{"id":1316,"date":"2010-12-30T11:36:03","date_gmt":"2010-12-30T11:36:03","guid":{"rendered":"http:\/\/mccltd.net\/blog\/?p=1316"},"modified":"2012-01-02T02:47:32","modified_gmt":"2012-01-02T02:47:32","slug":"asterisk-cli-commands","status":"publish","type":"post","link":"http:\/\/darenmatthews.com\/blog\/?p=1316","title":{"rendered":"Asterisk CLI Commands"},"content":{"rendered":"<p>Another aide-memoir reproduced from voip-info.org:\u00a0 <strong>Asterisk<\/strong> provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, IAX, SIP, H.323 (as both client and gateway), MGCP (call manager only) and SCCP\/Skinny. SEE ALSO:\u00a0 <strong><a title=\"The Asterisk Book\" href=\"http:\/\/www.the-asterisk-book.com\/unstable\/\" target=\"_blank\">The Asterisk Book<\/a><\/strong><\/p>\n<p><!--more--><\/p>\n<p>The command line interface (CLI) is reached by  using the Linux shell command<br \/>\nasterisk -r<\/p>\n<p>If you want debugging output, add one or many <strong>v<\/strong>:s<br \/>\nasterisk -vvvvvr<\/p>\n<p>The Asterisk server has to be running in the background for the CLI to start.<\/p>\n<p>If you want to run a CLI command in a shell script, use the <strong>x<\/strong> option<\/p>\n<p>asterisk -rx &#8220;logger reload&#8221;<\/p>\n<p>For help in the CLI mode, use the <strong>help<\/strong> command. To get help on various applications you can use in the <a title=\"Asterisk config extensions.conf\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/Asterisk+config+extensions.conf\">extensions.conf<\/a> config file, use the <strong>show applications<\/strong> command.<\/p>\n<h2 id=\"Generalcommands\">General commands<\/h2>\n<ul>\n<li> <strong>!&lt;command&gt;<\/strong>:   Executes a given shell command<\/li>\n<li> <strong>abort halt<\/strong>:   Cancel a running halt<\/li>\n<li> <strong>add extension<\/strong>:   Add new extension into context<\/li>\n<li> <strong>add ignorepat<\/strong>:   Add new ignore pattern<\/li>\n<li> <strong>add indication<\/strong>:   Add the given indication to the country<\/li>\n<li> <strong>debug channel<\/strong>:   Enable debugging on a channel<\/li>\n<li> <strong>dont include<\/strong>:   Remove a specified include from context<\/li>\n<li> <strong>help<\/strong>:   Display help list, or specific help on a command<\/li>\n<li> <strong>include context<\/strong>:   Include context in other context<\/li>\n<li> <strong>load<\/strong>:   Load a dynamic module by name<\/li>\n<li> <strong>logger reload<\/strong>:   Reopen log files. Use after rotating the log files.<\/li>\n<li> <strong>no debug channel<\/strong>:   Disable debugging on a channel<\/li>\n<li> <strong><a title=\"Asterisk cli originate\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/Asterisk+cli+originate\">originate<\/a><\/strong>: originate a call.<\/li>\n<li> <strong>remove extension<\/strong>:   Remove a specified extension<\/li>\n<li> <strong>remove ignorepat<\/strong>:   Remove ignore pattern from context<\/li>\n<li> <strong>remove indication<\/strong>:   Remove the given indication from the country<\/li>\n<li> <strong>save dialplan<\/strong>: <em>Overwrites<\/em> your current extensions.conf file with an exported version based on the  current state of the dialplan. A backup copy of your old  extensions.conf is <em>not<\/em> saved. The initial values of global  variables defined in the [globals] category retain their previous  initial values; the current values of global variables are not written  into the new extensions.conf. (:exclaim:) Using &#8220;save dialplan&#8221; will  result in losing any comments in your current extensions.conf.<\/li>\n<li> <strong>dialplan save (1.4)<\/strong>: <span style=\"background:yellow;\">BROKEN, doesn&#8217;t parse correctly.<\/span> <em>Overwrites<\/em> your current extensions.conf file with an exported version based on the  current state of the dialplan. A backup copy of your old  extensions.conf is <em>not<\/em> saved. The initial values of global  variables defined in the [globals] category retain their previous  initial values; the current values of global variables are not written  into the new extensions.conf. (:exclaim:) Using &#8220;save dialplan&#8221; will  result in losing any comments in your current extensions.conf.<\/li>\n<li> <strong>set verbose<\/strong>:   Set level of verboseness<\/li>\n<li> <strong>show agents<\/strong>:   Show status of <a title=\"Asterisk Agents\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/Asterisk+Agents\">agents<\/a><\/li>\n<li> <strong>show applications<\/strong>:   <a title=\"Asterisk - documentation of application commands\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/Asterisk+-+documentation+of+application+commands\">Shows registered applications<\/a><\/li>\n<li> <strong>show application<\/strong>:   Describe a specific application<\/li>\n<li> <strong>show channel<\/strong>:   Display information on a specific channel<\/li>\n<li> <strong><a title=\"show channels\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/show+channels\">show channels<\/a><\/strong>:   Display information on channels<\/li>\n<li> <strong><a title=\"Asterisk cli show codecs\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/Asterisk+cli+show+codecs\">show codecs<\/a><\/strong>:   Display information on codecs<\/li>\n<li> <strong>show conferences<\/strong>:   Show status of conferences<\/li>\n<li> <strong>show dialplan<\/strong>:   Show dialplan<\/li>\n<li> <strong><a title=\"Asterisk cli show hints\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/Asterisk+cli+show+hints\">show hints<\/a><\/strong>:   Show registered hints<\/li>\n<li> <strong>show image formats<\/strong>:   Displays image formats<\/li>\n<li> <strong>show indications<\/strong>:   Show a list of all <a title=\"Asterisk config indications.conf\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/Asterisk+config+indications.conf\">country\/indications<\/a><\/li>\n<li> <strong>show locals<\/strong>:   Show status of local channels<\/li>\n<li> <strong>show manager command<\/strong>:   Show manager commands<\/li>\n<li> <strong>show manager connect<\/strong>:   Show connected manager users<\/li>\n<li> <strong>show parkedcalls<\/strong>:   Lists parked calls<\/li>\n<li> <strong>show queues<\/strong>:   Show status of queues, see details <a title=\"asterisk cli command show queue\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/asterisk+cli+command+show+queue\">here<\/a><\/li>\n<li> <strong>show switches<\/strong>:   Show alternative switches<\/li>\n<li> <strong>show translation<\/strong>:   Display translation matrix<\/li>\n<li> <strong>soft hangup<\/strong>:   Request a hangup on a given channel &#8211; in Asterisk 1.6.2: &#8220;channel request hangup &lt;name&gt;&#8221;<\/li>\n<li> <strong>show voicemail users<\/strong>:  List defined voicemail boxes<\/li>\n<li> <strong>show voicemail zones<\/strong>:  List zone message formats<\/li>\n<li> <strong><a title=\"Asterisk cli devstate change\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/Asterisk+cli+devstate+change\">devstate change<\/a><\/strong>: Change state of a custom device (new in Asterisk 1.6.0)<\/li>\n<\/ul>\n<h2 id=\"Servermanagement\">Server management<\/h2>\n<ul>\n<li> <strong>restart gracefully<\/strong>:   Restart Asterisk gracefully, i.e. stop receiving new calls and restart at empty call volume<\/li>\n<li> <strong>restart now<\/strong>:   Restart Asterisk immediately<\/li>\n<li> <strong>restart when convenient<\/strong>:   Restart Asterisk at empty call volume<\/li>\n<\/ul>\n<p><strong>Note for Asterisk 1.2:<\/strong> Restart now is like a reload, not a real restart it just run the reload  routines (thus open ports are not closed). Often you don&#8217;t need really  need to restart asterisk, instead just need to issue e.g. &#8216;unload  chan_sip.so&#8217; and &#8216;load chan_sip.so&#8217;.<\/p>\n<ul>\n<li> <strong>reload<\/strong>:   Reload configuration<\/li>\n<li> <strong>stop gracefully<\/strong>:   Gracefully shut down Asterisk, i.e. stop receiving new calls and shut down at empty call volume<\/li>\n<li> <strong>stop now<\/strong>:   Shut down Asterisk imediately<\/li>\n<li> <strong>stop when convenient<\/strong>:   Shut down Asterisk at empty call volume<\/li>\n<li> <strong>dialplan reload<\/strong>:   Reload extensions and <em>only<\/em> extensions (formerly extensions reload)<\/li>\n<li> <strong>unload<\/strong>:   Unload a dynamic module by name<\/li>\n<li> <strong>show modules<\/strong>:   List <a title=\"Asterisk modules\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/Asterisk+modules\">modules<\/a> and info about them<\/li>\n<li> <strong>show uptime<\/strong>:   Show uptime information<\/li>\n<li> <strong>show version<\/strong>:   Display Asterisk version info<\/li>\n<\/ul>\n<h2 id=\"AGIcommands\">AGI commands<\/h2>\n<ul>\n<li> <strong>show agi<\/strong>:   Show AGI commands or specific help<\/li>\n<li> <strong>dump agihtml<\/strong>:   Dumps a list of agi command in html format<\/li>\n<\/ul>\n<h2 id=\"Databasehandlingcommands\">Database handling commands<\/h2>\n<ul>\n<li> <strong>database del<\/strong>:   Removes database key\/value<\/li>\n<li> <strong>database deltree<\/strong>:   Removes database keytree\/values<\/li>\n<li> <strong>database get<\/strong>:   Gets database value<\/li>\n<li> <strong><a title=\"Asterisk cli database put\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/Asterisk+cli+database+put\">database put<\/a><\/strong>:   Adds\/updates database value<\/li>\n<li> <strong>database show<\/strong>:   Shows database contents<\/li>\n<li> <strong>database showkey<\/strong>:   Shows database contents: An alternative to showing keys by family with <em>database show<\/em>, this command shows <strong>all<\/strong> the families with a particular key<\/li>\n<\/ul>\n<h2 id=\"IAXChannelcommands\">IAX Channel commands<\/h2>\n<ul>\n<li> <strong>iax2 debug<\/strong>:   Enable IAX debugging<\/li>\n<li> <strong>iax2 no debug<\/strong>:   Disable IAX debugging<\/li>\n<li> <strong>iax2 set jitter<\/strong>:   Sets IAX jitter buffer<\/li>\n<li> <strong>iax2 show cache<\/strong>:   Display IAX cached dialplan<\/li>\n<li> <strong>iax2 show channels<\/strong>:   Show active IAX channels<\/li>\n<li> <strong><a title=\"Asterisk cli iax2 show netstats\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/Asterisk+cli+iax2+show+netstats\">iax2 show netstats<\/a><\/strong>: Show network and jitter buffer statistics for active IAX calls<\/li>\n<li> <strong>iax2 show peers<\/strong>:   Show defined IAX peers<\/li>\n<li> <strong>iax2 show registry<\/strong>:   Show IAX registration status<\/li>\n<li> <strong>iax2 show stats<\/strong>:   Display IAX statistics<\/li>\n<li> <strong>iax2 show users<\/strong>:   Show defined IAX users<\/li>\n<li> <strong>iax2 trunk debug<\/strong>:   Request IAX trunk debug<\/li>\n<\/ul>\n<ul>\n<li> <strong>iax debug<\/strong>:   Enable IAX debugging<\/li>\n<li> <strong>iax no debug<\/strong>:   Disable IAX debugging<\/li>\n<li> <strong>iax set jitter<\/strong>:   Sets IAX jitter buffer<\/li>\n<li> <strong>iax show cache<\/strong>:   Display IAX cached dialplan<\/li>\n<li> <strong>iax show channels<\/strong>:   Show active IAX channels<\/li>\n<li> <strong>iax show peers<\/strong>:   Show defined IAX peers<\/li>\n<li> <strong>iax show registry<\/strong>:   Show IAX registration status<\/li>\n<li> <strong>iax show stats<\/strong>:   Display IAX statistics<\/li>\n<li> <strong>iax show users<\/strong>:   Show defined IAX users<\/li>\n<li> <strong>init keys<\/strong>:   Initialize RSA key passcodes<\/li>\n<li> <strong>show keys<\/strong>:   Displays RSA key information<\/li>\n<\/ul>\n<h2 id=\"H323channelcommands\">H323 channel commands<\/h2>\n<ul>\n<li> <strong>h.323 debug<\/strong>:   Enable chan_h323 debug<\/li>\n<li> <strong>h.323 gk cycle<\/strong>:   Manually re-register with the Gatekeper<\/li>\n<li> <strong>h.323 hangup<\/strong>:   Manually try to hang up a call<\/li>\n<li> <strong>h.323 no debug<\/strong>:   Disable chan_h323 debug<\/li>\n<li> <strong>h.323 no trace<\/strong>:   Disable H.323 Stack Tracing<\/li>\n<li> <strong>h.323 show codecs<\/strong>:   Show enabled codecs<\/li>\n<li> <strong>h.323 show tokens<\/strong>:   Manually try to hang up a call<\/li>\n<li> <strong>h.323 trace<\/strong>:   Enable H.323 Stack Tracing<\/li>\n<\/ul>\n<h2 id=\"SIPchannelcommands\">SIP channel commands<\/h2>\n<ul>\n<li>Debugging\n<ul>\n<li>Enable\n<ul>\n<li><strong>sip debug<\/strong><\/li>\n<li><strong>sip set debug on<\/strong> (valid on 1.6.2.7)<\/li>\n<\/ul>\n<\/li>\n<li>Disable\n<ul>\n<li> <strong>sip no debug<\/strong><\/li>\n<li> <strong>sip set debug off<\/strong> (valid on 1.6.2.7)<\/li>\n<\/ul>\n<\/li>\n<\/ul>\n<\/li>\n<li> <strong>sip reload<\/strong>: Reload sip.conf (added after 0.7.1 on 2004-01-23)<\/li>\n<li> <strong>sip show channels<\/strong>:   Show active SIP channels<\/li>\n<li> <strong>sip show channel<\/strong>:   Show detailed SIP channel info<\/li>\n<li> <strong><a title=\"asterisk cli command sip show inuse\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/asterisk+cli+command+sip+show+inuse\">sip show inuse<\/a><\/strong>:   List all inuse\/limit<\/li>\n<li> <strong>sip show peers<\/strong>:   Show defined SIP peers (clients that register to your Asterisk server),  see details <a title=\"asterisk cli command sip show peers\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/asterisk+cli+command+sip+show+peers\">here<\/a><\/li>\n<li> <strong>sip show registry<\/strong>:   Show SIP registration status (when Asterisk registers as a client to a SIP Proxy)<\/li>\n<li> <strong>sip show subscriptions<\/strong>: Lists all sip presence (busy lamp indication) subscriptions<\/li>\n<li> <strong>sip show users<\/strong>:   Show defined SIP users<\/li>\n<\/ul>\n<h2 id=\"Zapchannelcommands\">Zap channel commands<\/h2>\n<ul>\n<li> <strong>zap destroy channel<\/strong>:   Destroy a channel<\/li>\n<li> <strong>zap show channels<\/strong>:   Show active zapata channels<\/li>\n<li> <strong>zap show channel<\/strong>:   Show information on a channel<\/li>\n<li> <strong>zap show status<\/strong>: lists all the Zaptel spans. A span will apear here whether or not its channels are configured with chan_zap.<\/li>\n<li> <strong>zap show cadences<\/strong>: Show the configured ring cadences (available e.g with Zap\/1r2).<\/li>\n<li> <strong>zap set swgain<\/strong>(&lt;= 1.6): set the (software) gain for a hannel. Temporary equivalents of rxgain and txgain in zapata.conf.<\/li>\n<li> <strong>zap set hwgain<\/strong>(&lt;=1.6): set the hardware gain for channels that support it.<\/li>\n<li> <strong>zap set dnd<\/strong>(&lt;=1.6) set a channel&#8217;s do-not-disturb mode on or off.<\/li>\n<\/ul>\n<p>The following commands are available if the channel is built with support for libpri:<\/p>\n<ul>\n<li> <strong>pri debug span<\/strong>:   Enables PRI debugging on a span<\/li>\n<li> <strong>pri intense debug span<\/strong>:   Enables REALLY INTENSE PRI debugging<\/li>\n<li> <strong>pri no debug span<\/strong>:   Disables PRI debugging on a span<\/li>\n<li> <strong>pri show spans<\/strong>: List spans and their status.<\/li>\n<li> <strong>pri show span<\/strong>: Information about a span.<\/li>\n<li> <strong>pri show debug<\/strong>: show where debug is enabled.<\/li>\n<\/ul>\n<p>See section 3 of <a rel=\"nofollow\" href=\"http:\/\/www.att.com\/cpetesting\/pdf\/tr41459_99.pdf\" target=\"_blank\">AT&amp;T tr41459_99<\/a> to better understand what the output generated by the pri debug command means.<\/p>\n<p><a title=\"Bristuff\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/Bristuff\">Bristuff<\/a> adds <strong>bri debug<\/strong> which is an alias for pri debug.<\/p>\n<p>TODO: SS7 support in 1.6.<\/p>\n<h2 id=\"Consolechannelcommands\"><a title=\"Asterisk console channels\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/Asterisk+console+channels\">Console channel<\/a> commands<\/h2>\n<ul>\n<li> <strong><a title=\"Asterisk CLI dial\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/Asterisk+CLI+dial\">dial<\/a><\/strong> : Dials the given extension, if specified, from the console.  Can be  used to initiate a call, or to dial digits during an existing call.<\/li>\n<li> <strong>answer<\/strong>: Answer a call if one is currently ringing on the console.<\/li>\n<li> <strong>hangup<\/strong>: Hangup the call if there is currently one on the console.<\/li>\n<\/ul>\n<h2 id=\"AsteriskchannelMGCPcommands\">Asterisk channel MGCP commands<\/h2>\n<ul>\n<li> <strong>mgcp audit endpoint<\/strong>:   Audit specified MGCP endpoint<\/li>\n<li> <strong>mgcp debug<\/strong>:   Enable MGCP debugging<\/li>\n<li> <strong>mgcp no debug<\/strong>:   Disable MGCP debugging<\/li>\n<li> <strong>mgcp show endpoints<\/strong>:   Show defined MGCP endpoints<\/li>\n<\/ul>\n<h2 id=\"skinnychannelcommands\"><a title=\"Asterisk channel skinny\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/Asterisk+channel+skinny\">skinny channel<\/a> commands<\/h2>\n<ul>\n<li> <strong>skinny debug<\/strong>:   Enable Skinny debugging<\/li>\n<li> <strong> skinny no debug<\/strong>:   Disable Skinny debugging<\/li>\n<li> <strong> skinny show lines<\/strong>:   Show defined Skinny lines per device<\/li>\n<\/ul>\n<h2 id=\"AsteriskchannelCAPIcommands\">Asterisk channel CAPI commands<\/h2>\n<ul>\n<li> <strong>capi debug<\/strong>:   Enable CAPI debugging<\/li>\n<li> <strong>capi no debug<\/strong>:   Disable CAPI debugging<\/li>\n<li> <strong>capi info<\/strong>:   Show CAPI info<\/li>\n<\/ul>\n<h2 id=\"SirrixISDNchannelcommands\"><a title=\"Asterisk Sirrix ISDN channels\" href=\"http:\/\/www.voip-info.org\/wiki\/view\/Asterisk+Sirrix+ISDN+channels\">Sirrix ISDN channel<\/a> commands<\/h2>\n<ul>\n<li> <strong>srx reload<\/strong>: Reload channel driver configuration; active calls are not being terminated!<\/li>\n<li> <strong>srx show ccmsgs<\/strong>: Disable \/ enable output of incoming callcontrol messages.<\/li>\n<li> <strong>srx show chans<\/strong>: Show info about B-Channels<\/li>\n<li> <strong>srx show globals<\/strong>: Show info about global settings<\/li>\n<li> <strong>srx show groups<\/strong>: Show info about configured groups<\/li>\n<li> <strong>srx show layers<\/strong>: Show info about ISDN stack (Layer 1, 2, 3)<\/li>\n<li> <strong>srx show sxpvts<\/strong>: Show private info about active channels<\/li>\n<li> <strong>srx show timers<\/strong>: Show info about running timers<\/li>\n<\/ul>\n<h2 id=\"BatchfileswithCLI\">Batch files with CLI<\/h2>\n<div>If you meant &#8220;can Asterisk read a series of commands from a file&#8221; the<br \/>\nanswer is no, but something like the following may do:<\/p>\n<p>cat batch-file\\<br \/>\n| awk &#8216;{printf &#8220;\/usr\/sbin\/asterisk -r -x \\&#8221;%s\\&#8221;\\n&#8221;, $0}&#8217;\\<br \/>\n| sh<\/p><\/div>\n<p>The above is very slow, though. A faster option is to use socat and write the commands directly to the Asterisk socket.<\/p>\n<ol>\n<li>!\/bin\/sh<\/li>\n<\/ol>\n<p>while read line<br \/>\ndo<br \/>\necho -n &#8220;$line&#8221;<br \/>\nsleep 0.001<br \/>\ndone \\<br \/>\n| socat STDIN UNIX-CONNECT:\/var\/run\/asterisk\/asterisk.ctl<\/p>\n<p>The  short sleep is only needed to guarantee that every line is written in a  separate write() call. It will not print any output from any command,  though, or even report an error. And you&#8217;ll have to end your &#8220;programs&#8221;  with a &#8220;quit&#8221; line.<\/p>\n","protected":false},"excerpt":{"rendered":"<p>Another aide-memoir reproduced from voip-info.org:\u00a0 Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, IAX, SIP, H.323 (as both client and gateway), MGCP (call manager only) and SCCP\/Skinny. SEE ALSO:\u00a0 The Asterisk Book<\/p>\n","protected":false},"author":1,"featured_media":0,"comment_status":"closed","ping_status":"open","sticky":false,"template":"","format":"standard","meta":[],"categories":[41],"tags":[57,42],"_links":{"self":[{"href":"http:\/\/darenmatthews.com\/blog\/index.php?rest_route=\/wp\/v2\/posts\/1316"}],"collection":[{"href":"http:\/\/darenmatthews.com\/blog\/index.php?rest_route=\/wp\/v2\/posts"}],"about":[{"href":"http:\/\/darenmatthews.com\/blog\/index.php?rest_route=\/wp\/v2\/types\/post"}],"author":[{"embeddable":true,"href":"http:\/\/darenmatthews.com\/blog\/index.php?rest_route=\/wp\/v2\/users\/1"}],"replies":[{"embeddable":true,"href":"http:\/\/darenmatthews.com\/blog\/index.php?rest_route=%2Fwp%2Fv2%2Fcomments&post=1316"}],"version-history":[{"count":5,"href":"http:\/\/darenmatthews.com\/blog\/index.php?rest_route=\/wp\/v2\/posts\/1316\/revisions"}],"predecessor-version":[{"id":1390,"href":"http:\/\/darenmatthews.com\/blog\/index.php?rest_route=\/wp\/v2\/posts\/1316\/revisions\/1390"}],"wp:attachment":[{"href":"http:\/\/darenmatthews.com\/blog\/index.php?rest_route=%2Fwp%2Fv2%2Fmedia&parent=1316"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"http:\/\/darenmatthews.com\/blog\/index.php?rest_route=%2Fwp%2Fv2%2Fcategories&post=1316"},{"taxonomy":"post_tag","embeddable":true,"href":"http:\/\/darenmatthews.com\/blog\/index.php?rest_route=%2Fwp%2Fv2%2Ftags&post=1316"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}