renumber you phonepatch commands to reflect the following.\n\n*60 patch hangup / even *0 is better\n\n*61 phonepatch\n\nSo all our DTMF redirects can be hung-up with a *60 or *0 if/as needed\n\n===4 DialPlan Preparation===\n\nEdit extensions.conf and add the following line as it appears at the very bottom of the file, the last line. It is not commented out, # required.\n\n'#includeifexists custom/rsm.conf\n\nSave your edit.\n\n===5 The DialPlan===\n\nCreate the file /etc/asterisk/custom/rsm.conf and add these contents:\n\n[rsm] ;the context we are dialing into\n\nexten => 10,1,Playback(/etc/asterisk/msg/idmsg1) ; primary long id\n\nexten => 10,n,Hangup\n\n\nexten => 11,1,Playback(please-enter-your&vm-message&after-the-tone)\n\nexten => 11,n,Record(/etc/asterisk/msg/idmsg1:wav) \n\nexten => 11,n,Hangup\n\n\nexten => 20,1,Playback(/etc/asterisk/msg/idmsg2)  ; secondary quick id (provided but not suguested over cwid)\n\nexten => 20,n,Hangup\n\n\nexten => 21,1,Playback(please-enter-your&vm-message&after-the-tone)\n\nexten => 21,n,Record(/etc/asterisk/msg/idmsg2:wav) \n\nexten => 21,n,Hangup\n\n\nexten => 30,1,Playback(/etc/asterisk/msg/tmsg1)  ; tail message #1 \n\nexten => 30,n,Hangup\n\n\nexten => 31,1,Playback(please-enter-your&vm-message&after-the-tone)\n\nexten => 31,n,Record(/etc/asterisk/msg/tmsg1:wav)\n\nexten => 31,n,Hangup\n\n\nexten => 40,1,Playback(/etc/asterisk/msg/tmsg2)  ; tail message #2\n\nexten => 40,n,Hangup\n\n\nexten => 41,1,Playback(please-enter-your&vm-message&after-the-tone)\n\nexten => 41,n,Record(/etc/asterisk/msg/tmsg2:wav)\n\nexten => 41,n,Hangup\n\n\nexten => 50,1,Playback(/etc/asterisk/msg/tmsg3)  ; ; tail message #3\n\nexten => 50,n,Hangup\n\n\nexten => 51,1,Playback(please-enter-your&vm-message&after-the-tone)\n\nexten => 51,n,Record(/etc/asterisk/msg/tmsg3:wav)\n\nexten => 51,n,Hangup\n\n\nexten => 60,1,Playback(/etc/asterisk/msg/spmsg)  ; ; tail message #3\n\nexten => 60,n,Hangup\n\n\nexten => 61,1,Playback(please-enter-your&vm-message&after-the-tone)\n\nexten => 61,n,Record(/etc/asterisk/msg/spmsg:wav)\n\nexten => 61,n,Hangup\n\n\n\nexten => i,n,Playback(vm-goodbye)  ; if anything else was entered and not match above, play goodby and hangup\n\nexten => i,n,Hangup\n\n==Command List for Recording System Messages==\n\nYou will need to dial our prefix (*65) + The function we seek in 2 numbers.\n\nSo, *6510 will playback our existing Main ID\n\nand *6511 will record our new system ID\n\n\n*6530 will play our tail message 1\n\n*6531 will record our tail message 1 Overwriting the old\n\n*6540 will play our tail message 2\n\n*6541 will record our tail message 2 Overwriting the old\n\n*6550 will play our tail message 3\n\n*6551 will record our tail message 3 Overwriting the old\n\n*6560 will play our special message\n\n*6561 will record our special message Overwriting the old\n\n\nRecord all your messages and change the permissions of the files. So that owner and group users can execute.\n\n\nNow this dialplan may be revised, and likely will be. When it was originally written, the asterisk function of stopping on silence would not work correctly from app_rpt. Perhaps it was the older asterisk version we used at the time.\nThis may have changed. AT present, you will have to issue a (pound) or patch hangup command to end it. Hence some of the requested changes. Not as clean and neat as we would otherwise like it. But functionally better for those that have the need to change tail messages on the fly from where you are.\nThere is plenty of room for you to add something special of your own, if you think you understand how to work the asterisk dialplan. You are encouraged to do so. No better time to learn.\n\nSo, what is the special message ? Anything you want it to be. But you will need to call it via macro and scheduler with localplay or playback. Remember where it is located and it's name.\n\n==Summery==\n\nSo, with this tool, you can now alter it to suit your exact needs. Suggested you make it run correctly first. If you make some useful additions, please post them on the Users List \n\nhttps://community.allstarlink.org/latest?order=activity\n\n\n[[Category:How to]]"
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            "500": {
                "pageid": 500,
                "ns": 0,
                "title": "Remote Base",
                "revisions": [
                    {
                        "contentformat": "text/x-wiki",
                        "contentmodel": "wikitext",
                        "*": "[[Category:How to]]\n\nThis how-to will show you how to implement a '''remote base''' in Asterisk/app_rpt\n\nThe primary purpose of a remote base node is to make outgoing RF connections using a frequency Agile radio to allow access to frequency which are not part of your Internet connected radio system.\n\nRemote base nodes are configured differently than a standard node. A completely different set of internal functions in Asterisk/app_rpt is used when operating a node as a remote base. Usually, the only reason to set up a node as a remote base is when you wish to change the operating parameters of the the attached radio remotely, or if you only want the radio to be used by a single user at a time. The operating parameters would be frequency, emission mode, power level, and receive and transmit CTCSS tones.\n\nRemote bases which are configured to allow a remote user to command a compatible radio are known as frequency/mode agile remote bases.\n\n\n'''Security Issues'''\n\nUnfettered access to remote bases can be a security issue. If the remote base has no login protection, it could be used by unscrupulous individuals to violate amateur radio rules and regulations. We strongly advise that all remote bases be protected by requiring a login code.\nBehavior of Standard Nodes Versus Remote Base Nodes\n\n=== Behavior of Standard Nodes Versus Remote Base Nodes ===\n{| class=\"wikitable\"\n!Behavior\n!Standard None\n!Remote Base Node\n|-\n|Command Decoding\n|Remote or Local. DTMF can be optionally decoded on the receive audio input.\n|Remote only. No DTMF will be decoded on the receive audio input.\n|-\n|Duplexing\n|Configurable: duplex or half-duplex.\n|Half-duplex only\n|-\n|Frequency and Mode Agility\n|Fixed frequency operation, and channelized operation only using arguments passed in to app_rpt from\u00a0''extensions.conf''\n|Frequency and Mode agile. Support for several radio types using asynchronous serial, CAT, and synchronous serial.\n|-\n|Multiple connections\n|Multiple nodes can connect. Operates as a conference bridge.\n|Only one node can connect at a time.\n|-\n|Login Protection\n|No\n|Optional\n|}"
                    }
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