{"id":8580,"date":"2020-07-08T00:00:09","date_gmt":"2020-07-07T21:00:09","guid":{"rendered":"https:\/\/trueconf.com\/blog\/?p=8580"},"modified":"2022-06-06T23:25:47","modified_gmt":"2022-06-06T20:25:47","slug":"join-conferences-call-trueconf-users-regular-phone","status":"publish","type":"post","link":"https:\/\/trueconf.com/blog\/knowledge-base\/join-conferences-call-trueconf-users-regular-phone","title":{"rendered":"How to Join Conferences and Call TrueConf Users from a Regular Phone?"},"content":{"rendered":"<p style=\"text-align: justify;\"><span style=\"font-weight: 400;\">Thanks to <\/span><a href=\"https:\/\/trueconf.com\/features\/integration\/multigateway.html\" target=\"_blank\" rel=\"noopener\"><span style=\"font-weight: 400;\">integrated gateway<\/span><\/a><span style=\"font-weight: 400;\">, TrueConf Server can receive external calls over the most popular SIP and H.323 telephony protocols, which allows regular telephone subscribers to call users and conferences on TrueConf Server. The only requirement is an installed corporate PBX that would redirect calls from your telephone network to TrueConf Server.<\/span><\/p>\n<p style=\"text-align: justify;\"><span style=\"font-weight: 400;\">In this article, we\u2019re going to run through an example of such an integration based on Asterisk PBX.<\/span><!--more--><\/p>\n<h2>Create Special Prefix for Video Conferencing Sessions in Your PBX<\/h2>\n<p style=\"text-align: justify;\">Create a new SIP rule in your PBX to intercept special format calls from your telephone subscribers, which will redirect calls to another PBX. TrueConf Server will play as this PBX. Note that an algorithm for setting up a new SIP rule depends on your PBX.<\/p>\n<p style=\"text-align: justify;\"><span style=\"font-weight: 400;\">For example, in\u00a0<\/span><span style=\"font-weight: 400;\">Asterisk<\/span><span style=\"font-weight: 400;\"> PBX, just add the following string to a configuration file:<\/span><\/p>\n<p style=\"text-align: justify;\"><code>exten =&gt; _06X.,1,Dial(SIP\/${EXTEN:2}\\@192.168.1.100,360)<\/code><\/p>\n<p style=\"text-align: justify;\"><span style=\"font-weight: 400;\">Let\u2019s take a closer look at this string:<\/span><\/p>\n<p style=\"text-align: justify;\"><code>exten =&gt;<\/code>\u00a0<span style=\"font-weight: 400;\">creates a new rule for redirecting calls<\/span>;<\/p>\n<p style=\"text-align: justify;\"><code>_06X.<\/code>\u00a0<span style=\"font-weight: 400;\">is a special kind of regular expression determining which numbers will be processed by this rule. For example, <code>_ <\/code>means the beginning of a string, and <code>X.<\/code> is any sequence of random length symbols\u2014this expression redirects any calls starting with <code>06<\/code><\/span>;<\/p>\n<p style=\"text-align: justify;\"><code>1<\/code>\u00a0<span style=\"font-weight: 400;\">is a rule number in your PBX<\/span>;<\/p>\n<p style=\"text-align: justify;\"><code>Dial(SIP\/${EXTEN:2}\\@192.168.1.100,360)<\/code>\u00a0<span style=\"font-weight: 400;\">is a call command that will be applied to all numbers that satisfy this rule. It has two parameters: <\/span><code>SIP\/${EXTEN:2}\\@192.168.1.100<\/code><span style=\"font-weight: 400;\"> call string and time in seconds (<\/span><code>360<\/code><span style=\"font-weight: 400;\">) for a device to reach an external server. Where:<\/span><\/p>\n<ul style=\"text-align: justify;\">\n<li><code>SIP\/<\/code> <span style=\"font-weight: 400;\">is a call protocol<\/span>;<\/li>\n<li><code>${EXTEN:2}<\/code>\u00a0<span style=\"font-weight: 400;\">will be automatically substituted for a part of a call string starting with the third symbol (this expression contains 2, since we count from zero);<\/span><\/li>\n<li><code>\\@192.168.101<\/code> <span style=\"font-weight: 400;\">is an address of TrueConf Server gateway to which you want to forward a call. In your case, a server address may be different<\/span>.<\/li>\n<\/ul>\n<p style=\"text-align: justify;\"><span style=\"font-weight: 400;\">For example, your PBX redirects the call <\/span><code>06101<\/code><span style=\"font-weight: 400;\"> to the user <\/span><code>101<\/code><span style=\"font-weight: 400;\"> of the external server <\/span><code>192.168.1.100<\/code><span style=\"font-weight: 400;\"> by calling the URI <\/span><code>101@192.168.1.100<\/code><span style=\"font-weight: 400;\"> over the SIP protocol. In fact, we\u2019ve created a special prefix with all the calls forwarded to a video conferencing server.<\/span><\/p>\n<h2>How to Call a TrueConf User from a Phone?<\/h2>\n<p style=\"text-align: justify;\"><span style=\"font-weight: 400;\">By using our rule from the previous section, we can now call any TrueConf Server users from a phone to <\/span><code>06&lt;username&gt;<\/code> <span style=\"font-weight: 400;\">numbers. Here we\u2019re facing a problem as users usually have alphabetic account names, and phones have no keyboard to type them (T9 does not count.) Therefore, you need to add user digital pseudonyms on TrueConf Server.<\/span><\/p>\n<p style=\"text-align: justify;\"><span style=\"font-weight: 400;\">In TrueConf Server control panel, go to <\/span><a href=\"https:\/\/docs.trueconf.com\/server\/en\/admin\/web-config\/#aliases\" target=\"_blank\" rel=\"noopener\"><strong>Users \u2192 Aliases<\/strong><\/a>\u00a0<span style=\"font-weight: 400;\">section and add digital IDs for users who need to be called via telephony. After that, all the calls to a number entered in the <strong>Aliases\u00a0<\/strong><\/span><span style=\"font-weight: 400;\"><strong>\u200b\u200b<\/strong>field, including calls from SIP and H.323 devices, will be redirected to a corresponding user.<\/span><\/p>\n<p style=\"text-align: justify;\"><span style=\"font-weight: 400;\">Congratulations! You can now call any TrueConf subscribers directly from your phone by dialing the prefix <\/span><code>06<\/code><span style=\"font-weight: 400;\"> before their pseudonym or name.<\/span><\/p>\n<h2>How to Join a TrueConf Conference from a Phone?<\/h2>\n<p style=\"text-align: justify;\"><span style=\"font-weight: 400;\">To join a conference, regular applications use <\/span><code>\\c\\CID<\/code><span style=\"font-weight: 400;\"> string, where <\/span><code>CID<\/code><span style=\"font-weight: 400;\"> is a digital conference ID (for example, <\/span><code>\\c\\4154248070<\/code><span style=\"font-weight: 400;\">). To make calls from devices without keyboards more comfortable, conference prefix <\/span><code>\\c\\<\/code><span style=\"font-weight: 400;\"> has an alternative option as two zeros <\/span><code>00<\/code><span style=\"font-weight: 400;\">.<\/span><\/p>\n<p style=\"text-align: justify;\"><span style=\"font-weight: 400;\">For example, if you dial <\/span><code>06008001<\/code><span style=\"font-weight: 400;\"> from a phone connected to a PBX with the above rule set, conference <\/span><code>\\c\\8001<\/code><span style=\"font-weight: 400;\"> will be called on TrueConf Server. A server gateway will automatically connect a telephone subscriber to a conference.<\/span><\/p>\n<p style=\"text-align: justify;\"><span style=\"font-weight: 400;\">Please note that it is not possible to create a pseudonym for a conference ID, so it makes sense to specify a shorter digital CID on the <a href=\"https:\/\/docs.trueconf.com\/server\/en\/admin\/web-config#additional-tab\" target=\"_blank\" rel=\"noopener\"><strong>Advanced<\/strong> tab<\/a> beforehand when scheduling the conference.<\/span><\/p>\n<p><a href=\"https:\/\/trueconf.com\/blog\/wp-content\/uploads\/2019\/07\/cid.png\" data-rel=\"lightbox-gallery-YtxvYxPl\" data-rl_title=\"\" data-rl_caption=\"\" title=\"\" target=\"_blank\" rel=\"noopener\"><img decoding=\"async\" class=\"aligncenter wp-image-15518 size-medium\" style=\"border: 1px solid #D1CCCC;\" src=\"https:\/\/trueconf.com\/blog\/wp-content\/uploads\/2019\/07\/cid-690x340.png\" alt=\"cid\" width=\"690\" height=\"340\" loading=\"lazy\" title=\"\" srcset=\"https:\/\/trueconf.com/blog\/wp-content\/uploads\/2019\/07\/cid-690x340.png 690w, https:\/\/trueconf.com/blog\/wp-content\/uploads\/2019\/07\/cid-768x378.png 768w, https:\/\/trueconf.com/blog\/wp-content\/uploads\/2019\/07\/cid-1024x504.png 1024w, https:\/\/trueconf.com/blog\/wp-content\/uploads\/2019\/07\/cid-290x143.png 290w, https:\/\/trueconf.com/blog\/wp-content\/uploads\/2019\/07\/cid.png 1598w\" sizes=\"auto, (max-width: 690px) 100vw, 690px\" \/><\/a><\/p>\n<h2>How to send DTMF commands<\/h2>\n<p style=\"text-align: justify;\">TrueConf Server supports tone dialing, so you can send the following DTMF commands from your device during a <a href=\"https:\/\/trueconf.com\/features\/modes\/virtual-meeting.html\" target=\"_blank\" rel=\"noopener\">role-based conference<\/a>:<\/p>\n<ul>\n<li style=\"text-align: justify;\" aria-level=\"1\"><code>1<\/code> &#8211; request to take the podium.<\/li>\n<li style=\"text-align: justify;\" aria-level=\"1\"><code>2<\/code> &#8211; to leave the podium.<\/li>\n<\/ul>\n<h2 style=\"text-align: justify;\">How to Call TrueConf Subscribers and Conferences from PSTN?<\/h2>\n<p style=\"text-align: justify;\">Call methods, which we\u2019ve described above, require a telephone connected to a PBX with a set rule for a video conferencing server. And what if you need to call a conference from a GSM cell phone or a regular stationary telephone connected to a public switched telephone network (PSTN)?<\/p>\n<p style=\"text-align: justify;\">It\u2019s simple: add a new option to a voice menu (IVR) in your PBX, which would suggest entering an extension number of a session or user of a video conferencing system in DTMF format. Then assign the rule we\u2019ve created above to handle the numbers entered by a caller to this new section of your voice menu.<\/p>\n<p style=\"text-align: justify;\">Another option: connect an additional external PSTN number to your PBX, all calls to which would be processed immediately by the rule that we\u2019ve created in this article.<\/p>\n","protected":false},"excerpt":{"rendered":"<p>Thanks to integrated gateway, TrueConf Server can receive external calls over the most popular SIP and H.323 telephony protocols, which allows regular telephone subscribers to call users and conferences on TrueConf Server. The only requirement is an installed corporate PBX that would redirect calls from your telephone network to TrueConf Server. In this article, we\u2019re [&hellip;]<\/p>\n","protected":false},"author":45,"featured_media":0,"comment_status":"closed","ping_status":"open","sticky":false,"template":"","format":"standard","meta":{"inline_featured_image":false,"footnotes":""},"categories":[260],"tags":[240],"class_list":["post-8580","post","type-post","status-publish","format-standard","hentry","category-knowledge-base","tag-telephony-integration","wpautop"],"_links":{"self":[{"href":"https:\/\/trueconf.com/blog\/wp-json\/wp\/v2\/posts\/8580","targetHints":{"allow":["GET"]}}],"collection":[{"href":"https:\/\/trueconf.com/blog\/wp-json\/wp\/v2\/posts"}],"about":[{"href":"https:\/\/trueconf.com/blog\/wp-json\/wp\/v2\/types\/post"}],"author":[{"embeddable":true,"href":"https:\/\/trueconf.com/blog\/wp-json\/wp\/v2\/users\/45"}],"replies":[{"embeddable":true,"href":"https:\/\/trueconf.com/blog\/wp-json\/wp\/v2\/comments?post=8580"}],"version-history":[{"count":18,"href":"https:\/\/trueconf.com/blog\/wp-json\/wp\/v2\/posts\/8580\/revisions"}],"predecessor-version":[{"id":17981,"href":"https:\/\/trueconf.com/blog\/wp-json\/wp\/v2\/posts\/8580\/revisions\/17981"}],"wp:attachment":[{"href":"https:\/\/trueconf.com/blog\/wp-json\/wp\/v2\/media?parent=8580"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/trueconf.com/blog\/wp-json\/wp\/v2\/categories?post=8580"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/trueconf.com/blog\/wp-json\/wp\/v2\/tags?post=8580"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}