{"id":23,"date":"2011-05-02T06:13:09","date_gmt":"2011-05-02T06:13:09","guid":{"rendered":"http:\/\/www.myvoipapp.com\/blog\/?p=23"},"modified":"2011-05-04T02:21:19","modified_gmt":"2011-05-04T02:21:19","slug":"sip-trunk","status":"publish","type":"post","link":"https:\/\/www.myvoipapp.com\/blog\/2011\/05\/02\/sip-trunk\/","title":{"rendered":"sip trunk"},"content":{"rendered":"<p>In VOIP depolyment, &#8220;SIP trunk&#8221; is often used to establish a connection with peer sip servers or gateways. For example, in most DID services deployments, SIP trunk is required to send or receive DID calls.<\/p>\n<p>The difference between &#8220;SIP trunk&#8221; and &#8220;External lines&#8221; is that SIP trunk doesn&#8217;t require authorization during the call. That means, &#8220;external line&#8221; is server-to-users mode and &#8220;SIP trunk&#8221; is a server-to-server mode.<\/p>\n<p>It is very easy to establish SIP trunk in MSS.<\/p>\n<p>For example, we want to establish SIP trunk with peer server whose domain name is &#8220;sip.demo.com&#8221; and its SIP port is 5060 which is a default SIP UDP port.<\/p>\n<p><strong>step 1: add the server into MSS servers list<\/strong><\/p>\n<p>Please click menu &#8220;data \/ peer servers&#8221; and add a new record with following information:<\/p>\n<blockquote><p>peer server id=<strong><span style=\"color: #ff0000;\">1<\/span><\/strong><br \/>\ndescription = demo sip server<br \/>\nserver address = sip.demo.com<br \/>\nserver port \u00a0= 5060<\/p><\/blockquote>\n<p><strong>step 2: process incoming call<\/strong><\/p>\n<p>Once we receive incoming calls from peer servers, we want to route them to local users. We can use &#8220;dial plan&#8221; to do that.<\/p>\n<p>For example, we want the DID incoming calls whose called numbers prefix is &#8220;1234&#8221; to local users, such as 1234100 to local user 100, 1234101 to local user 101, etc.<\/p>\n<p>Please click menu &#8220;dial plan \/ transition&#8221; to configure a number transition:<\/p>\n<blockquote><p>transition ID = <strong><span style=\"color: #0000ff;\">1<\/span><\/strong><br \/>\ntransition type = delete<br \/>\nstart position = 0<br \/>\nlength = 4<\/p><\/blockquote>\n<p>Please click menu &#8220;dial plan \/ analysis called number&#8221; to configure a record to route DID numbers to local users:<\/p>\n<blockquote><p>dial plan = default<br \/>\ncalled number prefix = 1234<br \/>\nroute type = local user<br \/>\nchange called number = yes<br \/>\n<span style=\"color: #0000ff;\">transition id = 1<\/span><\/p><\/blockquote>\n<p><strong>step 3: process outgoing call<\/strong><\/p>\n<p>We want our outgoing calls to be routed to such peer SIP server\/gateway. We still need configure &#8220;dial plan&#8221; to do that.<\/p>\n<p>For example, we want all calls whose called number prefix is &#8220;00&#8221; should be routed to such SIP server, such as &#8220;008613800138000&#8221;, etc.<\/p>\n<p>Please click menu &#8220;dial plan \/ analysis called number&#8221; to add a new record with following information:<\/p>\n<blockquote><p>dial plan = default<br \/>\ncalled number prefix = 00<br \/>\nroute type = SIP trunk<br \/>\n<span style=\"color: #ff0000;\">peer server ID = 1<\/span><\/p><\/blockquote>\n<p>&nbsp;<\/p>\n","protected":false},"excerpt":{"rendered":"<p>In VOIP depolyment, &#8220;SIP trunk&#8221; is often used to establish a connection with peer sip servers or gateways. For example, in most DID services deployments, SIP trunk is required to send or receive DID calls. The difference between &#8220;SIP trunk&#8221; and &#8220;External lines&#8221; is that SIP trunk doesn&#8217;t require authorization during the call. That means, &#8220;external line&#8221; is server-to-users mode and &#8220;SIP trunk&#8221; is a server-to-server mode. It is very easy to establish SIP trunk in MSS. For example, we&#8230;<\/p>\n<p class=\"read-more\"><a class=\"btn btn-default\" href=\"https:\/\/www.myvoipapp.com\/blog\/2011\/05\/02\/sip-trunk\/\"> Read More<span class=\"screen-reader-text\">  Read More<\/span><\/a><\/p>\n","protected":false},"author":1,"featured_media":0,"comment_status":"open","ping_status":"open","sticky":false,"template":"","format":"standard","meta":{"footnotes":""},"categories":[6,4],"tags":[18,16,10],"class_list":["post-23","post","type-post","status-publish","format-standard","hentry","category-faq","category-how_to","tag-did","tag-sip-server","tag-voip"],"_links":{"self":[{"href":"https:\/\/www.myvoipapp.com\/blog\/wp-json\/wp\/v2\/posts\/23","targetHints":{"allow":["GET"]}}],"collection":[{"href":"https:\/\/www.myvoipapp.com\/blog\/wp-json\/wp\/v2\/posts"}],"about":[{"href":"https:\/\/www.myvoipapp.com\/blog\/wp-json\/wp\/v2\/types\/post"}],"author":[{"embeddable":true,"href":"https:\/\/www.myvoipapp.com\/blog\/wp-json\/wp\/v2\/users\/1"}],"replies":[{"embeddable":true,"href":"https:\/\/www.myvoipapp.com\/blog\/wp-json\/wp\/v2\/comments?post=23"}],"version-history":[{"count":7,"href":"https:\/\/www.myvoipapp.com\/blog\/wp-json\/wp\/v2\/posts\/23\/revisions"}],"predecessor-version":[{"id":30,"href":"https:\/\/www.myvoipapp.com\/blog\/wp-json\/wp\/v2\/posts\/23\/revisions\/30"}],"wp:attachment":[{"href":"https:\/\/www.myvoipapp.com\/blog\/wp-json\/wp\/v2\/media?parent=23"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/www.myvoipapp.com\/blog\/wp-json\/wp\/v2\/categories?post=23"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/www.myvoipapp.com\/blog\/wp-json\/wp\/v2\/tags?post=23"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}