diff --git a/restcomm/restcomm.docs/jdocbook-mobicents/.project b/restcomm/restcomm.docs/jdocbook-mobicents/.project deleted file mode 100644 index 2084f3cd22..0000000000 --- a/restcomm/restcomm.docs/jdocbook-mobicents/.project +++ /dev/null @@ -1,23 +0,0 @@ - - - restcomm-docs-jdocbook-mobicents - A Restful Communications Framework. NO_M2ECLIPSE_SUPPORT: Project files created with the maven-eclipse-plugin are not supported in M2Eclipse. - - - - - net.sf.eclipsecs.core.CheckstyleBuilder - - - - - org.eclipse.m2e.core.maven2Builder - - - - - - org.eclipse.m2e.core.maven2Nature - net.sf.eclipsecs.core.CheckstyleNature - - diff --git a/restcomm/restcomm.docs/jdocbook-mobicents/.settings/org.eclipse.jdt.core.prefs b/restcomm/restcomm.docs/jdocbook-mobicents/.settings/org.eclipse.jdt.core.prefs deleted file mode 100644 index 2d0195cdfd..0000000000 --- a/restcomm/restcomm.docs/jdocbook-mobicents/.settings/org.eclipse.jdt.core.prefs +++ /dev/null @@ -1,5 +0,0 @@ -#Wed May 08 12:56:12 EDT 2013 -org.eclipse.jdt.core.compiler.codegen.targetPlatform=1.6 -eclipse.preferences.version=1 -org.eclipse.jdt.core.compiler.source=1.6 -org.eclipse.jdt.core.compiler.compliance=1.6 diff --git a/restcomm/restcomm.docs/jdocbook-mobicents/.settings/org.eclipse.m2e.core.prefs b/restcomm/restcomm.docs/jdocbook-mobicents/.settings/org.eclipse.m2e.core.prefs deleted file mode 100644 index 64e89e5224..0000000000 --- a/restcomm/restcomm.docs/jdocbook-mobicents/.settings/org.eclipse.m2e.core.prefs +++ /dev/null @@ -1,5 +0,0 @@ -#Wed Jun 13 10:39:31 EDT 2012 -activeProfiles= -eclipse.preferences.version=1 -resolveWorkspaceProjects=true -version=1 diff --git a/restcomm/restcomm.docs/jdocbook-mobicents/pom.xml b/restcomm/restcomm.docs/jdocbook-mobicents/pom.xml deleted file mode 100644 index 38590b4097..0000000000 --- a/restcomm/restcomm.docs/jdocbook-mobicents/pom.xml +++ /dev/null @@ -1,99 +0,0 @@ - - - - - org.restcomm - restcomm-connect-docs - 7.6.0-SNAPSHOT - - - restcomm-connect-docs-jdocbook-mobicents - - 4.0.0 - - jdocbook - - - - - org.apache.maven.plugins - maven-dependency-plugin - - - unpack - generate-resources - - unpack - - - - - ${pom.groupId} - restcomm-connect-docs-sources-mobicents - ${pom.version} - jar - true - ${project.build.directory}/docbook/resources - - - - - - - - org.jboss.maven.plugins - maven-jdocbook-plugin - 2.3.5 - true - - - org.mobicents.jdocbook - telestax-xslt-ns - 1.3.0.FINAL - - - org.mobicents.jdocbook - telestax-community-style - jdocbook-style - 1.3.0.FINAL - - - - RestComm_User_Guide.xml - ${project.build.directory}/docbook/resources - - ${project.build.directory}/docbook/resources/en-US - - images/* - - - - - pdf - classpath:/xslt/org/jboss/pdf.xsl - RestComm_User_Guide.pdf - - - html - classpath:/xslt/org/jboss/xhtml.xsl - index.html - - - html_single - classpath:/xslt/org/jboss/xhtml-single.xsl - index.html - - - - true - - - 1.72.0 - - - - - - - diff --git a/restcomm/restcomm.docs/jdocbook-telscale/.project b/restcomm/restcomm.docs/jdocbook-telscale/.project deleted file mode 100644 index 2e29cc2200..0000000000 --- a/restcomm/restcomm.docs/jdocbook-telscale/.project +++ /dev/null @@ -1,23 +0,0 @@ - - - restcomm-docs-jdocbook-telscale - A Restful Communications Framework. NO_M2ECLIPSE_SUPPORT: Project files created with the maven-eclipse-plugin are not supported in M2Eclipse. - - - - - net.sf.eclipsecs.core.CheckstyleBuilder - - - - - org.eclipse.m2e.core.maven2Builder - - - - - - org.eclipse.m2e.core.maven2Nature - net.sf.eclipsecs.core.CheckstyleNature - - diff --git a/restcomm/restcomm.docs/jdocbook-telscale/.settings/org.eclipse.jdt.core.prefs b/restcomm/restcomm.docs/jdocbook-telscale/.settings/org.eclipse.jdt.core.prefs deleted file mode 100644 index 42be58495b..0000000000 --- a/restcomm/restcomm.docs/jdocbook-telscale/.settings/org.eclipse.jdt.core.prefs +++ /dev/null @@ -1,5 +0,0 @@ -#Wed May 08 12:56:51 EDT 2013 -org.eclipse.jdt.core.compiler.codegen.targetPlatform=1.6 -eclipse.preferences.version=1 -org.eclipse.jdt.core.compiler.source=1.6 -org.eclipse.jdt.core.compiler.compliance=1.6 diff --git a/restcomm/restcomm.docs/jdocbook-telscale/.settings/org.eclipse.m2e.core.prefs b/restcomm/restcomm.docs/jdocbook-telscale/.settings/org.eclipse.m2e.core.prefs deleted file mode 100644 index 64e89e5224..0000000000 --- a/restcomm/restcomm.docs/jdocbook-telscale/.settings/org.eclipse.m2e.core.prefs +++ /dev/null @@ -1,5 +0,0 @@ -#Wed Jun 13 10:39:31 EDT 2012 -activeProfiles= -eclipse.preferences.version=1 -resolveWorkspaceProjects=true -version=1 diff --git a/restcomm/restcomm.docs/jdocbook-telscale/pom.xml b/restcomm/restcomm.docs/jdocbook-telscale/pom.xml deleted file mode 100644 index 67435d9a54..0000000000 --- a/restcomm/restcomm.docs/jdocbook-telscale/pom.xml +++ /dev/null @@ -1,99 +0,0 @@ - - - - - org.restcomm - restcomm-connect-docs - 7.6.0-SNAPSHOT - - - restcomm-connect-docs-jdocbook-telscale - - 4.0.0 - - jdocbook - - - - - org.apache.maven.plugins - maven-dependency-plugin - - - unpack - generate-resources - - unpack - - - - - ${pom.groupId} - restcomm-connect-docs-sources-telscale - ${pom.version} - jar - true - ${project.build.directory}/docbook/resources - - - - - - - - org.jboss.maven.plugins - maven-jdocbook-plugin - 2.3.5 - true - - - org.mobicents.jdocbook - telestax-xslt-ns - 1.3.0.FINAL - - - org.mobicents.jdocbook - telestax-telscale-style - jdocbook-style - 1.3.0.FINAL - - - - RestComm_User_Guide.xml - ${project.build.directory}/docbook/resources - - ${project.build.directory}/docbook/resources/en-US - - images/* - - - - - pdf - classpath:/xslt/org/jboss/pdf.xsl - RestComm_User_Guide.pdf - - - html - classpath:/xslt/org/jboss/xhtml.xsl - index.html - - - html_single - classpath:/xslt/org/jboss/xhtml-single.xsl - index.html - - - - true - - - 1.72.0 - - - - - - - diff --git a/restcomm/restcomm.docs/pom.xml b/restcomm/restcomm.docs/pom.xml index c21d052476..fd82f4a26d 100644 --- a/restcomm/restcomm.docs/pom.xml +++ b/restcomm/restcomm.docs/pom.xml @@ -17,43 +17,6 @@ sources-asciidoc - diff --git a/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/api/geolocation-api.adoc b/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/api/geolocation-api.adoc index 84ab6e8ca0..cd410c56d9 100644 --- a/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/api/geolocation-api.adoc +++ b/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/api/geolocation-api.adoc @@ -379,7 +379,6 @@ positioning procedure:    Mon, 25 Jan 2016 16:36:12 -0500    Mon, 25 Jan 2016 16:36:10 -0500    ACae6e420f425248d6a26948c17a9e2acf -   david@cloud.restcomm.com    client:david    immediate    partially-successful @@ -415,7 +414,6 @@ not accomplished yet):    Mon, 25 Jan 2016 16:36:25 -0500    Mon, 25 Jan 2016 16:36:10 -0500    ACae6e420f425248d6a26948c17a9e2acf -   david@cloud.restcomm.com    client:david    immediate    partially-successful @@ -449,7 +447,6 @@ accuracy is accomplished through GPS assistance:    Mon, 25 Jan 2016 16:38:24 -0500    Mon, 25 Jan 2016 16:36:10 -0500    ACae6e420f425248d6a26948c17a9e2acf -   david@cloud.restcomm.com    client:david    immediate    partially-successful @@ -500,7 +497,6 @@ See the corresponding response below:    Mon, 25 Jan 2016 16:37:21 -0300    Mon, 25 Jan 2016 16:36:10 -0300    ACae6e420f425248d6a26948c17a9e2acf -   598995498785    59897018375    immediate    successful @@ -552,7 +548,6 @@ See the corresponding response below:    Mon, 25 Jan 2016 16:36:11 +0200    Mon, 25 Jan 2016 16:36:10 +0200    ACae6e420f425248d6a26948c17a9e2acf -   eTop    39897018375    immediate    partially-successful @@ -597,7 +592,6 @@ See the corresponding response below:    Mon, 25 Jan 2016 16:36:37 -0500    Mon, 25 Jan 2016 16:36:10 -0500    ACae6e420f425248d6a26948c17a9e2acf -   david@cloud.restcomm.com    sip:david@65.17.24.177    immediate    failed @@ -631,7 +625,6 @@ See the corresponding response below: Mon, 25 Jan 2016 20:40:10 -0500 Mon, 25 Jan 2016 16:36:10 -0500 ACae6e420f425248d6a26948c17a9e2acf - david@cloud.restcomm.com sip:david@65.17.24.177 Immediate last-known @@ -667,7 +660,6 @@ See the corresponding JSON response below (the XML response would be exactly as "date_updated": "Mon, 25 Jan 2016 20:40:10 -0500", "date_executed": "Mon, 25 Jan 2016 16:36:10 -0500", "account_sid": "ACae6e420f425248d6a26948c17a9e2acf", - "source": "david@cloud.restcomm.com", "device_identifier": "sip:david@65.17.24.177", "geolocation_type": "Immediate", "response_status": "last-known", @@ -867,7 +859,6 @@ is obtained:    Mon, 25 Jan 2016 16:36:15 -0500    Mon, 25 Jan 2016 16:36:10 -0500    ACae6e420f425248d6a26948c17a9e2acf -   BLE002    56790122158    notification    partially-successful @@ -902,7 +893,6 @@ not accomplished yet):    Mon, 25 Jan 2016 16:36:44 -0500    Mon, 25 Jan 2016 16:36:10 -0500    ACae6e420f425248d6a26948c17a9e2acf -   BLE002    56790122158    notification    partially-successful @@ -935,7 +925,6 @@ accuracy is accomplished through GPS assistance:    Mon, 25 Jan 2016 16:37:04 -0500    Mon, 25 Jan 2016 16:36:10 -0500    ACae6e420f425248d6a26948c17a9e2acf -   BLE002    56790122158    notification    partially-successful @@ -1056,7 +1045,6 @@ See the corresponding response below:    Mon, 25 Jan 2016 16:41:10 +0900    Mon, 25 Jan 2016 16:36:10 +0900    ACae6e420f425248d6a26948c17a9e2acf -   SW123    SB7089A    notification    successful @@ -1119,7 +1107,6 @@ See the corresponding response below:    Mon, 25 Jan 2016 16:37:18 -0300    Mon, 25 Jan 2016 16:36:10 -0300    ACae6e420f425248d6a26948c17a9e2acf -   59897018375    59897018375    notification    successful @@ -1173,7 +1160,6 @@ See the corresponding response below:    Mon, 25 Jan 2016 16:39:18 -0300    Mon, 25 Jan 2016 16:36:10 -0300    ACae6e420f425248d6a26948c17a9e2acf -   59897018375    59897018375    notification    partially-successful @@ -1220,7 +1206,6 @@ See the corresponding JSON response below (the XML response would be exactly as "date_updated": "Mon, 25 Jan 2016 16:39:18 -0300", "date_executed": "Mon, 25 Jan 2016 16:36:10 -0300", "account_sid": "ACae6e420f425248d6a26948c17a9e2acf", - "source": "59897018375", "device_identifier": "59897018375", "geolocation_type": "Notification", "response_status": "partially-successful", @@ -1271,7 +1256,7 @@ Clients and RestComm Mobile/Web SDKs -image:IPGeolocationSequenceDiagram.gif[image,width=633,height=230] +image:images/IPGeolocationSequenceDiagram.gif[image,width=633,height=230] @@ -1279,8 +1264,27 @@ For the Notification case the diagram is similar, except that the device can store the information and notify RestComm when it approaches a certain location area. +== Cellular Geolocation Sequence Diagrams +Next diagram shows the interaction between RestComm and GMLC within a GSM network, from where location services are reduced to retrieving Global Cell Identity, Age of Location information and network node address at which the target mobile subscriber is +currently currently attached. + +image:images/RestComm_GSM_Immediate_Geolocation.png[image,width=633,height=230] + + +Next diagram exhibits a Notification type of Geolocation signal flow in 3G cellular networks. An Immediate type of Geolocation signal flow in the same environment would be identical, except for the event detection and its derived signals. Besides, for the sake of simplicity, it only shows a single event detection. + +image:images/RestComm_UMTS_Notification_Geolocation.png[image,width=750,height=400] + + + +Next diagram is the analogue of the latter, but for EPS networks or LTE location services (where SS7/MAP operations do not apply anymore, but their analogous Diameter procedures with EPC and E-UTRAN entities). + + + +image:images/RestComm_LTE_Notification_Geolocation.png[image,width=750,height=400] + == Geolocation Status Callbacks Sequence Diagram @@ -1298,7 +1302,7 @@ last status callback, as the desired accuracy is ultimately achieved. -image:GeolocationStatusCallbacksSequenceDiagram.gif[image,width=633,height=348] +image:images/GeolocationStatusCallbacksSequenceDiagram.gif[image,width=633,height=348] diff --git a/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/api/GeolocationStatusCallbacksSequenceDiagram.gif b/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/api/images/GeolocationStatusCallbacksSequenceDiagram.gif similarity index 100% rename from restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/api/GeolocationStatusCallbacksSequenceDiagram.gif rename to restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/api/images/GeolocationStatusCallbacksSequenceDiagram.gif diff --git a/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/api/IPGeolocationSequenceDiagram.gif b/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/api/images/IPGeolocationSequenceDiagram.gif similarity index 100% rename from restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/api/IPGeolocationSequenceDiagram.gif rename to restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/api/images/IPGeolocationSequenceDiagram.gif diff --git a/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/api/images/RestComm_GSM_Immediate_Geolocation.png 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b/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/sdks/images/olympus-texting-olympus-2.jpg differ diff --git a/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/sdks/index.adoc b/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/sdks/index.adoc index 9dcd64654e..c897a45caa 100644 --- a/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/sdks/index.adoc +++ b/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/sdks/index.adoc @@ -1,21 +1,25 @@ = Restcomm SDKs -Restcomm comes with three powerful WebRTC SDKs that allows you to embed Communications as a feature in your Web, iOS and Android Native Applications. +Restcomm comes with three powerful WebRTC SDKs that allow you to embed Communications as a feature in your Web, iOS and Android Native Applications. Moreover, there's Olympus, a full fledged Client that take advantage of the respective Restcomm SDKs and provides webrtc video/audio calling and messaging capabilities on each of Web, iOS and Android platforms out of the box. Web SDK ~~~~~~~ * <> * link:http://documentation.telestax.com/restcomm-web-sdk/doc/[Restcomm Web SDK Reference Documentation] +* <> + iOS SDK ~~~~~~~ * <> * link:http://documentation.telestax.com/restcomm-ios-sdk/doc/html/[Restcomm iOS SDK Reference Documentation] +* <> Android SDK ~~~~~~~~~~~ * <> * link:http://documentation.telestax.com/restcomm-android-sdk/doc/[Restcomm Android SDK Reference Documentation] +* <> diff --git a/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/sdks/restcomm-android-olympus-user-guide.adoc b/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/sdks/restcomm-android-olympus-user-guide.adoc new file mode 100644 index 0000000000..a1c9603650 --- /dev/null +++ b/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/sdks/restcomm-android-olympus-user-guide.adoc @@ -0,0 +1,53 @@ +The Restcomm Android SDK comes with Olympus Client Android App that taps into the SDK and presents to the user a full fledged application capable of video/audio WebRTC calls and text messages. If we add Restcomm Platform to the mix, we get full telecom programmability and access to PSTN/SMS. Let's see how this works. + +You can get the latest version of Android Olympus .apk from the link found https://github.com/RestComm/restcomm-android-sdk#latest-olympus-apk[here]. For the TADHack event you will also need to register for a new account https://accounts.restcomm.com/tadhackRegistration.jsp[here] and follow along the registration process until you get your password via email. + +Then you need to launch the Olympus Android App and use the user part of your email and the password you received in the first, wizard screen. Also, you will need to change domain to tadhack.restcomm.com and tap Done: + +image:./images/android-olympus-signup.png[android-olympus-signup] + +Once you login, you can call any of the default contacts, be it a Restcomm Application, like +1235 or another client like Bob who needs to be registered from another Olympus or generic SIP client. Here's the contacts screen: + +image:./images/android-olympus-contacts.png[android-olympus-contacts] + +[[calling-a-restcomm-application]] +Calling a Restcomm Application +^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ + +Let's first try an audio call towards Restcomm Application +1235 that uses Text-to-Speech to greet you with and audio message. + +* Tap on the 'Say App' and on the new screen that opens tap on the Audio Call icon on the upper right corner: + +image:./images/android-olympus-calling-restcomm-app-1.png[android-olympus-calling-restcomm-app] + +* You should then see the call screen and hear the message: + +image:./images/android-olympus-calling-restcomm-app-2.png[android-olympus-calling-restcomm-app] + +[[calling-another-restcomm-client]] +Calling another Restcomm Client +^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ + +Now let's try a Client to Client video call. + +* Back in the Contacts screen tap on the plus sign on the lower right corner to create a new contact, add the client's details and tap Add: + +image:./images/android-olympus-add-contact.png[android-olympus-add-contact] + +* Tap on the new contact and when the message screen comes up tap on the Video Call icon on the upper right corner so that a video call is made to that contact: + +image:./images/android-olympus-calling-olympus-1.png[android-olympus-calling-olympus] + +* Remember that the called client (in my case Despina) needs to be registered from another device. Once you answer from the called client you should see something like this: + +image:./images/android-olympus-calling-olympus-2.jpg[android-olympus-calling-olympus] + +[[sending-a-text-message-to-another-restcomm-client]] +Sending a text message to another Restcomm Client +^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ + +Let's send a text message from Olympus to another client. Back to Contacts screen tap on the little (i) icon on the right of the peer you want to message (in my case Despina). On the new screen that opens tap on the Message and then write a message and tap Send: + +image:./images/android-olympus-texting-olympus.png[android-olympus-texting-olympus] + +That's about it. By taking further advantage of Restcomm's Apps capabilities you can now add a wealth of features in your flow, be it IVR, Conferencing, etc. diff --git a/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/sdks/restcomm-ios-olympus-user-guide.adoc b/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/sdks/restcomm-ios-olympus-user-guide.adoc new file mode 100644 index 0000000000..af55b54f50 --- /dev/null +++ b/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/sdks/restcomm-ios-olympus-user-guide.adoc @@ -0,0 +1,43 @@ +The Restcomm iOS SDK comes with Olympus Client iOS App that taps into the SDK and presents to the user a full fledged application capable of video/audio WebRTC calls and text messages. If we add Restcomm Platform to the mix, we get full telecom programmability and access to PSTN/SMS. Let's see how this works. + +You can get the latest version of iOS Olympus .ipa from the link found https://github.com/RestComm/restcomm-ios-sdk#latest-olympus-ipa[here]. For the TADHack event you will also need to register for a new account https://accounts.restcomm.com/tadhackRegistration.jsp[here] and follow along the registration process until you get your password via email. + +Before you launch the App you need to trust Telestax Inc. on your iOS device by following this https://support.apple.com/en-us/HT204460[Apple Guide]. Then you need to launch the Olympus iOS App and use the user part of your email and the password you received in the first, wizard screen. Also, you will need to change domain to tadhack.restcomm.com and tap Done: + +image:./images/ios-olympus-signup.png[ios-olympus-signup] + +Once you login, you can call any of the default contacts, be it a Restcomm Application, like +1235 or another client like Bob who needs to be registered from another Olympus or generic SIP client. Here's the contacts screen: + +image:./images/ios-olympus-contacts.png[ios-olympus-contacts] + +[[calling-a-restcomm-application]] +Calling a Restcomm Application +^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ + +Let's first try an audio call towards Restcomm Application +1235 that uses Text-to-Speech to greet you with and audio message. Tap on the little (i) icon on the right of the 'Say App' and on the new screen that opens tap on the Audio Call: + +image:./images/ios-olympus-calling-restcomm-app.png[ios-olympus-calling-restcomm-app] + +[[calling-another-restcomm-client]] +Calling another Restcomm Client +^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ + +Now let's try a Client to Client video call. + +* Back in the Contacts screen tap on the plus sign on the top right corner to create a new contact, add the client's details and tap Done: + +image:./images/ios-olympus-add-contact.png[ios-olympus-add-contact] + +* Tap on the new contact (not on the (i) icon but anywhere else) so that a video call is made to that contact. Remember that the called client (in my case Despina) needs to be registered from another device. Once you answser from the called client you should see something like this: + +image:./images/ios-olympus-calling-olympus.jpg[ios-olympus-calling-olympus] + +[[sending-a-text-message-to-another-restcomm-client]] +Sending a text message to another Restcomm Client +^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ + +Let's send a text message from Olympus to another client. Back to Contacts screen tap on the little (i) icon on the right of the peer you want to message (in my case Despina). On the new screen that opens tap on the Message and then write a message and tap Send: + +image:./images/ios-olympus-texting-olympus.png[ios-olympus-texting-olympus] + +That's about it. By taking further advantage of Restcomm's Apps capabilities you can now add a wealth of features in your flow, be it IVR, Conferencing, etc. diff --git a/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/sdks/restcomm-web-olympus-user-guide.adoc b/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/sdks/restcomm-web-olympus-user-guide.adoc new file mode 100644 index 0000000000..2184df7caa --- /dev/null +++ b/restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/sdks/restcomm-web-olympus-user-guide.adoc @@ -0,0 +1,46 @@ +The Restcomm Web SDK comes with Olympus Client Web App that taps into the SDK and presents to the user a full fledged application capable of video/audio WebRTC calls and text messages. If we add Restcomm Platform to the mix, we get full telecom programmability and access to PSTN/SMS. Let's see how this works. + +You can use the latest version of Web Olympus by first registering for a new account https://accounts.restcomm.com/tadhackRegistration.jsp[here] and following along the registration process until you get your password via email. Then you can login to Olympus by using the user part of your email and the password you received. + +Once you login, you can use the Contacts menu on the left to call any of the default contacts, be it a Restcomm Application, like +1235 or another client like Bob who needs to be registered from another Olympus or generic SIP client. + +[[calling-a-restcomm-application]] +Calling a Restcomm Application +^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ + +Let's first try an audio call towards Restcomm Application +1235 that uses Text-to-Speech to greet you: + +image:./images/olympus-calling-restcomm-app.jpg[olympus-calling-restcomm-app] + +[[calling-another-restcomm-client]] +Calling another Restcomm Client +^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ + +Now let's try a Client to Client video call. In the screenshots below Bob is calling Alice. + +* Click on Contact icon, then Alice and finally do a video call as can be seen below: + +image:./images/olympus-calling-olympus-1.jpg[olympus-calling-olympus] + +* At that point, Olympus will start to ring in Alice’s window: + +image:./images/olympus-calling-olympus-2.jpg[olympus-calling-olympus] + +* Pick up from Alice tab and you are ready to talk (I'm calling myself which admittedly isn't very productive, but just to get an idea): + +image:./images/olympus-calling-olympus-3.jpg[olympus-calling-olympus] + +[[sending-a-text-message-to-another-restcomm-client]] +Sending a text message to another Restcomm Client +^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ + +Click on the Contacts icon, then Alice and finally send a text message as can be seen below: + +image:./images/olympus-texting-olympus-1.jpg[olympus-texting-olympus] + +On the new text window that comes up write a text message and hit send. Go to the other client's window to see the incoming message and do the same to reply + +image:./images/olympus-texting-olympus-2.jpg[olympus-texting-olympus] + +That's about it. By taking further advantage of Restcomm's Apps capabilities you can now add a wealth of features in your flow, be it IVR, Conferencing, etc. + diff --git a/restcomm/restcomm.docs/sources-mobicents/.classpath b/restcomm/restcomm.docs/sources-mobicents/.classpath deleted file mode 100644 index cb76c82489..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/.classpath +++ /dev/null @@ -1,27 +0,0 @@ - - - - - - - - - - - - - - - - - - - - - - - - - - - diff --git a/restcomm/restcomm.docs/sources-mobicents/.gitignore b/restcomm/restcomm.docs/sources-mobicents/.gitignore deleted file mode 100644 index ea8c4bf7f3..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/.gitignore +++ /dev/null @@ -1 +0,0 @@ -/target diff --git a/restcomm/restcomm.docs/sources-mobicents/.project b/restcomm/restcomm.docs/sources-mobicents/.project deleted file mode 100644 index a0325a25b4..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/.project +++ /dev/null @@ -1,29 +0,0 @@ - - - restcomm-docs-sources-mobicents - A Restful Communications Framework. NO_M2ECLIPSE_SUPPORT: Project files created with the maven-eclipse-plugin are not supported in M2Eclipse. - - - - - org.eclipse.jdt.core.javabuilder - - - - - net.sf.eclipsecs.core.CheckstyleBuilder - - - - - org.eclipse.m2e.core.maven2Builder - - - - - - org.eclipse.m2e.core.maven2Nature - org.eclipse.jdt.core.javanature - net.sf.eclipsecs.core.CheckstyleNature - - diff --git a/restcomm/restcomm.docs/sources-mobicents/.settings/org.eclipse.jdt.core.prefs b/restcomm/restcomm.docs/sources-mobicents/.settings/org.eclipse.jdt.core.prefs deleted file mode 100644 index ec4300d5d0..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/.settings/org.eclipse.jdt.core.prefs +++ /dev/null @@ -1,5 +0,0 @@ -eclipse.preferences.version=1 -org.eclipse.jdt.core.compiler.codegen.targetPlatform=1.7 -org.eclipse.jdt.core.compiler.compliance=1.7 -org.eclipse.jdt.core.compiler.problem.forbiddenReference=warning -org.eclipse.jdt.core.compiler.source=1.7 diff --git a/restcomm/restcomm.docs/sources-mobicents/.settings/org.eclipse.m2e.core.prefs b/restcomm/restcomm.docs/sources-mobicents/.settings/org.eclipse.m2e.core.prefs deleted file mode 100644 index 1bafefd517..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/.settings/org.eclipse.m2e.core.prefs +++ /dev/null @@ -1,5 +0,0 @@ -#Thu Nov 01 22:16:21 EDT 2012 -activeProfiles= -eclipse.preferences.version=1 -resolveWorkspaceProjects=true -version=1 diff --git a/restcomm/restcomm.docs/sources-mobicents/pom.xml b/restcomm/restcomm.docs/sources-mobicents/pom.xml deleted file mode 100644 index 7ca1ded677..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/pom.xml +++ /dev/null @@ -1,79 +0,0 @@ - - - - 4.0.0 - - - org.restcomm - restcomm-connect-docs - 7.6.0-SNAPSHOT - - - restcomm-connect-docs-sources-mobicents - - - Mobicents - RestComm - restcomm - ${project.version} - ivelin.atanasoff.ivanov (at) gmail.com - jean.deruelle (at) gmail.com - quintana.thomas (at) gmail.com - - - - - - ${basedir}/src/main/resources - true - - - - - org.apache.maven.plugins - maven-dependency-plugin - - - unpack - generate-resources - - unpack - - - - - ${pom.groupId} - restcomm-connect-docs-sources - ${pom.version} - jar - true - ${basedir}/src/main/resources - - - - - - - - maven-antrun-plugin - - - clean-resources - clean - - run - - - - - - - - - - - - - - - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/META-INF/MANIFEST.MF b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/META-INF/MANIFEST.MF deleted file mode 100644 index 806126f0f6..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/META-INF/MANIFEST.MF +++ /dev/null @@ -1,6 +0,0 @@ -Manifest-Version: 1.0 -Built-By: croufay -Build-Jdk: 1.7.0_79 -Created-By: Apache Maven 3.3.9 -Archiver-Version: Plexus Archiver - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/META-INF/maven/com.telestax.servlet/restcomm-docs-sources/pom.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/META-INF/maven/com.telestax.servlet/restcomm-docs-sources/pom.xml deleted file mode 100644 index 122a733397..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/META-INF/maven/com.telestax.servlet/restcomm-docs-sources/pom.xml +++ /dev/null @@ -1,30 +0,0 @@ - - - - 4.0.0 - - - org.restcomm - restcomm-connect-docs - 7.4.0-SNAPSHOT - - - restcomm-connect-docs-sources - - - RestComm - 2011 - RestComm_User_Guide - Mobicents - - - - - - ${basedir}/src/main/resources - true - - - - - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Author_Group.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Author_Group.xml deleted file mode 100644 index 870ca675f2..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Author_Group.xml +++ /dev/null @@ -1,27 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - Jean - Deruelle - jean.deruelle@gmail.com - - - Thomas - Quintana - quintana.thomas@gmail.com - - - Charles - Roufay - charles.roufay@telestax.com - - - George - Vagenas - gvagenas@telestax.com - - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Book_Info.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Book_Info.xml deleted file mode 100644 index 43a8fd81f7..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Book_Info.xml +++ /dev/null @@ -1,44 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - RestComm User Guide - The Guide to the RestComm - Mobicents Platform - 1.0.0 - 1.0 - 1 - - - RestComm is a carrier-grade open source platform that provides developers the tools - to integrate fax, voice, and SMS functionality in to their own applications with ease. RestComm - is designed to have 100% compatibility with Twilio's APIs allowing easy porting between - platforms. - - Furthermore, the RestComm platform is built on top of the industry leading Mobicents - Sip Servlet Container and Mobicents Media Server providing the robustness and performance - these platforms are already known to deliver. - - - - - &YEAR; - - &HOLDER; - - - - - - - - - - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Common_Content/Conventions.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Common_Content/Conventions.xml deleted file mode 100644 index 93fb0518a6..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Common_Content/Conventions.xml +++ /dev/null @@ -1,334 +0,0 @@ - - -%BOOK_ENTITIES; -]> - -
- Document Conventions - - This manual uses several conventions to highlight certain words - and phrases and draw attention to specific pieces of information. - - - In PDF and paper editions, this manual uses typefaces drawn from the - Liberation Fonts - set. The Liberation Fonts set is also used in HTML editions if the set - is installed on your system. If not, alternative but equivalent - typefaces are displayed. Note: Red Hat Enterprise Linux 5 and later - includes the Liberation Fonts set by default. - -
- Typographic Conventions - - Four typographic conventions are used to call attention to - specific words and phrases. These conventions, and the circumstances - they apply to, are as follows. - - - Mono-spaced Bold - - - Used to highlight system input, including shell commands, file - names and paths. Also used to highlight key caps and - key-combinations. For example: - -
- - To see the contents of the file - my_next_bestselling_novel - in your current working directory, enter the - cat my_next_bestselling_novel - command at the shell prompt and press - Enter - to execute the command. - -
- - The above includes a file name, a shell command and a key cap, - all presented in Mono-spaced Bold and all distinguishable thanks to - context. - - - Key-combinations can be distinguished from key caps by the - hyphen connecting each part of a key-combination. For example: - -
- - Press - Enter - to execute the command. - - - Press - - Ctrl - Alt - F1 - - to switch to the first virtual terminal. Press - - Ctrl - Alt - F7 - - to return to your X-Windows session. - -
- - The first sentence highlights the particular key cap to press. - The second highlights two sets of three key caps, each set pressed - simultaneously. - - - If source code is discussed, class names, methods, functions, - variable names and returned values mentioned within a paragraph will - be presented as above, in - Mono-spaced Bold - . For example: - -
- - File-related classes include - filesystem - for file systems, - file - for files, and - dir - for directories. Each class has its own associated set of - permissions. - -
- - Proportional Bold - - - This denotes words or phrases encountered on a system, including - application names; dialogue box text; labelled buttons; check-box and - radio button labels; menu titles and sub-menu titles. For example: - -
- - Choose - System > Preferences > Mouse - from the main menu bar to launch - Mouse Preferences - . In the - Buttons - tab, click the - Left-handed mouse - check box and click - Close - to switch the primary mouse button from the left to the right - (making the mouse suitable for use in the left hand). - - - To insert a special character into a - gedit - file, choose - Applications > Accessories > Character Map - from the main menu bar. Next, choose - Search > Find… - from the - Character Map - menu bar, type the name of the character in the - Search - field and click - Next - . The character you sought will be highlighted in the - Character Table - . Double-click this highlighted character to place it in the - Text to copy - field and then click the - Copy - button. Now switch back to your document and choose - Edit > Paste - from the - gedit - menu bar. - -
- - The above text includes application names; system-wide menu - names and items; application-specific menu names; and buttons and - text found within a GUI interface, all presented in Proportional Bold - and all distinguishable by context. - - - Note the - > - shorthand used to indicate traversal through a menu and its - sub-menus. This is to avoid the difficult-to-follow 'Select - Mouse - from the - Preferences - sub-menu in the - System - menu of the main menu bar' approach. - - - - Mono-spaced Bold Italic - - or - - Proportional Bold Italic - - - - Whether Mono-spaced Bold or Proportional Bold, the addition of - Italics indicates replaceable or variable text. Italics denotes text - you do not input literally or displayed text that changes depending - on circumstance. For example: - -
- - To connect to a remote machine using ssh, type - - ssh - username - @ - domain.name - - at a shell prompt. If the remote machine is - example.com - and your username on that machine is john, type - ssh john@example.com - . - - - The - - mount -o remount - file-system - - command remounts the named file system. For example, to remount the - /home - file system, the command is - mount -o remount /home - . - - - To see the version of a currently installed package, use the - - rpm -q - package - - command. It will return a result as follows: - - package-version-release - - . - -
- - Note the words in bold italics above — username, - domain.name, file-system, package, version and release. Each word is - a placeholder, either for text you enter when issuing a command or - for text displayed by the system. - - - Aside from standard usage for presenting the title of a work, - italics denotes the first use of a new and important term. For - example: - -
- - When the Apache HTTP Server accepts requests, it dispatches child - processes or threads to handle them. This group of child processes - or threads is known as a - server-pool - . Under Apache HTTP Server 2.0, the responsibility for creating and - maintaining these server-pools has been abstracted to a group of - modules called - Multi-Processing Modules - ( - MPMs - ). Unlike other modules, only one module from the MPM group can be - loaded by the Apache HTTP Server. - -
-
- -
- Pull-quote Conventions - - Two, commonly multi-line, data types are set off visually from - the surrounding text. - - - Output sent to a terminal is set in - Mono-spaced Roman - and presented thus: - - - -books Desktop documentation drafts mss photos stuff svn -books_tests Desktop1 downloads images notes scripts svgs - - - Source-code listings are also set in - Mono-spaced Roman - but are presented and highlighted as follows: - - - -package org.jboss.book.jca.ex1; - -import javax.naming.InitialContext; - -public class ExClient -{ - public static void main(String args[]) - throws Exception - { - InitialContext iniCtx = new InitialContext(); - Object ref = iniCtx.lookup("EchoBean"); - EchoHome home = (EchoHome) ref; - Echo echo = home.create(); - - System.out.println("Created Echo"); - - System.out.println("Echo.echo('Hello') = " + echo.echo("Hello")); - } - -} - -
- -
- Notes and Warnings - - Finally, we use three visual styles to draw attention to - information that might otherwise be overlooked. - - - Note - - A note is a tip or shortcut or alternative approach to the task - at hand. Ignoring a note should have no negative consequences, but - you might miss out on a trick that makes your life easier. - - - - Important - - Important boxes detail things that are easily missed: - configuration changes that only apply to the current session, or - services that need restarting before an update will apply. Ignoring - Important boxes won't cause data loss but may cause irritation - and frustration. - - - - Warning - - A Warning should not be ignored. Ignoring warnings will most - likely cause data loss. - - -
- -
- - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Common_Content/Feedback.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Common_Content/Feedback.xml deleted file mode 100644 index 7610140577..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Common_Content/Feedback.xml +++ /dev/null @@ -1,35 +0,0 @@ - - -%BOOK_ENTITIES; -]> - -
- - Provide feedback to the authors! - - - feedback - - - - If you find a typographical error in this manual, or if you have - thought of a way to make this manual better, we would love to hear - from you! Please submit a report in Bugzilla: - http://http://code.google.com/p/restcomm/issues/ - against the product - RestComm - , or contact the authors. - - - When submitting a bug report, be sure to mention the manual's - identifier: &BOOKID; - - If you have a suggestion for improving the documentation, try to - be as specific as possible when describing it. If you have found an - error, please include the section number and some of the surrounding - text so we can find it easily. - -
- - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Common_Content/Java_Development_Kit-Installing_Configuring_and_Running.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Common_Content/Java_Development_Kit-Installing_Configuring_and_Running.xml deleted file mode 100644 index 4aec89aadb..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Common_Content/Java_Development_Kit-Installing_Configuring_and_Running.xml +++ /dev/null @@ -1,449 +0,0 @@ - - -%BOOK_ENTITIES; - -]> -
- - Java Development Kit ( - <acronym>JDK</acronym> - ): Installing, Configuring and Running - - - The - &PLATFORM_NAME; - Platform - - is written in Java; therefore, before running any - &PLATFORM_NAME; - - server, you must have a working Java Runtime Environment ( - JRE - ) or Java Development Kit ( - JDK - ) installed on your system. In addition, the JRE or JDK you are using - to run - &PLATFORM_NAME; - - must be version 5 or higher - - - At this point in time, it is possible to run most - &PLATFORM_NAME; - - servers, such as the JAIN SLEE Server, using a Java 6 JRE or JDK. Be - aware, however, that presently the XML Document Management Server - does not run on Java 6. We suggest checking the &PLATFORM_NAME; - web site, forums or discussion pages if you need to inquire about - the status of running the XML Document Management Server with Java - 6. - - - . - - - Should I Install the JRE or JDK? - - Although you can run - &PLATFORM_NAME; - - servers using the Java Runtime Environment, we assume that most users - are developers interested in developing Java-based, - &PLATFORM_NAME; - - -driven solutions. Therefore, in this guide we take the tact of - showing how to install the full Java Development Kit. - - - - Should I Install the 32-Bit or the 64-Bit JDK, and Does It - Matter? - - Briefly stated: if you are running on a 64-Bit Linux or Windows - platform, you should consider installing and running the 64-bit JDK - over the 32-bit one. Here are some heuristics for determining whether - you would rather run the 64-bit Java Virtual Machine (JVM) over its - 32-bit cousin for your application: - - - - - - Wider datapath: the pipe between RAM and CPU is doubled, which - improves the performance of memory-bound applications when using a - 64-bit JVM. - - - - - 64-bit memory addressing gives virtually unlimited (1 exabyte) - heap allocation. However large heaps affect garbage collection. - - - - - Applications that run with more than 1.5 GB of RAM (including - free space for garbage collection optimization) should utilize the - 64-bit JVM. - - - - - Applications that run on a 32-bit JVM and do not require more - than minimal heap sizes will gain nothing from a 64-bit JVM. Barring - memory issues, 64-bit hardware with the same relative clock speed - and architecture is not likely to run Java applications faster than - their 32-bit cousin. - - - - - Note that the following instructions detail how to download and - install the 32-bit JDK, although the steps are nearly identical for - installing the 64-bit version. - - - - - Downloading - - You can download the Sun JDK 5.0 (Java 2 Development Kit) from - Sun's website: - - . Click on the - Download - link next to "JDK 5.0 Update - <x> - " (where - <x> - is the latest minor version release number). On the next page, select - your language and platform (both architecture—whether 32- or - 64-bit—and operating system), read and agree to the - Java Development Kit 5.0 License Agreement - , and proceed to the download page. - - - - The Sun website will present two download alternatives to you: one is - an RPM inside a self-extracting file (for example, - jdk-1_5_0_16-linux-i586-rpm.bin - ), and the other is merely a self-extracting file (e.g. - jdk-1_5_0_16-linux-i586.bin - ). If you are installing the JDK on Red Hat Enterprise Linux, Fedora, - or another RPM-based Linux system, we suggest that you download the - self-extracting file containing the RPM package, which will set up and - use the SysV service scripts in addition to installing the JDK. We - also suggest installing the self-extracting RPM file if you will be - running - &PLATFORM_NAME; - - in a production environment. - - - - Installing - - The following procedures detail how to install the Java - Development Kit on both Linux and Windows. - - - - Installing the JDK on Linux - - - Regardless of which file you downloaded, you can install it on - Linux by simply making sure the file is executable and then running - it: - - - ~]$ chmod +x "jdk-1_5_0_<minor_version>-linux-<architecture>-rpm.bin" -~]$ ./"jdk-1_5_0_<minor_version>-linux-<architecture>-rpm.bin" - - - - - You Installed Using the Non-RPM Installer, but Want the SysV - Service Scripts - - If you download the non-RPM self-extracting file (and installed it), - and you are running on an RPM-based system, you can still set up the - SysV service scripts by downloading and installing one of the - -compat - packages from the JPackage project. Remember to download the - -compat - package which corresponds correctly to the minor release number of - the JDK you installed. The compat packages are available from - - . - - - - - You do not need to install a - -compat - package in addition to the JDK if you installed the self-extracting - RPM file! The - -compat - package merely performs the same SysV service script set up that the - RPM version of the JDK installer does. - - - - Installing the JDK on Windows - - - Using Explorer, simply double-click the downloaded - self-extracting installer and follow the instructions to install the - JDK. - - - - - - - Configuring - - Configuring your system for the JDK consists in two tasks: setting - the - JAVA_HOME - environment variable, and ensuring that the system is using the - proper JDK (or JRE) using the - alternatives - command. Setting - JAVA_HOME - usually overrides the values for - java - , - javac - and - java_sdk_1.5.0 - in - alternatives - , but we will set them all just to be safe and consistent. - - - - - - Setting the - JAVA_HOME - Environment Variable on Generic Linux - - - - After installing the JDK, you must ensure that the - JAVA_HOME - environment variable exists and points to the location of your JDK - installation. - - - - Setting the - <envar>JAVA_HOME</envar> - Environment Variable on Linux - - - You can determine whether - JAVA_HOME - is set on your system by - echo - ing it on the command line: - - - - ~]$ echo $JAVA_HOME - - If - JAVA_HOME - is not set already, then you must set its value to the location of - the JDK installation on your system. You can do this by adding two - lines to your personal - ~/.bashrc - configuration file. Open - ~/.bashrc - (or create it if it doesn't exist) and add a line similar to - the following one anywhere inside the file: - - - export JAVA_HOME="/usr/lib/jvm/jdk1.5.0_<version>" - - - You should also set this environment variable for any other users - who will be running - &PLATFORM_NAME; - - (any environment variables - export - ed from - ~/.bashrc - files are local to that user). - - - - - - Setting - java - , - javac - and - java_sdk_1.5.0 - Using the - alternatives - command - - - - - Selecting the Correct System JVM on Linux using - <command>alternatives</command> - - - On systems with the - alternatives - command, including Red Hat Enterprise Linux and Fedora, you can - easily choose which JDK (or JRE) installation you wish to use, as - well as which - java - and - javac - executables should be run when called. - - - - As the root user - , call - /usr/sbin/alternatives - with the - - option to select between JDKs and JREs installed on your system: - - - root@localhost ~]$ /usr/sbin/alternatives --config java - -There are 3 programs which provide 'java'. - - Selection Command ------------------------------------------------ - 1 /usr/lib/jvm/jre-1.5.0-gcj/bin/java - 2 /usr/lib/jvm/jre-1.6.0-sun/bin/java -*+ 3 /usr/lib/jvm/jre-1.5.0-sun/bin/java - - -Enter to keep the current selection[+], or type selection number: - - - In our case, we want to use the Sun JDK, version 5, that we - downloaded and installed, to run the - java - executable. In the - alternatives - information printout above, a plus ( - + - ) next to a number indicates the one currently being used. As per - alternatives - ' instructions, pressing - Enter - will simply keep the current JVM, or you can enter the number - corresponding to the JVM you would prefer to use. - - - Repeat the procedure above for the - javac - command and the - java_sdk_1.5.0 - environment variable, - as the root user - : - - - ~]$ /usr/sbin/alternatives --config javac - - ~]$ /usr/sbin/alternatives --config java_sdk_1.5.0 - - - - - Setting the - JAVA_HOME - Environment Variable on Windows - - - - For information on how to set environment variables in Windows, - refer to - - . - - - - - - - Testing - - Finally, to make sure that you are using the correct JDK or Java - version (5 or higher), and that the java executable is in your - PATH - , run the - - java - - - command in the terminal from your home directory: - - - - ~]$ java -version -java version "1.5.0_16" -Java(TM) 2 Runtime Environment, Standard Edition (build 1.5.0_16-b03) -Java HotSpot(TM) Client VM (build 1.5.0_16-b03, mixed mode, sharing) - - - - Uninstalling - - There is usually no reason (other than space concerns) to remove a - particular JDK from your system, given that you can switch between - JDKs and JREs easily using - alternatives - , and/or by setting - JAVA_HOME - . - - - - Uninstalling the JDK on Linux - - On RPM-based systems, you can uninstall the JDK using the - - yum - - - command. - - - - Uninstalling the JDK on Windows - - On Windows systems, check the JDK entry in the - Start - menu for an uninstall command, or use - Add/Remove Programs - . - - -
- - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Common_Content/Preface.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Common_Content/Preface.xml deleted file mode 100644 index 7bbe6b2560..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Common_Content/Preface.xml +++ /dev/null @@ -1,12 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - Preface - - - - - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Common_Content/Setting_the_JBOSS_HOME_Environment_Variable.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Common_Content/Setting_the_JBOSS_HOME_Environment_Variable.xml deleted file mode 100644 index 7b9ff427ad..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Common_Content/Setting_the_JBOSS_HOME_Environment_Variable.xml +++ /dev/null @@ -1,353 +0,0 @@ - - -%BOOK_ENTITIES; - -]> -
- Setting the JBOSS_HOME Environment Variable - - The - &PLATFORM_NAME; - Platform - - ( - &PLATFORM_NAME; - - ) is built on top of the - JBoss Application Server - ( - JBoss AS - ). You do not need to set the - JBOSS_HOME - environment variable to run any of the - &PLATFORM_NAME; - Platform - - servers - unless - JBOSS_HOME - is - already - set. - - - The best way to know for sure whether - JBOSS_HOME - was set previously or not is to perform a simple check which may save - you time and frustration. - - - Checking to See If JBOSS_HOME is Set on Unix - - At the command line, - echo - $JBOSS_HOME - to see if it is currently defined in your environment: - - - - ~]$ echo $JBOSS_HOME - - The - &PLATFORM_NAME; - Platform - - and most &PLATFORM_NAME; - servers are built on top of the - JBoss Application Server - ( - JBoss AS - ). When the - &PLATFORM_NAME; - Platform - - or &PLATFORM_NAME; - servers are built - from source - , then - JBOSS_HOME - must - be set, because the &PLATFORM_NAME; - files are installed into (or - over top of - if you prefer) a clean - JBoss AS - installation, and the build process assumes that the location pointed - to by the - JBOSS_HOME - environment variable at the time of building is the - JBoss AS - installation into which you want it to install the &PLATFORM_NAME; - files. - - - This guide does not detail building the - &PLATFORM_NAME; - Platform - - or any &PLATFORM_NAME; - servers from source. It is nevertheless useful to understand the role - played by - JBoss AS - and - JBOSS_HOME - in the &PLATFORM_NAME; - ecosystem. - - - The immediately-following section considers whether you need to set - JBOSS_HOME - at all and, if so, when. The subsequent sections detail how to set - JBOSS_HOME - on Unix and Windows - - - - Even if you fall into the category below of - not needing - to set - JBOSS_HOME - , you may want to for various reasons anyway. Also, even if you are - instructed that you do - not need - to set - JBOSS_HOME - , it is good practice nonetheless to check and make sure that - JBOSS_HOME - actually - isn't - set or defined on your system for some reason. This can save you both - time and frustration. - - - - You - DO NOT NEED - to set - JBOSS_HOME - if... - - - - - ...you have installed the - &PLATFORM_NAME; - Platform - - binary distribution. - - - - - ...you have installed a &PLATFORM_NAME; - server binary distribution - - which bundles - JBoss AS - . - - - - - - You - MUST - set - JBOSS_HOME - if... - - - - - ...you are installing the - &PLATFORM_NAME; - Platform - - or any of the &PLATFORM_NAME; - servers - from source - . - - - - - ...you are installing the - &PLATFORM_NAME; - Platform - - binary distribution, or one of the &PLATFORM_NAME; - server binary distributions, which - do not - bundle - JBoss AS - . - - - - - Naturally, if you installed the - &PLATFORM_NAME; - Platform - - or one of the &PLATFORM_NAME; - server binary releases which - do not - bundle - JBoss AS - , yet requires it to run, then you should - - install - JBoss AS - - before setting - JBOSS_HOME - or proceeding with anything else. - - - Setting the JBOSS_HOME Environment Variable on Unix - - The - JBOSS_HOME - environment variable must point to the directory which contains all - of the files for the - - &PLATFORM_NAME; - Platform - - or individual &PLATFORM_NAME; - server - - that you installed. As another hint, this topmost directory contains - a - bin - subdirectory. - - - - Setting - JBOSS_HOME - in your personal - ~/.bashrc - startup script carries the advantage of retaining effect over reboots. - Each time you log in, the environment variable is sure to be set for - you, as a user. On Unix, it is possible to set - JBOSS_HOME - as a system-wide environment variable, by defining it in - /etc/bashrc - , but this method is neither recommended nor detailed in these - instructions. - - - To Set JBOSS_HOME on Unix... - - - Open the - ~/.bashrc - startup script, which is a hidden file in your home directory, in a - text editor, and insert the following line on its own line while - substituting for the actual install location on your system: - - - export JBOSS_HOME="/home/<username>/<path>/<to>/<install_directory>" - - - - Save and close the - .bashrc - startup script. - - - - - You should - source - the - .bashrc - script to force your change to take effect, so that - JBOSS_HOME - becomes set for the current session - - - Note that any other terminals which were opened prior to your - having altered - .bashrc - will need to - source - ~/.bashrc - as well should they require access to - JBOSS_HOME - . - - - . - - - ~]$ source ~/.bashrc - - - - Finally, ensure that - JBOSS_HOME - is set in the current session, and actually points to the correct - location: - - - - The command line usage below is based upon a binary installation of - the - &PLATFORM_NAME; - Platform - - . In this sample output, - JBOSS_HOME - has been set correctly to the - topmost_directory - of the - &PLATFORM_NAME; - - installation. Note that if you are installing one of the standalone - &PLATFORM_NAME; - - servers (with - JBoss AS - bundled!), then - JBOSS_HOME - would point to the - topmost_directory - of your server installation. - - - - ~]$ echo $JBOSS_HOME -/home/silas/&HOME_LIN; - - - - - Setting the JBOSS_HOME Environment Variable on Windows - - The - JBOSS_HOME - environment variable must point to the directory which contains all - of the files for the - &PLATFORM_NAME; - Platform or individual &PLATFORM_NAME; - server - - that you installed. As another hint, this topmost directory contains - a - bin - subdirectory. - - - - For information on how to set environment variables in recent versions - of Windows, refer to - - . - -
- - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Preface.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Preface.xml deleted file mode 100644 index 88c4ed5563..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Preface.xml +++ /dev/null @@ -1,12 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - Preface - - - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/RestComm_User_Guide.ent b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/RestComm_User_Guide.ent deleted file mode 100644 index feb86c9866..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/RestComm_User_Guide.ent +++ /dev/null @@ -1,19 +0,0 @@ - - - - - - - - - - - - - - - - - - - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/RestComm_User_Guide.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/RestComm_User_Guide.xml deleted file mode 100644 index 9c5645fb24..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/RestComm_User_Guide.xml +++ /dev/null @@ -1,28 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - - - - - - - - - - - - - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Revision_History.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Revision_History.xml deleted file mode 100644 index 8df7b8d7e4..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/Revision_History.xml +++ /dev/null @@ -1,74 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - Revision History - - - - - 2.1 - Tuesday 27 2014 - - Charles - Roufay - charles.roufay@telestax.com - - - - Update doc with RestComm USSD Integration and new RVD features - - - - - - 2.0 - Friday Sep 2013 - - Charles - Roufay - charles.roufay@telestax.com - - - - Update doc with RestComm JBoss AS7 and Advanced Examples - - - - - 1.0 - Mon Apr 18 2011 - - George - Vagenas - gvagenas@gmail.com - - - - First release CDI-Telco-Framework user guide - - - - - - 1.0 - Mon Apr 18 2011 - - George - Vagenas - gvagenas@gmail.com - - - - Changes to reflect changes made regarding JBoss AS5 and - the refactoring made to the project. - - - - - - - - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Admin-UI.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Admin-UI.xml deleted file mode 100644 index b0db553960..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Admin-UI.xml +++ /dev/null @@ -1,462 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - Restcomm Admin User Interface - -The admin user interface is a great way to perform repetive Restcomm task in a orderly manner. Care has been taken to make the user experience intuitive for those new to the platform. In this chapter, you will learn how to use some of the features available to create Clients, Phone Numbers, Check Call Logs, Get Speech-to-Text Transcription and many more - - -Please also visit the official Telestax documentation page for more usage tutorials:http://www.telestax.com/docs - - -
-Login Interface - -You need to make sure Restcomm is running before you can use the Admin User Interface. When you are running on a local install, open your web browser and go to this url - http://127.0.0.1:8080/restcomm-management . You will see a screenshot similar to the one below - - - - -Please see the for more information about changing the default password. - - - - -
- -
-Passwords, Sub Accounts Settings - -When you click on the Administrators Account at the top right corner of the window, as shown in the screenshot below, you will be able to change your Administrators passwords and create Sub Accounts. - - - - - - - - - - Account Settings - - - - - - Number - Description - - - - - 1 - Shows the current Acount profile - - - 2 - You can create new Sub Accounts - - - 3 - Descriptive name of the user account - - - 4 - New password of the current logged in account - - - 5 - You can activate or suspend an account - - - 6 - There are three options, Trial, Basic and Full(default) - - - 7 - List of Sub Accounts - - - -
- - -
- - - -
-Dashboard - -This is the default page where you can get an overview of your Restcomm Installation. - - - - - - - - - Dashboard - - - - - - Number - Description - - - - - 1 - Account SID is the default used by the Administrator's account - - - 2 - The Auth Token is the password that is required for any Restcomm operation. You can click on the hidden button to reveal the hashed password. - - - 3 - The Debugger lets you troubleshoot any issues you might encounter using Restcomm - - - 4 - A quick way to get all the API exposed by Restcomm - - - 5 - Additional information about Calls parsed by Restcomm - - - 6 - Additional information about SMS parsed by Restcomm - - - - 7 - This will take you to the Restcomm Visual Designer (RVD) page located a - - - - -
- - -
- -
-Restcomm Numbers - -This will show the default demo applications that come with Restcomm. When you start creating applications and attaching numbers, they will also be displayed on this page. - - - - - - - - - Restcomm Numbers - - - - - - Number - Description - - - - - 1 - The number 1235 is attached to the hello-world.xml application. You must have configured VoiceRSS text-to-speech to use this application - - - 2 - The number 1236 is attached to a the Gather verb. It will ask you to enter a number and you can hear the number you enter. You must have configured VoiceRSS text-to-speech to use this application - - - 3 - The number 1234 plays a pre-recorded file. - - - 4 - This icon lets you edit the Name of the entry to a more descriptive one. - - - 5 - This button lets you create a new number that can be attached to a RCML - - - - - -
- - - -
-Register Number - - -When you click on the Register Number button, you will see a screenshot similar to the one below. This will allow you to create a new phone number that can be attached to a Restcomm application. - - - - - - - - - Restcomm Numbers - - - - - - Number - Description - - - - - 1 - The field to enter the phone number. - - - 2 - This button will show the Area Code. You must have Voip Innovations API account configured in the restcomm.xml file to use this feature. It will search for DID in the area code specified. At the moment, only US DIDs are available. - - - - 3 - This button will show advanced options if you want to add more features to the phone number like the VoiceUrl - - - 4 - See the Rest API for more information - - - 5 - This button will show available Restcomm Visual Designer (RVD) applications that can be linked to a phone number. - - - 6 - See the Rest API for more information - - - - - - -
-
-
-Edit Update Number - - -Editing a phone number can be done by clicking on the number, the screenshot below shows how you can edit the number 1235. You can change the VoiceUrl to which the number is attached. - - - - - - - - - Edit Update Numbers - - - - - - Number - Description - - - - - 1 - You can link the phone number to a VoiceUrl application. See the REST API for more details. - - - 2 - You can link the phone number to a SMS application. See the REST API for more details - - - 3 - You can link the phone number to a USSD application. See the REST API for more details - - - 4 - Caller Id lookup requires a CNAM provier - - - 5 - You can save your changes or press close to discard the changes - - - - - -
- -
-
- - -
-RestComm Clients - -RestComm Client is a feature that allows you to create a Restcomm profile that can be used for P2P or B2BUA calls. This will be empty until you create a new client. You can create a new client by clicking on the Resgister RestComm Clent button. - - - - - - - - - RestComm Clients - - - - - - Number - Description - - - - - 1 - The client name. (ex. alice or bob) - - - 2 - The password that will be used to when you want to register the client with restcomm - - - 3 - Use to open optional parameters windows - - - 4 - This can be the full name of the client (ex.Alice Wilkinson) or any descriptive name - - - 5 - This is where you specify the VoiceUrl that is automatically invoked when the client is called. See the REST API for more details. - - - 6 - See the REST API for more details. - - - 7 - This will validate your changes and create the client. - - - - -
- - - -
-
-Outgoing CallerId - -Will be available in future release - -
- -
-Logs - -The log section gives you an overview of current Restcomm system information. - - - -
-Logs - Calls - - - -A list of all calls that have been processed by Restcomm - - - - - - - -
- -
-Logs - Messages - - - -A list of all SMS messages that have been processed by Restcomm - - - - - - - - -
- -
-Logs - Recordings - - -A list of all Recordings (using the Record Verb) that have been processed by Restcomm - - - - - - - -
- -
-Logs - Transcriptions - - -A list of all Transcriptions that have (using the Transcribe parameters of the Record Verb) that have been processed by Restcomm. See the - - - - - - - -
- -
-Logs - Notifications - - -A list of all Notifications received by Restcomm. - - - - - - - -
- -
- - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Advanced_Examples.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Advanced_Examples.xml deleted file mode 100644 index b8ea2abc32..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Advanced_Examples.xml +++ /dev/null @@ -1,317 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - -Advanced RestComm Examples - - -In this chapter, you will learn how to use more advanced features of Restcomm verbs to create communications applications - - -
-Activating Text-to-Speech (TTS) - - -You must get an API key from http://www.voicerss.org in order to proceed with this section. You can register for a free account and an API key will be emailed to you. Once you have the API key, open the $RESTCOMM_HOME/standalone/deployments/restcomm.war/WEB-INF/conf/restcomm.xml file and find the speech-synthesizer VoiceRSS section. Add your API key as shown below and restart RestComm - - - - - http://api.voicerss.org - 2901c0aXXXXXXXXXXXXXX -]]> - - - - -Once you have done that, you are ready to use Text-to-Speech feature of RestComm. - - -
-Say Verb -Create a file called test.xml in the $RESTCOMM_HOME/standalone/deployments/restcomm.war/demos/ directory and copy the content of the application below into the test.xml file and save it - - - - - - Welcome to RestComm, you have successfully tested the Say Verb - -]]> - - - -From a command terminal run the Curl command below to bind the 5555 phone number to the test.xml file. - - - - - -If the command is successful, you will see an output similar to the following: - - - - -Next, you have to configure your SIP phone and make a call to the test.xml dialing number 5555 - - -Known Issues with some SIP Phones - -Some SIP phones have been known to have codec problems and do not correctly render the application as desired. In this example, we shall be using Ekiga. - - - - -Start the SIP phone Ekiga (see below) and dial 5555@127.0.0.1:5080. You will hear the content of the Say Verb in the test.xml file. - - - - - - - - -
- -
-Record Verb -Create a file called test.xml in the $RESTCOMM_HOME/standalone/deployments/restcomm.war/demos/ directory and copy the content of the application below into the test.xml file and save it. Dial the number 5555 and when you hear the beep sound, leave a message and hangup when you are finished. - - - - - - Welcome to RestComm, leave a message after the beep - - -]]> - - - - -A wave file will be recorded and saved in the $RESTCOMM_HOME/standalone/deployments/restcomm.war/recordings/ directory. You can use any media player to listen to the recorded voice message. - -
- -
-Dial Verb and Client - -In order to use the Dial verb, you will need two users and register them to RestComm. - -Restcomm has already two users created for you for testing purposes, alice and bob - - -Alice Restcomm client -Username = alice / Password = 1234 - - - -Bob Restcomm client -Username = bob / Password = 1234 - - - - - -Register Alice and Bob using softphone Sflphone - - - - -In order to start two instances of Sflphone on the same computer, you need to start the second instance using sudo Sflphone. - - - -When Sflphone is started, go to the the menu Edit->Accounts, then press the Add button - - - -Fill out the configuration for Bob as shown in the screenshot below: - - -Sflphone Basic Configuration - - - - - - - - - -Sflphone Advanced Configuration -Make sure the port number is set to 5061 - - - - - - - - - -In the second instance of Sflphone, register user Alice following the same procedure used for Bob. In the Advanced settings, make sure the port number for user Alice is set to 5060 - - - -Copy the content of the application below into the test.xml file and save. - - - - - - Welcome to RestComm, you are being forwarded to Alice - - alice - - -]]> - -From the Sflphone registered with user Bob, make a call to Alice. Enter the name alice and make the call - -
- -
-Dial Verb and Uri - -Copy the Dial Uri application below to the test.xml. Remember that the port number for user Alice is 5060 and that is the port that is used in the application. If you want to use a different port, you also have to change the SIP Uri to reflect that. You can then make a call from user Bob to Alice. - - - - - Welcome to RestCom, you are being forward to Alice - - sip:alice@127.0.0.1:5060 - - - -]]> - -
- -
-Dial Verb and Conference - -Copy the Dial Conference application below to the test.xml and save the file. You can dial any number like 4321 and RestComm will read the Say verb and make a beep sound when you join the conference room. You can use another SIP phone to join the same conference. It works better if you are on a different computer as this reduces the echo generated from the microphones. - - - - - Welcome to RestCom, you are now joining the conference room - - test-conference - - -]]> - -
- -
-Gather Verb - -This verb is used to get user input and instruct RestComm to perform a specific task. This example is a little bit more elaborate and it will require the creation of a php file. You also must host the php file on a web server like Apache. Copy the Gather application below into the test.xml and save the file. - - - - - - Welcome to Telestax RestCom. - For opening hours, press 1. - to talk to Alice, press 2. - - - Sorry, I didn't get your response. - http://127.0.0.1:8080/restcomm/demos/test.xml - -]]> - - - -Create a php file in the Apache /var/www/html directory - -You can use any web server of your choice, in this example, we shall be using Apache on a Linux computer -Start the httpd server as follows sudo service httpd start -Create a new file called test-user-input.php in the /var/www/html directory -Copy the php application below into the test-user-input.php and save - - - -'; - - echo ''; - - $user_input = (int) ($_POST['Digits']); - - if ($user_input == 1) - - { - echo 'Our Opening hours are 24 hours 7 Days a week '; - echo 'Telestax appreciates your business'; - echo 'You may Hang up or wait to be redirected to the main menu'; - echo 'http://127.0.0.1:8080/restcomm/demos/test.xml'; - - } elseif($user_input == 2) { - - echo 'You are being forwarded to Alice'; - echo ''; - echo 'alice'; - echo ''; - - } - echo ''; -?> -]]> - - -This example welcomes the user and offers two options. If the user presses 1, he hears the openining hours message. If the user presses 2, he will be redirected to user Alice. - -To test the application using the Gather verb, dial any number from user Bob and follow the application instruction. - - -Using the Fax, SMS and other features of RestComm - -Please note that Telestax also offers TelScale RestComm hosted on Amazon AWS. If you want to learn more about how to use DID provisioning and more, please visit www.telestax.com - - -
- -
- - - - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Advantages.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Advantages.xml deleted file mode 100644 index bc6f75951d..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Advantages.xml +++ /dev/null @@ -1,21 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - Advantages of the RestComm - Loose coupling and reusability of SIP Components across different - applications are the most obvious advantages over the current SIP - Servlets 1.1 specification. In addition to that, you get better - integration with Java EE 6. - - In the future we will try to explore further usage of this - component-based approach to allow application developers to compose - their applications in a new way and benefits from Modularization - within a single application which should give additional performance - improvements over SIP Servlets Application Routing and Chaining while - you can still use the latter. - - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Conclusion.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Conclusion.xml deleted file mode 100644 index a6c6c703c3..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Conclusion.xml +++ /dev/null @@ -1,18 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - Conclusion - CTF provides an extensive set of features and advantages to the - developer but we believe there are improvements to be done. Developers - hold a vital role for the further improvement of the framework. - Please give it a try and provide early feedback ! - This is a perfect time to offer any ideas for or contribute Telco - modules/extensions that you would find useful in your applications. - - Leave us your comments on the discussion started at mobicents - google group - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Configuration.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Configuration.xml deleted file mode 100644 index d0254809b4..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Configuration.xml +++ /dev/null @@ -1,1577 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - -Getting Started and General Configuration - -
-Quick Start Guide - -RestComm is a robust, powerful platform that will facilitate building comprehensive real-time communication solutions. The steps below will help you get started with ease. - - - - -On a production system, please take into consideration the following: - - -MMS can get into a bad state and report "Too many open files" error because of Linux default value on the total number of file descriptors. - - -You can go to the following sites to see how to increase the maximum number of open files in your server: - - - - - - - - - - - - - -Running RestComm on JBoss - -JBoss Security - -Running a secure application is dependent on multiple factors. Restcomm runs on JBoss which implies that system security implemention can be handled at JBoss level. Please see the link below on how you can make your server environment more secure. - - - - - -Download TelScale-Restcomm-JBoss-AS7-XX.XX.GA.zip - - - - -Using a terminal of your choice, extract the content of TelScale-Restcomm-XX.XX.GA.zip to a local directory on your computer. The root directoy into which you extract the content of the .zip file will be referred to as $RESTCOMM_HOME. - - - - -Go to $RESTCOMM_HOME/telscale-media/telscale-media-server/bin - - - - -change the permission of the run.sh as follows - - - - -sudo chmod +x ./run.sh - - - - -then start the TelScale Media Server as follows - - - - -sudo ./run.sh - - - - -If all is correctly started you will see the following at the end of the bash terminal window - -[MainDeployer] [[[[[[[[[ Mobicents Media Server: release.version=3.0.0.FINAL Started ]]]]]]]]] - - - - - - -Open another terminal and proceed as follows: - - - - - -Go to $RESTCOMM_HOME/bin - - - - -change the permissions of all the .sh files in the bin directory as follows - - - - -sudo chmod +x ./*.sh - - - - -Start RestComm by running the following command - - - - -sudo ./standalone.sh -c standalone-sip.xml - - - - - -If RestComm is correctly started you will see the following at the end of the terminal - - - -INFO [Version] TelScale Sip Servlets 6.1.3.GA-TelScale (build: Git Hash=r8947f2732ee64c76566ed6c0b236204c048538e1 date=201306131639) Started. -17:30:05,854 INFO [Version] -============================================================================== -== == -== Thank you for running TelScale == -== Carrier Grade Communications Platform by the creators of Mobicents == -== Copyright 2011-2013 Telestax, Inc. == -== http://www.telestax.com/ == -== == -============================================================================== - - - - - - - - - -Running RestComm on Tomcat - - - -Download TelScale-Restcomm-Tomcat-XX.XX.GA.zip - - - - -Using a terminal of your choice, extract the content of TelScale-Restcomm-Tomcat-XX.XX.GA.zip to a local directory on your computer. The root directoy into which you extract the content of the .zip file will be referred to as $RESTCOMM_HOME. - - - - -Go to $RESTCOMM_HOME/telscale-media/telscale-media-server/bin - - - - -change the permission of the run.sh as follows - - - - -sudo chmod +x ./run.sh - - - - -then start the TelScale Media Server as follows - - - - -sudo ./run.sh - - - - -If all is correctly started you will see the following at the end of the bash terminal window - -[MainDeployer] [[[[[[[[[ Mobicents Media Server: release.version=3.0.0.FINAL Started ]]]]]]]]] - - - - - - -Open another terminal and proceed as follows: - - - - - -Go to $RESTCOMM_HOME/bin - - - - -change the permissions of all the .sh files in the bin directory as follows - - - - -sudo chmod +x ./*.sh - - - -The content of the bin directory should be similar to the list below - -bootstrap.jar digest.sh tomcat-juli.jar -catalina.bat setclasspath.bat tomcat-native.tar.gz -catalina.sh setclasspath.sh tool-wrapper.bat -catalina-tasks.xml shutdown.bat tool-wrapper.sh -commons-daemon.jar shutdown.sh version.bat -commons-daemon-native.tar.gz startup.bat version.sh -cpappend.bat startup.sh -digest.bat telestax-license.xml - - - -Start RestComm by running the following command - - - - - sudo ./catalina.sh run - - - -If RestComm is correctly started on Tomcat, you will see an output similar to the one below - -Started. -2013-08-15 19:02:46,903 INFO [Version] (main) -============================================================================== -== == -== Thank you for running TelScale == -== Carrier Grade Communications Platform by the creators of Mobicents == -== Copyright 2011-2013 Telestax, Inc. == -== http://www.telestax.com/ == -== == -============================================================================== -Aug 15, 2013 7:02:50 PM org.apache.catalina.startup.Catalina start -INFO: Server startup in 14090 ms - - - - - - - - - - - - - - -
- -
-Login into Restcomm and Changing the Default Password - -Before you start using Restcomm, you must change the default password. Please follow the steps below - - - - -Open your browser and go to http://127.0.0.1:8080/restcomm-management - -Enter the following information: - - Email/Account SID = administrator@company.com - - Password = RestComm - - Press the Login button - - - - -You will see a screen similar to the one below - - - - - - - - - - - -In the screen below, change the password and update. - - - - - - - - - - -Once the password has been changed, you can now log into the Admin Management interface and start using Restcomm. - - -
- - -
-Testing the Demo Applications - - - -Restcomm comes prepackaged with multiple example applications designed to help you quickly get started. For more demos and configuration details please visit www.telestax.com/doc - - - - - -
-Demo 1 - Testing the Play Verb - - -Start a SIP phone (see below) and dial 1234@127.0.0.1:5080. You will hear a welcome message. - - - - - - - - -Some SIP phones have codec incompatibility issues, in the above example, we used Ekiga. You may also try Jitsi or Sflphone. - - - - - Getting Started and General Configuration - -
- Quick Start Guide - - RestComm is a robust, powerful platform that will facilitate building - comprehensive real-time communication solutions. The steps below will - help you get started with ease. - - - - - On a production system, please take into consideration the following: - - - MMS can get into a bad state and report "Too many open files" error - because of Linux default value on the total number of file - descriptors. - - - You can go to the following sites to see how to increase the maximum - number of open files in your server: - - - - - - - - - - - - - - Running RestComm on JBoss - - JBoss Security - - Running a secure application is dependent on multiple factors. - Restcomm runs on JBoss which implies that system security - implemention can be handled at JBoss level. Please see the link - below on how you can make your server environment more secure. - - - - - - - Download - TelScale-Restcomm-JBoss-AS7-XX.XX.GA.zip - - - - - Using a terminal of your choice, extract the content of - TelScale-Restcomm-XX.XX.GA.zip - to a local directory on your computer. The root directoy into which - you extract the content of the .zip file will be referred to as - $RESTCOMM_HOME. - - - - - Go to - $RESTCOMM_HOME/telscale-media/telscale-media-server/bin - - - - - change the permission of the - run.sh - as follows - - - - - sudo chmod +x ./run.sh - - - - - then start the TelScale Media Server as follows - - - - - sudo ./run.sh - - - - - If all is correctly started you will see the following at the end - of the bash terminal window - - [MainDeployer] [[[[[[[[[ Mobicents Media Server: release.version=3.0.0.FINAL - Started ]]]]]]]]] - - - - - - - Open another terminal and proceed as follows: - - - - - - Go to - $RESTCOMM_HOME/bin - - - - - change the permissions of all the - .sh - files in the bin directory as follows - - - - - sudo chmod +x ./*.sh - - - - - Start RestComm by running the following command - - - - - sudo ./standalone.sh -c standalone-sip.xml - - - - - - If RestComm is correctly started you will see the following at the - end of the terminal - - - - INFO [Version] TelScale Sip Servlets 6.1.3.GA-TelScale (build: Git - Hash=r8947f2732ee64c76566ed6c0b236204c048538e1 date=201306131639) - Started. - 17:30:05,854 INFO [Version] - ============================================================================== - == == - == Thank you for running TelScale == - == Carrier Grade Communications Platform by the creators of Mobicents - == - == Copyright 2011-2013 Telestax, Inc. == - == http://www.telestax.com/ == - == == - ============================================================================== - - - - - - - - - - Running RestComm on Tomcat - - - - Download - TelScale-Restcomm-Tomcat-XX.XX.GA.zip - - - - - Using a terminal of your choice, extract the content of - TelScale-Restcomm-Tomcat-XX.XX.GA.zip - to a local directory on your computer. The root directoy into which - you extract the content of the .zip file will be referred to as - $RESTCOMM_HOME. - - - - - Go to - $RESTCOMM_HOME/telscale-media/telscale-media-server/bin - - - - - change the permission of the - run.sh - as follows - - - - - sudo chmod +x ./run.sh - - - - - then start the TelScale Media Server as follows - - - - - sudo ./run.sh - - - - - If all is correctly started you will see the following at the end - of the bash terminal window - - [MainDeployer] [[[[[[[[[ Mobicents Media Server: release.version=3.0.0.FINAL - Started ]]]]]]]]] - - - - - - - Open another terminal and proceed as follows: - - - - - - Go to - $RESTCOMM_HOME/bin - - - - - change the permissions of all the - .sh - files in the bin directory as follows - - - - - sudo chmod +x ./*.sh - - - - The content of the bin directory should be similar to the list - below - - bootstrap.jar digest.sh tomcat-juli.jar - catalina.bat setclasspath.bat tomcat-native.tar.gz - catalina.sh setclasspath.sh tool-wrapper.bat - catalina-tasks.xml shutdown.bat tool-wrapper.sh - commons-daemon.jar shutdown.sh version.bat - commons-daemon-native.tar.gz startup.bat version.sh - cpappend.bat startup.sh - digest.bat telestax-license.xml - - - - Start RestComm by running the following command - - - - - sudo ./catalina.sh run - - - - If RestComm is correctly started on Tomcat, you will see an - output similar to the one below - - Started. - 2013-08-15 19:02:46,903 INFO [Version] (main) - ============================================================================== - == == - == Thank you for running TelScale == - == Carrier Grade Communications Platform by the creators of Mobicents - == - == Copyright 2011-2013 Telestax, Inc. == - == http://www.telestax.com/ == - == == - ============================================================================== - Aug 15, 2013 7:02:50 PM org.apache.catalina.startup.Catalina start - INFO: Server startup in 14090 ms - - - - - - - - - - - - - - -
-
- Testing the Demo Applications - - - - Restcomm comes prepackaged with multiple example applications designed to - help you quickly get started. - - -
- Demo 1 - Testing the Play Verb - - - Start a SIP phone (see below) and dial - 1234@127.0.0.1:5080 - . You will hear a welcome message. - - - - - - - - - Some SIP phones have codec incompatibility issues, in the above example, - we used Ekiga. You may also try Jitsi or Sflphone. - - - - -$RESTCOMM_HOME/standalone/deployments/restcomm.war/demos/hello-play.xml. -]]> - - - - -
-
- Demo 2 - Testing Say Verb - - - You must first activate Text-to-Speech before you can proceed. - - - You must get an API key from - http://www.voicerss.org - in order to proceed with this section. You can register for a free - account and an API key will be emailed to you. Once you have the API - key, open the - $RESTCOMM_HOME/standalone/deployments/restcomm.war/WEB-INF/conf/restcomm.xml - file and find the speech-synthesizer VoiceRSS section. Add your API - key as shown below and restart RestComm - - - - - http://api.voicerss.org - 2901c0aXXXXXXXXXXXXXX -]]> - - - - Start a SIP phone dial - 1235@127.0.0.1:5080 - . You will hear a welcome message in multiple languages. - - - - - - - - - -
- -
- Demo 3 - Testing Gather Verb - This demo creates a simple IVR system - - - - Activate DTMF using Ekiga - Make sure your DTMF setting in Ekiga is set to RFC2833. In - order to set it correctly, go to the menu - Edit->Preference->Protocols->SIP Settings->DTMF Mode - You may also use SFLPHONE instead of Ekiga - - - - - Start a SIP phone dial - 1236@127.0.0.1:5080 - . You will hear a message asking you to enter a digit and press - star. If the digit is correctly received, Restcomm will replay the - number you entered. - - - - -$RESTCOMM_HOME/standalone/deployments/restcomm.war/demos/gather/hello-gather.xml. -and -$RESTCOMM_HOME/standalone/deployments/restcomm.war/demos/gather/gather.jsp. -]]> - - -
- -
- Demo 4 - Testing the Dial Sip Noun - - This demo makes a call from one SIP phone to another. The Demo uses - the SIP noun. You can calll any SIP account. All you have to do is - change the content of the - dial-sip.xml - - - In order to use this demo, you may use the default accounts, Alice and - Bob, and register them on two separate SIP phones. Start both SIP - phones and make sure Alice and Bob are registerd with the password - (1234). These users come pre-configured with Restcomm for test - purposes. - - - - Start Two SIP Phone Sessions - If you are using the SIP sflphone here is what to do: - Start first instance ex. - sflphone - Start second instance ex. - sudo sflphone - - - - In the application - dial-sip.xml - you can change the default to - sip:alice@127.0.0.1:5061? - - - This will allow you to make a call to Alice. Note that Alice must be - registered on port 5061 for the call to succeed. - - - - From the the phone on which Bob is registerd, dial the number - 1237 - . The phone on which Alice is registered will ring and the - connection will be made when you answer the call. - - - - - - - - - -
- -
- Demo 5 - Testing the Client Noun - This demo makes a call from one SIP Client to Another. The demo - uses the Client noun - - - In order to use this demo, you must have user Alice and Bob registered - on two separate SIP phones. Start both SIP phones and make sure - Alice and Bob are registerd with the password (1234). These users - come pre-configured with Restcomm for test purposes. - - - - Start Two SIP Phone Sessions - - You can start the second instance of your SIP phones by prepending - the executable with - sudo - . If you are using the SIP sflphone here is what to do: - - Start first instance ex. - sflphone - Start second instance ex. - sudo sflphone - - - - - From the phone on which Bob is registerd, dial the number - 1238 - . The phone on which Alice is registered will ring and the - connection will be made when you answer the call. - - - - -$RESTCOMM_HOME/standalone/deployments/restcomm.war/demos/dial/client/dial-client.xml. - -]]> - - - - - Advanced Examples - - Please go to this chapter for more examples - - . - - - -
- - -
- Demo 6 - Testing Conference Noun - - This demo Lets a user join a conference as a moderator and the other - user as a participant. The participant will dial - 1310 - and will hear a hold music. The moderator will dial - 1311 - and the hold music will stop and the conference will be started. - - - - Most SIP phones will require you to register before you can make a call. - You can use the default accounts, Alice and Bob with password - (1234)to register. - - - - Start Two SIP Phone Sessions - - You can start the second instance of your SIP phones by prepending - the executable with - sudo - . If you are using the SIP sflphone here is what to do: - - Start first instance ex. - sflphone - Start second instance ex. - sudo sflphone - - - - - From the phone on which Bob is registerd, dial the number - 1310 - . From the phone on which Alice is registered, dial - 1311 - - - - - - - - - - Advanced Examples - - Please go to this chapter for more examples - - . - - - -
- - -
-Demo 7 - RVD Say Verb - -This example is built using the Restcomm Visual Designer. You can register a SIP phone with the user bob or alice and password (1234). Then dial 1239. You will hear a welcome message. - - - - - - -
- -
-Demo 8 - RVD Gather Verb - -This demo uses the Gather verb to get user input and map it to the corresponding modules. You can register a SIP phone with the user bob or alice and password (1234). Then, dial 1240. You will hear a welcome message asking you to press 1 for Sales or 2 for Technical Support. - - - - - - -
- - -
-Demo 9 - RVD External Services - -This demo uses the External Services to Restcomm to an external program for more complex solutions. There are two parts to this demo application, the RVD app and the external servlet http://127.0.0.1:8080/RvdExternalServicesDemo/rvdESdemo which manages the logic. You can register a SIP phone with the user bob or alice and password (1234). Then, dial 1241. You will be asked to enter a customer ID. You can enter number 11 or 12. You will hear a corresponding welcome message depending on the customer ID you entered. - - - - - - -
- - -
- -
- - - - - -
- - Detailed Configuration - - - In order for TelScale RestComm to function properly it must first be - configured. In this chapter we will cover - the settings for TelScale RestComm and it's respective plug-ins. In - order to make TelScale RestComm easy to manage - all of the configuration is done in only one file. The file is located - at - RESTCOMM_HOME/WEB-INF/webapps/restcomm/conf/restcomm.xml - and is composed of the sections that follow. - - - TelScale RestComm provides a default set of plug-ins for the - storage engine, fax service, SMS aggregator, - automatic speech recognition, and speech synthesis. All these services are - pluggable and TelScale RestComm is not - limited to any service provider or software platformz. - -
-
- Runtime Settings - - The runtime-settings are used by TelScale RestComm at runtime to - customize it's behavior. A list of the runtime settings - and a description is provided below. - - - - Runtime Settings - - - - - - Element - Description - - - - - api-version - The version of the TelScale RestComm API that we will be - executing by default. - - - prompts-uri - The location where the audio prompts are located. - - - cache-path - The local path the cache folder. - - - cache-uri - The HTTP URI to the cache folder. - - - recordings-path - The local path to the folder containing the recordings. - - - recordings-uri - The HTTP URI to the folder containing the recordings. - - - error-dictionary-uri - The HTTP URI to the TelScale RestComm error dictionary. - - - external-ip - - The IP to use for out-bound SIP REGISTER requests. This is useful - when you - want to report a different IP than the one TelScale RestComm picked - by default. - - - - use-to - - If set to true TelScale RestComm will use the To header to - determine the destination. - If set to false TelScale RestComm will use the Request URI to - determine the destination. - - - - outbound-proxy-user - - The user name used to authenticate with the outbound proxy. - (Optional) - - - - outbound-proxy-password - - The password used to authenticated with the outbound proxy. - (Optional) - - - - outbound-proxy-uri - - The SIP URI to the outbound proxy. Note: Do not prepend 'sip:' to - the proxy uri. - If necessary the port can be included, for example - alice@localhost:5080 - - - - fallback-outbound-proxy-user - - The user name used to authenticate with the fallback outbound proxy. - (Optional) - - - - fallback-outbound-proxy-password - - The password used to authenticated with the fallback outbound proxy. - (Optional) - - - - fallback-outbound-proxy-uri - - The SIP URI to the fallback outbound proxy. Note: Do not prepend 'sip:' to - the proxy uri. - If necessary the port can be included, for example - alice@localhost:5080 - - - - allow-fallback - - Set this to true so Restcomm will fallback to the backup proxy in - case of failed calls - - - - max-failed-calls - - Maximum number of failed call before switching from primary to fallback - outbound proxy - - - - allow-fallback-to-primary - - Allow fallback from Backup to Primary proxy in case fail called at - backup proxy. - - - - -
- - TelScale Restcomm Resource Security - The security model is based on role based access control. - TelScale RestComm defines a set of permissions - that can be defined for each role. There are two predefined roles - Administrator and Developer. The Developer role - can be configured or removed all together depending on your needs but - the Administrator role can not be changed. - The Administrator role can not be modified or removed and it has - access to every resource. - - First we will define the list of permissions and what they - mean. - - TelScale RestComm Permissions - - - - - - Permission - Description - - - - - Create - The role has access to create a type resource. - - - Read - The role has access to read a type of resource. - - - Modify - The role has access to modify a type of resource. - - - Delete - The role has access to delete a type of resource. - - - -
- These permissions apply to every resource exposed by the - Restful APIs. Once a role is defined it can - be specified for new Account resources that are created. - - If no role is specified when creating a new Account resource - then the new Account will inherit - the security role of the account that created it. If this role is the - Administrator role the system - may become compromised. - - - Wildcard Permission - The asterisk '*' is a wildcard that means grant all - permissions and can be used in place of - typing out all the permissions. - - - To see how this all comes together please check out the - restcomm.xml - configuration - file. - -
-
- VoIP Innovations Restful API Access - The default auto provisioning API used by TelScale RestComm. - - - VoIP Innovations Settings - - - - - - Element - Description - - - - - login - The user name configured in the VoIP Innovations Portal - under API users. - - - password - The password configured in the VoIP Innovations Portal - under API users. - - - endpoint - The end point ID of the end point that will be used by - TelScale RestComm - - - uri - The path to the VoIP Innovations service end point. - - - -
-
-
- Dao Manager - The data access object manager is used to gain access to the - data store that TelScale RestComm will use. - - The default data access object manager is based on MyBatis and - provides access to RDBM systems. - - - MyBatis Settings - - - - - - Element - Description - - - - - configuration-file - The path to the mybatis.xml configuration file. - - - data-files - The path to the data files used to store the database - tables. - - - sql-files - The path to the XML files containing the SQL statements - that will be executed by the database. - - - -
-
-
- Media Server Manager - - - The media server manager is responsible for all the media servers - managed by TelScale RestComm. - - - Media Server Manager Settings - - - - - - Property - Description - - - - - local-address - The local IP address for the MGCP stack. - - - local-port - The local port for the MGCP stack. - - - remote-address - The IP address for the media server. - - - remote-port - The port for the media server. - - - response-timeout - - In milliseconds the maximum amount of time to wait for a response - from - the media server before abandoning the request. This does NOT apply - to - RQNT/NOTIFY request/response. - - - - external-address - - Sometimes there is interest to use a different address in the SDP than - the IP address the media server is reporting. This parameter if - set tells TelScale RestComm to - patch the Connection attribute in the SDP on behalf of the media - server to the specified IP address. - Note: TelScale RestComm will only do NAT resolution when necessary so - if your server already has a - routable IP address setting this parameter will have no effect. - - - - -
-
-
- SMS Aggregator - The SMS aggregator is responsible for the handling of SMS - messages on behalf of TelScale RestComm. - - - Requirements - - The default SMS aggregator is SMS over IP as used by VoIP - Innovations. Therefore, an account must be - created @ - http://www.voipinnovations.com - before any text - messages can be sent. - - - - Below is a list of settings that must be configured to send - text messages. - - - SMS Aggregator Settings - - - - - - Element - Description - - - - - outbound-prepend-string - - A string that you would like to prepend to the destination - address. For example, - to use service from VoIP Innovations a '#' must be prepended to - every destination - address. - - - - outbound-endpoint - - The SIP endpoint to which outbound SMS messages should be sent. - Note: Do not prepend sip: to the endpoint. - - - - -
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- Fax Service - The fax service sends faxes out on behalf of TelScale RestComm. - - - Requirements - - The default fax service is provided by Interfax. Therefore, an - account must be created @ - http://www.interfax.net - before any faxes can be - sent. - - - - - Fax Service Settings - - - - - - Element - Description - - - - - user - The user name used to authenticate with Interfax. - - - password - The password used to authenticate with Interfax. - - - -
-
-
- Speech Recognizer - This speech recognizer turns speech into text on behalf of - TelScale Restcomm. - - - Requirements - - The default speech recognizer uses the ASR service provided by - iSpeech. Therefore, an account must be created @ - http://www.ispeech.org - before any speech can be converted to text. - - - - - Speech Recognizer Settings - - - - - - Element - Description - - - - - api-key - The Web API key provided by iSpeech. This is used as an - authentication token for the service. - - - -
-
- - -
- Speech Synthesizer - The speech synthesizer turns text to speech on behalf of - TelScale RestComm. - - - Requirements - - The default speech synthesizer uses the TTS service provided by - VoiceRSS. Therefore, an account must be created @ - http://www.voicerss.org - before any text can be converted to - speech. - - - - - VoiceRSS Speech Synthesizer Settings - - - - - - Element - Description - - - - - service-root - The VoiceRSS service root URI. - - - - apikey - The API key used to authenticate with the VoiceRSS. - - - languages - - A list of male and female speakers for different languages. These - can be replaced - with other alternatives provided by VoiceRSS. - - - - -
-
- - - -
- - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Introduction.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Introduction.xml deleted file mode 100644 index f9ded1aa37..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Introduction.xml +++ /dev/null @@ -1,59 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - Introduction to &SHORT_PLATFORM_NAME; - &SHORT_PLATFORM_NAME; is a turnkey Cloud Communications platform based on Open Source building blocks from the team behind Mobicents. Restcomm offers a clean room implementation of the Twilio.com APIs and much more. - - - -Restcomm can be integrated with VoIP and legacy SS7 network providers either in the cloud or via on-premise Resource Adaptors. - -
- How it Works - In order to demonstrate how &SHORT_PLATFORM_NAME; works we will go over an application that - instructs &SHORT_PLATFORM_NAME; to answer a phone call, say "Hello World" and finally hang up the - call. - - A more thorough explanation of the &SHORT_PLATFORM_NAME; Markup Language is available in Chapter 3. &SHORT_PLATFORM_NAME; - Markup Language - - In this particular example, the first thing that must happen is that TelScale RestComm receives a call. - - - - - - After TelScale RestComm can confirm that the call is destined for an application that it handles. - TelScale RestComm calls out to your application for instructions on how to handle the new incoming call. - Keep in mind that when TelScale RestComm calls out to your application it provides useful information such - as from what number the call was dialed, to what number the call is destined, caller ID information, etc. - - - - - - Once your application responds with instructions on how to handle the call, TelScale RestComm gets busy executing the - provided instruction set. - - <Response> - <Say>Hello World!</Say> - </Response> - - - - - - - In this case TelScale RestComm synthesizes the text to speech, says "Hello World!" to the caller and hangs up. - - - - - -
- Keep in mind that before you continue you should configure your ASR, Fax, SMS, and TTS plugins in order to have - everything function as you would expect it to. - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-RVD.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-RVD.xml deleted file mode 100644 index a851947de4..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-RVD.xml +++ /dev/null @@ -1,177 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - Restcomm Visual Designer - -Restcomm Visual Designer is a practical feature that lets you create applications easily using a GUI based interface. - - - -Please also visit the official Telestax documentation page for more usage tutorials:http://www.telestax.com/docs - -
-Getting Started - -In order to access the designer, open your web browser and go to this url http://127.0.0.1:8080/restcomm-rvd. This page lets you select which type of project you will like to create. There are three options, Voice, USSD and SMS. - - - - - - - - -
- -
-Voice Projects - -In the screenshot below, some demo voice projects and you are also able to create a new project. - - - - - - - - - - -In the screenshot below, you can see the main dashboard of the RVD voice project. It has a drag and drop feature. - - - - - - - - -Please also visit the official Telestax documentation page for more RVD usage tutorials:http://www.telestax.com/docs -
- - - -
-USSD Projects - - -The main page of the USSD Restcomm integration is similar to the screenshot below. You can create new projects or access existing projects from this window. - - - - - - - - - - -In the screenshot below, you can see the main dashboard of the USSD project. It has a drag and drop feature. - - - - - - - - -Please also visit the official Telestax documentation page for more RVD USSD usage tutorials:http://www.telestax.com/docs -
- - - -
-SMS Projects - - -The main page of the SMS Restcomm integration is similar to the screenshot below. You can create new projects or access existing projects from this window. - - - - - - - - - - -In the screenshot below, you can see the main dashboard of the SMS project. It has a drag and drop feature. - - - - - - - - -Please also visit the official Telestax documentation page for more RVD SMS usage tutorials:http://www.telestax.com/docs -
- - - - -
-Demo Application - -In this section, you will learn how to build a sample application. - - -Create a new project called Demo. The result will be similar to the screenshot below. - - - - - - - -Press the save button and click on the Startup Url to make sure the project is correctly saved. The result will be similar to the screenshot below. - - - - - - - -Next you need to go to the Restcomm Admin User Interface http://127.0.0.1:8080/restcomm-management/ . Login and press the menu "Numbers", then press the button, "Register Number". You will see a window similar to the screenshot below. - - - -Leave Account SID as default. Enter a "Phone Number" you want to use. In the demo we shall be using 9999. - - -Under "Voice" -> Voice Request URL, click on the List icon to reveal the Demo application you created using the Restcomm Visual Designer. See screenshot below. - - - -Press "Register" to save your work and return to the list of available phone numbers. - - - - - - - - - -You have just created a new phone number "9999" and have linked it to your Demo application. See below. - - - - - - - - - -You can then start a SIP phone to test your Demo application. Note that you must have enabled Text-to-speech for this application to work. See the Getting started chapter to learn how to configure VoiceRSS. - - - -
- - - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-RestComm_Markup_Language.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-RestComm_Markup_Language.xml deleted file mode 100644 index b286734fe0..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-RestComm_Markup_Language.xml +++ /dev/null @@ -1,1794 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - RestComm Markup Language - The RestComm Markup Language (RCML) is composed of a set of XML tags that can be used to instruct - RestComm on how to handle an on-going phone call. RestComm applications are built by combining these - XML verbs and nouns in a way that is sensible for a given set of application requirements. In this - chapter we will discuss what a request from RestComm to your application looks like as well as what the - response from your application should look like. Then, we will dive in to how each verb and noun in the - XML instruction set is used. -
- RestComm Request - - How RestComm Passes Data to Your Application - The way RestComm passes data to you application depends on the request method for the given - URI. If the request method is GET then the data is passed in the query string or the part after - the question mark. If the request method is POST then the data is sent a multi-part form data just - like when a browser submits a form. - - - RestComm Voice Request - Any time RestComm makes a request to you applications it will include the following data - as request parameters. - - - Request Parameters - - - - - - Parameter - Description - - - - - CallSid - The unique identifier for this call. - - - AccountSid - Your account id. - - - From - The phone number of the originator of the call. - - - To - The phone number of the call recipient. - - - CallStatus - The status of the call. The possible values are queued, ringing, in-progress, completed, busy, failed, and no-answer. - - - ApiVersion - The version of the RestComm API used to handle this call. - - - Direction - The direction of the call. The possible values are inbound and outbound-dial. - - - CallerName - The caller ID for the caller in the case of inbound calls. - - - -
-
-
- Your Response - In your response to the request from RestComm you want to provide RCML that will instruct RestComm on how - to handle the current call. - - MIME Types - RestComm supports the MIME types described in the table below. - - - Supported MIME Types - - - - - - Parameter - Description - - - - - text/xml, - application/xml - RestComm interprets the returned document as an XML instruction set. - - - -
- - When your application returns the RCML document the root element of the document must - always be <Response> or the parser will complain. - -
-
- Say - The <Say> verb is used to synthesize text to speech and play it to the remote party. - The voices supported depends on the TTS Service provider plug-in. Below are the voices for our default - TTS service provider plug-in which uses Acapela Voice and VoiceRSS (default) as a Service. - - Say Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - voice - man, woman - man - - - language - bf, bp, en, en-gb,cf, cs, dan, fi es, fr, de, el, - it, nl, no, pl, pt, ru, ar, ca, sv, tr - en - - - loop - integer > 1 - 1 - - - -
- - voice - The 'voice' attribute allows you to select the gender of the voice used to synthesize - the text to speech for playback. - - - language - The 'language' attribute allows you pick a specific language for speech synthesis. - RestComm currently supports languages 'bf' (Belgium-French), 'bp' (Brazilian-Portugues), 'en' (English), - 'en-gb' (British-English), 'cf' (Canadian-French), 'cs' (Czech), 'dan' (Dannish), 'fi' (Finnish), 'es' (Spanish), - 'fr' (French), 'de' (German), 'el' (Greek), 'it' (Italian), 'nl' (Netherlands-Dutch), 'no' (Norwegian), - 'pl' (Polish), 'pt' (Portuguese), 'ru' (Russian), 'ar' (Saudi-Arabia Arabic), 'ca' (Spain Catalan), - 'sv' (Swedish), and 'tr' (Turkish). - - - loop - The 'loop' attribute specifies how many times you'd like the text repeated. Specifying '0' will - cause the the <Say> verb to loop until the call is hung up. - - - Nesting - The <Say> verb can not have any other verbs or nouns nested. Only text. - - - Examples - For an example of how to use the <Say> verb see below. - - - <Response> - <Say>Hello World</Say> - </Response> - - -The example below shows how you can set the language, voice and loop parameters of the Say Verb. - - - - Bienvenue à RestComm un projet parrainé par TeleStax - - - ]]> - - - - - Usefull Information - - - - When translating text to speech, the Say verb will make assumptions about how to pronounce numbers, dates, times, amounts of money and other abbreviations. - - - - - - - When saying numbers, '12345' will be spoken as "twelve thousand three hundred forty-five." Whereas '1 2 3 4 5' will be spoken as "one two three four five." - - - - - - - - -
-
- Play - The <Play> verb is used to play an audio file to the remote party. - - Play Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - loop - integer > 1 - 1 - - - -
- - loop - The 'loop' attribute specifies how many times you'd like the audio file to be repeated. Specifying '0' will - cause the the <Play> verb to loop until the call is hung up. - - - Supported Audio Formats - - - - - - MIME type - Description - - - - - audio/wav - wav format audio - - - audio/wave - wav format audio - - - audio/x-wav - wav format audio - - - -
- -Media Server Audio File Format - -The recommended audio file format is linear( 8000,16 bit , Mono ). - - - - - Nesting - The <Play> verb can not have any other verbs or nouns nested. Only a URL. - - - - - Basic example - For an example of how to use the <Play> verb see below. - - - <Response> - <Play>http://foobar.com/demo.wav</Play> - </Response> - - - -
-
- Gather - The <Gather> verb "gathers" digits that a caller enters into his or her telephone keypad. When the caller - is done entering digits, RestComm submits that digits to the provided 'action' URL in an HTTP GET or POST request. - If no input is received before timeout, <Gather> falls through to the next verb in the RestComm document. - You may optionally nest <Say>, <Play>, and <Pause> verbs within a <Gather> verb while - waiting for input. This allows you to read menu options to the caller while letting her enter a menu selection - at any time. After the first digit is received the audio will stop playing. - - Gather Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - action - relative or absolute URL - current document URL - - - method - GET, POST - POST - - - timeout - positive integer - 5 seconds - - - finishOnKey - any digit, #, * - # - - - numDigits - integer >= 1 - unlimited - - - -
- - action - The 'action' attribute takes an absolute or relative URL as a value. When the caller has finished - entering digits RestComm will make a GET or POST request to this URL including the parameters below. - If no 'action' is provided, RestComm will by default make a POST request to the current document's URL. - - - Request Parameters - - - - - - Parameter - Description - - - - - Digits - The digits the caller pressed, excluding the finishOnKey digit. - - - -
- - method - The 'method' attribute takes the value 'GET' or 'POST'. This tells RestComm whether to request the 'action' URL - via HTTP GET or POST. - - - timeout - The 'timeout' attribute sets the limit in seconds that RestComm will wait for the caller to press another digit before - moving on and making a request to the 'action' URL. For example, if 'timeout' is '10', RestComm will wait ten seconds for - the caller to press another key before submitting the previously entered digits to the 'action' URL. RestComm waits until - completing the execution of all nested verbs before beginning the timeout period. - - - finishOnKey - The 'finishOnKey' attribute lets you choose one value that submits the received data when entered. For example, if you - set 'finishOnKey' to '#' and the user enters '1234#', RestComm will immediately stop waiting for more input when the '#' is - received and will submit "Digits=1234" to the 'action' URL. Note that the 'finishOnKey' value is not sent. The allowed - values are the digits 0-9, '#' , '*' and the empty string (set 'finishOnKey' to ''). If the empty string is used, - <Gather> captures all input and no key will end the <Gather> when pressed. In this case RestComm will submit - the entered digits to the 'action' URL only after the timeout has been reached. The value can only be a single character. - - - numDigits - The 'numDigits' attribute lets you set the number of digits you are expecting, and submits the data to the 'action' URL - once the caller enters that number of digits. - - - Nesting - You can nest the following verbs within <Gather>: <Say>, <Play>, <Pause> - - - Examples - For an example of how to use the <Gather> verb see below. - - - <Response> - <Gather action="handle-user-input.php" numDigits="1"> - <Say>Welcome to TPS.</Say> - <Say>For store hours, press 1.</Say> - <Say>To speak to an agent, press 2.</Say> - <Say>To check your package status, press 3.</Say> - </Gather> - <!-- If customer doesn't input anything, prompt and try again. --> - <Say>Sorry, I didn't get your response.</Say> - <Redirect>handle-incoming-call.xml</Redirect> - </Response> - -
-
- Record - The <Record> verb records the caller's voice and returns to you the URL of a file containing the audio recording. - - Record Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - action - relative or absolute URL - current document URL - - - method - GET, POST - POST - - - timeout - positive integer - 5 - - - finishOnKey - any digit, #, * - 1234567890*# - - - maxLength - integer greater than 1 with the number of seconds to wait - 3600 (1 hour) - - - transcribe - true, false - false - - - transcribeCallback - relative or absolute URL - none - - - playBeep - true, false - true - - - -
- - action - The 'action' attribute takes an absolute or relative URL as a value. When recording is finished - RestComm will make a GET or POST request to this URL including the parameters below. If no 'action' is - provided, <Record> will default to requesting the current document's URL. After making this request, - RestComm will continue the current call using the RCML received in your response. Any RCML verbs occuring - after a <Record> are unreachable. There is one exception: if RestComm receives an empty recording, - it will not make a request to the 'action' URL. The current call flow will continue with the next verb in - the current RCML document. - - - Request Parameters - - - - - - Parameter - Description - - - - - RecordingUrl - The URL of the recorded audio. - - - RecordingDuration - The time duration of the recorded audio. - - - Digits - The digits the caller pressed, excluding the finishOnKey digit. - - - -
- - method - The 'method' attribute takes the value 'GET' or 'POST'. This tells RestComm whether to - request the URL via HTTP GET or POST. - - - timeout - The 'timeout' attribute tells RestComm to end the recording after a number of seconds of silence has passed. - - - finishOnKey - The 'finishOnKey' attribute lets you choose a set of digits that end the recording when entered. For example, if you - set 'finishOnKey' to '#' and the caller presses '#', RestComm will immediately stop recording and submit 'RecordingUrl', - 'RecordingDuration', and the '#' as parameters in a request to the 'action' URL. The allowed values are the digits 0-9, '#' - and '*'. Unlike <Gather>, you may specify more than one character as a 'finishOnKey' value. - - - maxLength - The 'maxLength' attribute lets you set the maximum length for the recording in seconds. - - - transcribe - The 'transcribe' attribute tells RestComm that you would like a text representation of the audio of the recording. - - - transcribeCallback - The 'transcribeCallback' attribute is used in conjunction with the 'transcribe' attribute. It allows you to specify a URL - to which RestComm will make an asynchronous POST request when the transcription is complete. This is not a request for RCML - and the response will not change call flow, but the request will contain the standard RCML request parameters as well as - 'TranscriptionStatus', 'TranscriptionText', 'TranscriptionUrl' and 'RecordingUrl'. If 'transcribeCallback' is specified, - then there is no need to specify 'transcribe=true'. It is implied. If you specify 'transcribe=true' without a 'transcribeCallback', - the completed transcription will be stored for you to retrieve later (see the REST API Transcriptions section), but RestComm will - not asynchronously notify your application. - - - Request Parameters - - - - - - Parameter - Description - - - - - TranscriptionText - Contains the text of the transcription. - - - TranscriptionStatus - The status of the transcription attempt: either 'completed' or 'failed'. - - - TranscriptionUrl - The URL for the transcription's REST API resource. - - - RecordingUrl - The URL for the transcription's source recording resource. - - - -
- - playBeep - The 'playBeep' attribute allows you to toggle between playing a sound before the start of a recording. - - - Nesting - The <Record> verb can not have any other verbs or nouns nested. - - - Examples - For an example of how to use the <Record> verb see below. - - - <Response> - <Record maxLength="30"/> - </Response> - -
-
- Fax - The <Fax> verb sends a fax to some a fax machine. - - Fax Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - to - phone number - see below - - - from - phone number - see below - - - action - relative or absolute URL - none - - - method - GET, POST - POST - - - statusCallback - relative or absolute URL - none - - - -
- - to - The 'to' attribute takes a valid E.164 phone number as a value. RestComm will send a fax to this number. - When sending a fax during an incoming call, 'to' defaults to the caller. When sending an fax during an outgoing call, - 'to' defaults to the called party. The value of 'to' must be a valid phone number. - - - from - The 'from' attribute takes a valid E.164 phone number as an argument. When sending a fax during an incoming call, - 'from' defaults to the calling party. When sending a fax during an outgoing call, 'from' defaults to the called party. - The value of 'from' must be a valid phone number. - - - action - The 'action' attribute takes a URL as an argument. After processing the <Fax> verb, RestComm will make a GET - or POST request to this URL with the form parameters 'FaxStatus' and 'FaxSid'. Using an 'action' URL, your application - can receive synchronous notification that the message was successfully enqueued. If you provide an 'action' URL, RestComm - will use the RCML received in your response to the 'action' URL request to continue the current call. Any RCML verbs - occuring after a <Fax> which specifies an 'action' attribute are unreachable. If no 'action' is provided, <Fax> - will finish and RestComm will move on to the next RCML verb in the document. If there is no next verb, RestComm will end the - phone call. - - - Request Parameters - - - - - - Parameter - Description - - - - - FaxSid - The Sid for the Sms message. - - - FaxStatus - The current status of the Sms message. Either 'sent' or 'failed'. - - - -
- - method - The 'method' attribute takes the value 'GET' or 'POST'. This tells RestComm whether to request the 'action' URL via HTTP GET - or POST. This attribute is modeled after the HTML form 'method' attribute. - - - statusCallback - The 'statusCallback' attribute takes a URL as an argument. When the fax is actually sent, or if sending fails, - RestComm will make an asynchronous POST request to this URL with the parameters 'FaxStatus' and 'FaxSid'. - Note, 'statusCallback' always uses HTTP POST to request the given url. - - - Request Parameters - - - - - - Parameter - Description - - - - - FaxSid - The Sid for the fax message. - - - FaxStatus - The current status of the fax. Either 'sent' or 'failed'. - - - -
- - Nesting - The <Fax> verb can not have any other verbs or nouns nested. - - - Examples - For an example of how to use the <Fax> verb see below. - - - <Response> - <Fax>This is a test fax.</Fax> - </Response> - -
-
- Sms - The <Sms> verb sends an SMS message to a phone number during a phone call. - - Sms Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - to - phone number - see below - - - from - phone number - see below - - - action - relative or absolute URL - none - - - method - GET, POST - POST - - - statusCallback - relative or absolute URL - none - - - -
- - to - The 'to' attribute takes a valid E.164 phone number as a value. RestComm will send an SMS message to this number. - When sending an SMS during an incoming call, 'to' defaults to the caller. When sending an SMS during an outgoing call, - 'to' defaults to the called party. The value of 'to' must be a valid phone number. - - - from - The 'from' attribute takes a valid E.164 phone number as an argument. When sending an SMS during an incoming call, - 'from' defaults to the calling party. When sending an SMS during an outgoing call, 'from' defaults to the called party. - - - action - The 'action' attribute takes a URL as an argument. After processing the <Sms> verb, RestComm will make a GET - or POST request to this URL with the form parameters 'SmsStatus' and 'SmsSid'. Using an 'action' URL, your application - can receive synchronous notification that the message was successfully enqueued. If you provide an 'action' URL, RestComm - will use the RCML received in your response to the 'action' URL request to continue the current call. Any RCML verbs - occuring after an <Sms> which specifies an 'action' attribute are unreachable. If no 'action' is provided, <Sms> - will finish and RestComm will move on to the next RCML verb in the document. If there is no next verb, RestComm will end the - phone call. - - - method - The 'method' attribute takes the value 'GET' or 'POST'. This tells RestComm whether to request the 'action' URL via HTTP GET - or POST. This attribute is modeled after the HTML form 'method' attribute. - - - statusCallback - The 'statusCallback' attribute takes a URL as an argument. When the SMS message is actually sent, or if sending fails, - RestComm will make an asynchronous POST request to this URL with the parameters 'SmsStatus' and 'SmsSid'. - Note, 'statusCallback' always uses HTTP POST to request the given url. - - - Request Parameters - - - - - - Parameter - Description - - - - - SmsSid - The Sid for the Sms message. - - - SmsStatus - The current status of the Sms message. Either 'sent' or 'failed'. - - - -
- - Nesting - The <Sms> verb can not have any other verbs or nouns nested. - - - Examples - For an example of how to use the <Sms> verb see below. - - - <Response> - <Sms>Hello World!</Sms> - </Response> - - - - Sms Custom headers - - Restcomm will check for any custom headers and make use of them. - - - In case of an incoming SMS (incoming SIP MESSAGE) Restcomm will scan the message for any headers starting with X- and if any found will be send to - the application server with the request (part of the query string in the case of HTTP GET or part of the message body in the case of HTTP POST) - - - The application server can also provide some custom http headers (again starting with X-) along with the RCML response, that Restcomm will store and use them - when creating the outgoing SMS message (in the case of RCML response is Sms). - - - -
-
- Dial - The <Dial> verb connects the current caller to another phone. If the called party picks up, the two parties are - connected and can communicate until one hangs up. If the called party does not pick up, if a busy signal is received, or if - the number doesn't exist, the dial verb will finish. - - Dial Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - action - relative or absolute URL - no default for <Dial> - - - method - GET, POST - POST - - - timeout - positive integer in seconds - 30 seconds - - - timeLimit - positive integer (seconds) - 14400 seconds (4 hours) - - - callerId - a valid phone number, or client identifier if you are dialing a <Client>. - Caller's callerId - - - record - true, false - false - - - -
- - action - The 'action' attribute takes a URL as an argument. When the dialed call ends, RestComm will make a GET or POST - request to this URL including the parameters below. If you provide an 'action' URL, RestComm will continue the current - call after the dialed party has hung up, using the RCML received in your response to the 'action' URL request. - Any RCML verbs occuring after a <Dial> which specifies an 'action' attribute are unreachable. If no 'action' - is provided, <Dial> will finish and RestComm will move on to the next RCML verb in the document. - If there is no next verb, RestComm will end the phone call. - - - Request Parameters - - - - - - Parameter - Description - - - - - DialCallStatus - The outcome of the <Dial> attempt. See the DialCallStatus section below for details. - - - DialCallSid - The call sid of the new call leg. This parameter is not sent after dialing a conference. - - - DialCallDuration - The duration in seconds of the dialed call. This parameter is not sent after dialing a conference. - - - - RecordingUrl - The URL of the recorded audio. This parameter is only sent if record="true" is set on the Dial verb, and does not include recordings from the Record verb or Record=True on REST API calls.. - - - - -
- - DialCallStatus Values - - - - - - Parameter - Description - - - - - completed - The called party answered the call and was connected to the caller. - - - busy - RestComm received a busy signal when trying to connect to the called party. - - - no-answer - The called party did not pick up before the timeout period passed. - - - failed - RestComm was unable to route to the given phone number. This is frequently caused by dialing a - properly formated but non-existent phone number. - - - canceled - The call was canceled via the REST API before it was answered. - - - -
- - method - The 'method' attribute takes the value 'GET' or 'POST'. This tells RestComm whether to request the 'action' - URL via HTTP GET or POST. This attribute is modeled after the HTML form 'method' attribute. - - - timeout - The 'timeout' attribute sets the limit in seconds that <Dial> waits for the called party to answer the call. - - - timelimit - The 'timeLimit' attribute sets the maximum duration of the <Dial> in seconds. - - - callerId - The 'callerId' attribute lets you specify the caller ID that will appear to the called party when RestComm calls. - By default, when you put a <Dial> in your RCML response to RestComm's inbound call request, the caller ID that - the dialed party sees is the inbound caller's caller ID. If you are dialing to a <Client>, you can set a client - identifier as the callerId attribute. For instance, if you've set up a client for incoming calls and you are dialing to - it, you could set the callerId attribute to client:thomas. - - - record - The 'record' attribute lets you specify whether the call will be recorded or not. - By default, the call is not recorded. If you set the attribute to 'true' Restcomm will start recording when the two calls are bridged - - -Record attribute for Dial Conference - -The current release of Restcomm doesn't support recording for Dial Conference. This feature will be available in the next release - - - - Nesting - You can nest the following nouns within <Dial>: <Number>, <Client>, <Conference> - - - Examples - For examples of how to use the <Dial> verb see below. - - - <Response> - <Dial>1-444-555-666</Dial> - </Response> - - - <Response> - <Dial record="true">1-444-555-666</Dial> - </Response> - - - <Response> - <Dial callerId="1555666777" record="true">1-444-555-666</Dial> - </Response> - -
- Number - The <Number> noun specifies a phone number to dial. You can use multiple <Number> nouns within a - <Dial> verb to simultaneously call all of them at once. The first call to pick up is connected to the current call and - the rest are hung up. - - Number Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - sendDigits - any digits - none - - - url - any url - none - - - -
- - sendDigits - The 'sendDigits' attribute tells RestComm to play DTMF tones when the call is answered. - This is useful when dialing a phone number and an extension. RestComm will dial the number, and - when the automated system picks up, send the DTMF tones to connect to the extension. - - - url - The 'url' attribute allows you to specify a url for a RCML document that will run on the - called party's end, after he/she answers, but before the parties are connected. You can use this RCML - to privately play or say information to the called party, or provide a chance to decline the phone call - using <Gather> and <Hangup>. The current caller will continue to hear ringing while the - RCML document executes on the other end. RCML documents executed in this manner are not allowed to contain - the <Dial> verb. - - - Examples - For an example of how to use the <Number> noun see below. - - - <Response> - <Dial> - <Number sendDigits="wwww1234">1-444-555-6666</Number> - </Dial> - </Response> - -
-
- Client - The <Client> noun specifies a client identifier to dial. You can use multiple <Client> nouns within - a <Dial> verb to simultaneously attempt a connection with many clients at once. The first client to accept the - incoming connection is connected to the call and the other connection attempts are canceled. - - Examples - For an example of how to use the <Client> none see below. - - - <Response> - <Dial> - <Client>thomas</Client> - </Dial> - </Response> - -
-
- Conference - The <Conference> noun allows you to connect to a named conference room and talk with the other callers - who have also connected to that room. The name of the room is up to you and is namespaced to your account. - - Conference Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - muted - true, false - false - - - beep - true, false - true - - - startConferenceOnEnter - true, false - true - - - endConferenceOnExit - true, false - false - - - waitUrl - RCML url, empty string - default RestComm hold music - - - waitMethod - GET or POST - POST - - - maxParticipants - positive integer <= 40 - 40 - - - -
- - muted - The 'muted' attribute lets you specify whether a participant can speak in the conference. - If this attribute is set to 'true', the participant will only be able to listen to people in the conference. - - - beep - The 'beep' attribute lets you specify whether a notification beep is played to the conference when a participant - joins or leaves the conference. - - - startConferenceOnEnter - This attribute tells a conference to start when this participant joins the conference, if it is not already started. - If this is false and the participant joins a conference that has not started, they are muted and hear background music - until a participant joins where startConferenceOnEnter is true. This is useful for implementing moderated conferences. - - - endConferenceOnExit - If a participant has this attribute set to 'true', then when that participant leaves, the conference ends and all - other participants drop out. This is useful for implementing moderated conferences that bridge two calls and allow either - call leg to continue executing RCML if the other hangs up. - - - waitUrl - The 'waitUrl' attribute lets you specify a URL for music that plays before the conference has started. The URL may - be a WAV or a RCML document that uses <Play> or <Say> for content. This defaults to a selection of Creative - Commons licensed background music, but you can replace it with your own music and messages. If the 'waitUrl' responds with - RCML, RestComm will only process <Play>, <Say>, and <Redirect> verbs. If you do not wish anything to play - while waiting for the conference to start, specify the empty string (set 'waitUrl' to ''). - - - waitMethod - This attribute indicates which HTTP method to use when requesting 'waitUrl'. It defaults to 'POST'. Be sure to use - 'GET' if you are directly requesting static audio files such as WAV files so that RestComm properly caches the files. - - - maxParticipants - This attribute indicates the maximum number of participants you want to allow within a named conference room. The default - maximum number of participants is 40. The value must be a positive integer less than or equal to 100. - - - Examples - For an example of how to use the <Conference> noun see below. - - - <Response> - <Dial> - <Conference>1234</Conference> - </Dial> - </Response> - -
- - - - Uri noun deprecated - The Uri noun has been deprecated and will no longer be available in future releases. Please use the SIP noun instead. - - -
- Sip - The <Sip> noun specifies a SIP URI to dial. You can use multiple <Sip> nouns within - a <Dial> verb to simultaneously attempt a connection with many user agents at once. The first - user agent to accept the incoming connection is connected to the call and the other connection attempts - are canceled. - - The Dial verb's Sip noun lets you set up VoIP sessions by using SIP -- Session Initiation Protocol. With this feature, you can send a call to any SIP endpoint. Set up your RCML to use the Sip noun within the Dial verb. - - -Currently, only one Sip noun may be specified per Dial, and the INVITE message may be sent to only one SIP endpoint. Also, you cannot add any other nouns (eg Number, Client) in the same Dial as the SIP. If you want to use another noun, set up a callback on the Dial to use alternate methods. - - - Examples - For an example of how to use the <Sip> noun see below. - - - <Response> - <Dial> - <Sip>sip:alice@127.0.0.1:5080</Sip> - </Dial> - </Response> - - - -Authentication - - - -Send username and password attributes for authentication to your SIP infrastructure as attributes on the Sip noun. - - - - Request Parameters - - - - - - Attribute Name - Values - - - - - username - Username for SIP authentication. - - - password - Password for SIP authentication - - - - -
- - -Authentication Example - - - - - - - sip:alice@example.com - - - -]]> - - - - -Custom headers - - - -Send custom headers by appending them to the SIP URI -- just as you'd pass headers in a URI over HTTP. For example: - - - - - - - - sip:alice@example.com?mycustomheader=tata&myotherheader=toto - - - - -]]> - - - -Character Limit -While the SIP URI itself must be under 255 chars, the headers must be under 1024 characters. - - - - -Transport - - - -Set a parameter on your SIP URI to specify what transport protocol you want to use. Currently, this is limited to TCP and UDP. By default, Restcomm sends your SIP INVITE over UDP. Change this by using the transport parameter: - - - - - - - - - - sip:alice@example.com;transport=tcp - - - - -]]> - - - - -Attributes - - - - - Request Parameters - - - - - - - Attribute Name - Allowed Values - Default Value - - - - - url - call screening url. - none. - - - method - GET, POST - POST - - - - -
- - - - -The url attribute allows you to specify a url for a RCML document that runs on the called party's end, after they answer, but before the two parties are connected. You can use this RCML to privately Play or Say information to the called party, or provide a chance to decline the phone call using Gather and Hangup. The current caller continues to hear ringing while the RCML document executes on the other end. RCML documents executed in this manner cannot contain the Dial verb. - - - -method - - - -The method attribute allows you to specify which HTTP method Restcomm should use when requesting the URL specified in the url attribute. The default is POST. - - - - -Call Screening HTTP parameters - - - - -When a call is answered, Restcomm passes the following parameters with its request to your screening URL (in addition to the standard RCML Voice request parameters): - - - - - - Request Parameters - - - - - - Attribute Name - Values - - - - - - SipCallId - The SIP call ID header of the request made to the remote SIP infrastructure. - - - SipHeader - The name/value of any X-headers returned in the 200 response to the SIP INVITE request. - - - - -
- - - -Dial Action HTTP parameters - - - -Use the action callback parameters to modify your application based on the results of the SIP dial attempt: - - - - - Request Parameters - - - - - - Attribute Name - Values - - - - - - DialSipCallId - The SIP call ID header of the request made to the remote SIP infrastructure. - - - DialSipResponseCode - The SIP response code as a result of the INVITE attempt. - - - - DialSipHeader_ - The name/value of any X-headers returned in the final response to the SIP INVITE request. - - - - -
- - -Dial with Multiple Examples - - - -A more complex Dial, specifying custom settings as attributes on Dial, including call screening and setting the protocol to TCP. - - - - - - - - sip:alice@example.com?customheader=foo - - - - -]]> - - - -
- -
-
- Hangup - The <Hangup> verb ends a call. - - Nesting - The <Hangup> verb can not have any other verbs or nouns nested. - - - Examples - For an example of how to use the <Hangup> verb see below. - - - <Response> - <Hangup/> - </Response> - -
-
- Redirect - The <Redirect> verb transfers control of a call to the RCML at a different URL. - All verbs after <Redirect> are unreachable and ignored. - - Redirect Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - method - GET, POST - POST - - - -
- - method - The 'method' attribute takes the value 'GET' or 'POST'. This tells RestComm whether to - request the URL via HTTP GET or POST. - - - Nesting - The <Redirect> verb can not have any other verbs or nouns nested. - - - Examples - For an example of how to use the <Redirect> verb see below. - - - <Response> - <Redirect>http://foobar.com/instructions</Redirect> - </Response> - -
-
- Reject - The <Reject> verb rejects an incoming call to your RestComm endpoint. This is useful for blocking unwanted calls. - If the first verb in a RCML response is <Reject>, RestComm will not pick up the call. The call ends with a status of - 'busy' or 'no-answer', depending on the verb's 'reason' attribute. Any verbs after <Reject> are unreachable and ignored. - - Reject Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - reason - rejected, busy - rejected - - - -
- - reason - The reason attribute takes the values "rejected" and "busy." This tells RestComm what message to play when - rejecting a call. Selecting "busy" will play a busy signal to the caller, while selecting "rejected" will - play a standard not-in-service response. - - - Nesting - The <Reject> verb can not have any other verbs or nouns nested. - - - Examples - For an example of how to use the <Reject> verb see below. - - - <Response> - <Reject reason="busy"/> - </Response> - -
-
- Pause - The <Pause> verb waits silently for a specific number of seconds. If <Pause> is the first verb - in a RCML response, RestComm will wait the specified number of seconds before picking up the call. - - Pause Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - length - integer > 0 - 1 second - - - -
- - length - The 'length' attribute specifies how many seconds RestComm will wait silently before continuing on. - - - Nesting - The <Pause> verb can not have any other verbs or nouns nested. - - - Examples - For an example of how to use the <Pause> verb see below. - - - <Response> - <Pause length="5"/> - </Response> - -
- - - -
- UssdCollect - The UssdCollect verb is a USSD integration with Restcomm resource. When fully configured with a USSD gateway, Restcomm will send all UssdMessage to the configured gateway - - - - Example - For an example of how to use the USSDCollect verb see below. - - - - - 1 for first option - 2 for first option - 3 for first option - - - ]]> - - - -
-UssdMessage - - - -Before you can send an USSD message you must first configure the restcomm.xml with the USSD gateway information. See for more information. - - - The UssdMessage verb contains the message that Restcomm will send to the USSD gateway - - - Nesting - The UssdMessage can be nested inside of the USSDCollect verb. - - - - Examples - For an example of how to use the USSDCollect verb see below. - - - - Test USSD Message - - ]]> - - -
- -
-Language - The language noun let you specify the language of the USSD message that is sent to the USSD gateway - - - - - Language Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - Language - bf, bp, en, en-gb,cf, cs, dan, fi es, fr, de, el, it, nl, no, pl, pt, ru, ar, ca, sv, th, tr - en - - - -
- - - Nesting - The Language noun cannot be nested. If not specified, English language is default - - - Example - For an example of how to use the USSD Language noun see below. - - - - fr - Test USSD Message - - ]]> - - -
- -
- - - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Restful-API.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Restful-API.xml deleted file mode 100644 index ef98657df4..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Restful-API.xml +++ /dev/null @@ -1,2901 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - Restful APIs - The RestComm Restufl APIs are very similar to Twilio's and allow - you to query meta-data about your account, phone numbers, - calls, text - messages, and recordings. Through the Restful API you can also - start - outbound calls and send text messages. - - Resource Encoding - When an HTTP client makes a request to RestComm for a resource - the HTTP client has the option to pick which encoding - is used for the - response. Currently, RestComm supports XML and JSON - encoding. The - default encoding is XML but - to receive the resource in JSON just - append .json to the end of the - resource name. - - - - User Authentication - Before accepting any Restful API call, Restcomm will - authenticate the request source using either a valid Account SID or - an Email Account combined with an Authentication Token. - - See below for the default Administrator account. - - - - - - - - or - - - - - - - - -
- Accounts - Accounts and sub-accounts are useful for things like segmenting - phone numbers and usage data for your users and - controlling access to - data. - - Account Resource URI - /2012-04-24/Accounts/{AccountSid} - - - Resource Properties - - - - - - Property - Description - - - - - Sid - A string that uniquely identifies this account. - - - DateCreated - The date that this account was created. - - - DateUpdated - The date that this account was last updated. - - - FriendlyName - A description of this account, up to 64 characters long. By - default - the FriendlyName is your email address. - - - Status - The status of this account. Possible values are active, - suspended, and closed. - - - AuthToken - The authorization token for this account. This should not - be shared. - - - Uri - The URI for this account, relative to - http://localhost:port/restcomm. - - - -
-
- Supported Operations - - HTTP GET - Returns the representation of an Account resource, including - the properties above. - - - - - - - Account Resource URI - /2012-04-24/Accounts/{EmailAddress} - - - HTTP POST/PUT - Modifies an Account resource and returns the representation, - including the properties above. Below - you will find a list of - optional parameters. - - - Request Parameters - - - - - - Parameter - Description - - - - - FriendlyName - A description of this account, up to 64 characters long. - - - Status - The status of this account. Possible values are active, - suspended, and closed. - - - Password - A password that will be used to generate the AuthToken for - the new Account resource. - - - -
- - - Get information about the default account - - - - - - - - Change default account password(AuthToken) - - - - - - - Command line versus Browser - - The above command uses the Account SID and the one below uses the - Email Account. Note the - administrator%40company.com - is used instead of - administrator@company.com - . This is because using curl on the bash terminal doesn't parse the - @ correctlyl. If you were to running on a browser, you can safely - use the @ as the web browser will correctly handle it. - - - - - - - The above commands will print an output similar to the one - below: - - - - ACae6e420f425248d6a26948c17a9e2acf - Default Administrator Account - active - Full - 2012-04-24T00:00:00.000-06:00 - 2012-04-24T00:00:00.000-06:00 - 77f8c12cc7b8f8423e5c38b035249166 - /2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf - - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/AvailablePhoneNumbers - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Calls - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Conferences - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/IncomingPhoneNumbers - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Notifications - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/OutgoingCallerIds - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Recordings - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Sandbox - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/SMS/Messages - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Transcriptions - - - -]]> - - - -
-
- Account List Resource - - Account List Resource URI - /2012-04-24/Accounts - -
- Supported Operations - - HTTP GET - Returns the list representation of all the Sub-Account - resources for this Account, including the - properties above. - - - - - - - - - HTTP POST - Creates a new Sub-Account and returns the representation of - the Sub-Account resource, including - the properties above. Below you - will find a list of required and - optional parameters. - - - - - - Request Parameters - - - - - - Parameter - Description - - - - - EmailAddress(Required) - The email address to use for this account. - - - FriendlyName - A description of this account, up to 64 characters long. - Default, is your email address. - - - Status - The status of this account. Default is active, possible - values are active, suspended, and closed. - - - Password(Required) - A password that will be used to generate the AuthToken - for the new Account resource. - - - Role(Required) - The security role that this Account resource will use. If - no role is provided then - the role of the account resource - creating this will be inherited - to the new Account resource - and - may compromise the system. - - - -
- - Here is an example of how to createa a sub-account. The - sub-account will inherit the same permissions has the - Administrator's account. - - - - - - About Sub-accounts - Note that the SID, Email and the AuthToken (see output below) - of the sub-account can now be used instead of the Administrator's - account - - - - - AC3b8f0dd2e5026abde018446cbb3b185d - MySubAccount - active - Full - 2013-10-16T09:22:28.708-06:00 - 2013-10-16T09:22:28.712-06:00 - 53134d7a9914e2b47c8435ebdb50ded3 - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d - - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/AvailablePhoneNumbers - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/Calls - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/Conferences - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/IncomingPhoneNumbers - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/Notifications - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/OutgoingCallerIds - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/Recordings - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/Sandbox - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/SMS/Messages - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/Transcriptions - - - -]]> - - - - -
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- AvailablePhoneNumbers - The AvailablePhoneNumbers subresources let you search for - incoming local and toll-free phone numbers that are available - for you - to purchase from a TeleStax partner. - - AvailablePhoneNumbers List Resource URI - /2012-04-24/Accounts/{AccountSid}/AvailablePhoneNumbers/US/Local - - - Searching For Numbers. - When using RestComm the way to search for new phone numbers is - by searching - the AvailablePhoneNumbers list resource and providing - the desired area - code as a filter. - - - Resource Properties - - - - - - Property - Description - - - - - FriendlyName - A friendly version of the phone number. - - - PhoneNumber - The phone number, in E.164 format. - - - Lata - The LATA for this phone number. - - - RateCenter - The rate center for this phone number. - - - Latitude - The latitude coordinate for this phone number. - - - Longitude - The longitude coordinate for this phone number. - - - Region - The two-letter state or province abbreviation for this - phone number. - - - PostalCode - The zip code for this phone number. - - - IsoCountry - The ISO country code for this phone number.. - - - -
- - - - -
- Supported Operations - - HTTP GET - Returns the representation of an AvailablePhoneNumber - resource, including the properties above. - - - Request Parameters - - - - - - Property - Description - - - - - AreaCode - A three digit area code inside the U.S. - - - -
- - - Querying Available Phone Numbers - - You need to be using RestComm for VoIP Innovations in order to be - able to use this feature. See here for more details - RestComm AMI for VoIP Innovation - - - - - Here is an example, the AreaCode is any valid United States - Code - -:8080/restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/AvailablePhoneNumbers/US/Local -d "AreaCode=305" - -]]> - - - -
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- -
- Gateways - The Gateways subresources let you create sip accounts that Restcomm will use to register itself to the Gateway and receive incoming traffic. - - - Gatewat List Resource URI - /2012-04-24/Accounts/{AccountSid}/Management/Gateways - - - Register Restcomm instance to a SIP Gateway. - You might need to register Restcomm instnace to a SIP Gateway and receive incoming traffic from that gateway. - For that you need to use Gateway REST endpoint - - - Resource Properties - - - - - - Property - Description - - - - - FriendlyName - A friendly version of the gateway. - - - UserName - The username that will be used to register to this gateway - - - Password - The password that will be used to register to this gateway - - - Proxy - The proxy address of the gateway - - - Register - Boolean flag to register or not the gateway - - - TTL - Time to live for the Register - - - -
- -
- Supported Operations - - HTTP GET - Returns the representation of a Gateway - resource, including the properties above. - Resource URI: - /2010-04-01/Accounts/{AccountSid}/Management/Gateways/{GatewaySid} - - - HTTP POST - Creates a new Gateway - resource and returns the representation of the resource, including the properties above. - Resource URI: - /2010-04-01/Accounts/{AccountSid}/Management/Gateways - - - HTTP POST/PUT - Update a Gateway - resource and returns the representation of the resource, including the properties above. - Resource URI: - /2010-04-01/Accounts/{AccountSid}/Management/Gateways/{GatewaySid} - - - HTTP DELETE - Deletes a Gateway resource. - Resource URI: - /2010-04-01/Accounts/{AccountSid}/Management/Gateways/{GatewaySid} - -
- -
- Examples - - Create a new Gateway - -:8080/restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Management/Gateways -d "FriendlyName=mygateway" -d "UserName=username" -d "Password=password" -d "Proxy=my.gateway.com" -d "Register=true" -d "TTL=3600" - -]]> - - - - - Get a list of available Gateways - -:8080/restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Management/Gateways - -]]> - - - - - Update an existing Gateway - -:8080/restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Management/Gateways/GW106bc6f34bd24790a435eaeccc1aed72 -d "FriendlyName=MyGatewayNewName" -d "UserName=newUserName" - -]]> - - - - - Delete an existing Gateway - -:8080/restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Management/Gateways/GW1cffb069192a45f2b5f5af2e76489550 - -]]> - - - -
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- -
- Clients - An Client instance resource represents a user agent registered - with RestComm. - - Client Resource URI - /2012-04-24/Accounts/{AccountSid}/Clients/{ClientSid} - - - Using SIP User Agents - When using RestComm to handle SIP user agent you have to create - a new Client resource, - this resource acts as an account for your user - agent and also dictates - how calls made by the - user agent should be - handled. - - - - Client without VoiceUrl - Restcomm has a new implied behavior when VoiceUrl is not - provided for a Client account. Restcomm will proxy calls from such - Clients to the destination Client (only if registered) or to the - destination Application DID. - - - Only registered Clients are allowed to use the B2BUA/P2P/Proxy - functionalities of Restcomm. Proxying and P2P calls are only - allowed between registered(authenticated) Clients. - - - - Resource Properties - - - - - - Property - Description - - - - - Sid - A string that uniquely identifies this client. - - - DateCreated - The date that this client was created. - - - DateUpdated - The date that this clientr was last updated. - - - FriendlyName - A friendly name for this client. - - - AccountSid - The unique id of the Account that owns this phone number. - - - ApiVersion - Calls to this phone number will create a new RCML session - with this API version. - - - Login - The name that is used inside the <Client> noun. This - is also used by - the user agent as the user name used for - registration and outbound - dialing. - - - Password - The password used by the user agent during registration and - outbound dialing. - - - Status - The client status the possible values are 0 for disabled - and 1 for enabled. - - - VoiceUrl - The URL RestComm will request when this client makes an - outbound call. - - - VoiceMethod - The HTTP method RestComm will use when requesting the above - Url. Either GET or POST. - - - VoiceFallbackUrl - The URL that RestComm will request if execution of VoiceUrl - fails for any reason. - - - VoiceFallbackMethod - The HTTP method RestComm will use when requesting the - VoiceFallbackUrl. Either GET or POST. - - - VoiceApplicationSid - If this entry contains an Sid to a voice application then - RestComm will ignore these voice URLs - and use the voice URLs - specified by the voice application. - - - StatusCallback - The URL that RestComm will request to pass status - parameters (such as the call state) to your - application. - - - StatusCallbackMethod - The HTTP method RestComm will use to make requests to the - StatusCallback URL. Either GET or POST. - - - Uri - The URI for this Client, relative to - http://localhost:port/restcomm. - - - -
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- Supported Operations - - HTTP GET - Returns the representation of an Client resource, including - the properties above. - - - HTTP POST/PUT - Modifies a Client resource and returns the representation, - including the properties above. Below - you will find a list of - optional parameters. - - - Request Parameters - - - - - - Parameter - Description - - - - - FriendlyName - A formatted version of this phone number. - - - Password - The password used by the user agent during registration - and outbound dialing. - - - Status - The client status the possible values are 0 for disabled - and 1 for enabled. - - - VoiceUrl - The URL RestComm will request when this phone number - receives a call. - - - VoiceMethod - The HTTP method RestComm will use when requesting the - above Url. Either GET or POST. - - - VoiceFallbackUrl - The URL that RestComm will request if execution of - VoiceUrl fails for any reason. - - - VoiceFallbackMethod - The HTTP method RestComm will use when requesting the - VoiceFallbackUrl. Either GET or POST. - - - VoiceApplicationSid - If this entry contains an Sid to a voice application then - RestComm will ignore these voice URLs - and use the voice URLs - specified by the voice application. - - - StatusCallback - The URL that RestComm will request to pass status - parameters (such as the call state) to your - application. - - - StatusCallbackMethod - The HTTP method RestComm will use to make requests to the - StatusCallback URL. Either GET or POST. - - - -
- - HTTP DELETE - Deletes a Client from the user's Account. - -
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- Client List Resource - - Client List Resource URI - /2012-04-24/Accounts/{AccountSid}/Clients - -
- Supported Operations - - HTTP GET - Returns the list representation of all the Client resources - for this Account, including the - properties above. - - - HTTP POST - Creates a new Client and returns the representation of the - resource, including - the properties above. Below you will find a - list of required and - optional parameters. - - - Request Parameters - - - - - - Parameter - Description - - - - - FriendlyName - A formatted version of this phone number. - - - Login - The name that is used inside the <Client> noun. - This is also used by - the user agent as the user name used for - registration and outbound - dialing. - - - Password - The password used by the user agent during registration - and outbound dialing. - - - Status - The client status the possible values are 0 for disabled - and 1 for enabled. - - - VoiceUrl - The URL RestComm will request when this phone number - receives a call. - - - VoiceMethod - The HTTP method RestComm will use when requesting the - above Url. Either GET or POST. - - - VoiceFallbackUrl - The URL that RestComm will request if execution of - VoiceUrl fails for any reason. - - - VoiceFallbackMethod - The HTTP method RestComm will use when requesting the - VoiceFallbackUrl. Either GET or POST. - - - VoiceApplicationSid - If this entry contains an Sid to a voice application then - RestComm will ignore these voice URLs - and use the voice URLs - specified by the voice application. - - - StatusCallback - The URL that RestComm will request to pass status - parameters (such as the call state) to your - application. - - - StatusCallbackMethod - The HTTP method RestComm will use to make requests to the - StatusCallback URL. Either GET or POST. - - - -
- - - - Create a Client - The client name will be Alice as shown below - - - - - The output of the command will be similar to the one below - - - - - Delete a Client - You must use the Client SID - - - - - - - - Change Client's Password - You must use the Client SID as shown below: - - - - - - - - Get List of available Clients - The command below shows all Clients created using the default - Admin Account - - - - - -
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- IncomingPhoneNumbers - An IncomingPhoneNumber instance resource represents a RestComm - phone number. - - IncomingPhoneNumber Resource URI - /2012-04-24/Accounts/{AccountSid}/IncomingPhoneNumbers/{IncomingPhoneNumberSid} - - - Binding a Phone Number to an Application - When using RestComm the way to bind a phone number to an - application is be creating - a new IncomingPhoneNumber resource and - providing the VoiceUrl to your - application. - - - Resource Properties - - - - - - Property - Description - - - - - Sid - A string that uniquely identifies this incoming phone - number. - - - DateCreated - The date that this incoming phone number was created. - - - DateUpdated - The date that this incoming phone number was last updated. - - - FriendlyName - A formatted version of this phone number. - - - AccountSid - The unique id of the Account that owns this phone number. - - - PhoneNumber - The incoming phone number in E.164 format ex. +2223334444 - - - ApiVersion - Calls to this phone number will create a new RCML session - with this API version. - - - VoiceCallerIdLookup - Look up the caller's caller-ID name. Either true or false. - - - VoiceUrl - The URL RestComm will request when this phone number - receives a call. - - - VoiceMethod - The HTTP method RestComm will use when requesting the above - Url. Either GET or POST. - - - VoiceFallbackUrl - The URL that RestComm will request if execution of VoiceUrl - fails for any reason. - - - VoiceFallbackMethod - The HTTP method RestComm will use when requesting the - VoiceFallbackUrl. Either GET or POST. - - - StatusCallback - The URL that RestComm will request to pass status - parameters (such as the call state) to your - application. - - - StatusCallbackMethod - The HTTP method RestComm will use to make requests to the - StatusCallback URL. Either GET or POST. - - - SmsUrl - The URL that RestComm will request when receiving an - incoming SMS message to this number. - This may not be supported. - Please consult with your DID provider. - - - SmsMethod - The HTTP method RestComm will use when making requests to - the SmsUrl. Either GET or POST. - - - SmsFallbackUrl - The URL that RestComm will request if SmsUrl fail for any - reason. Please see SmsUrl - as this feature may not be supported. - - - SmsFallbackMethod - The HTTP method RestComm will use when making requests to - SmsFallbackUrl. Either GET or POST. - - - Uri - The URI for this incoming phone number, relative to - http://localhost:port/restcomm. - - - -
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- Supported Operations - - HTTP GET - Returns the representation of an IncomingPhoneNumber resource, - including the properties above. - - - HTTP POST/PUT - Modifies an IncomingPhoneNumber resource and returns the - representation, including the properties above. Below - you will find - a list of optional parameters. - - - Request Parameters - - - - - - Parameter - Description - - - - - FriendlyName - A formatted version of this phone number. - - - ApiVersion - Calls to this phone number will create a new RCML session - with this API version. - - - VoiceCallerIdLookup - Look up the caller's caller-ID name. Either true or false. - - - VoiceUrl - The URL RestComm will request when this phone number - receives a call. - - - VoiceMethod - The HTTP method RestComm will use when requesting the - above Url. Either GET or POST. - - - VoiceFallbackUrl - The URL that RestComm will request if execution of - VoiceUrl fails for any reason. - - - VoiceFallbackMethod - The HTTP method RestComm will use when requesting the - VoiceFallbackUrl. Either GET or POST. - - - StatusCallback - The URL that RestComm will request to pass status - parameters (such as the call state) to your - application. - - - StatusCallbackMethod - The HTTP method RestComm will use to make requests to the - StatusCallback URL. Either GET or POST. - - - SmsUrl - The URL that RestComm will request when receiving an - incoming SMS message to this number. - This may not be supported. - Please consult with your DID provider. - - - SmsMethod - The HTTP method RestComm will use when making requests to - the SmsUrl. Either GET or POST. - - - SmsFallbackUrl - The URL that RestComm will request if SmsUrl fail for any - reason. Please see SmsUrl - as this feature may not be supported. - - - SmsFallbackMethod - The HTTP method RestComm will use when making requests to - SmsFallbackUrl. Either GET or POST. - - - -
- - HTTP DELETE - Releases an IncomingPhoneNumber from the user's Account. - -
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- IncomingPhoneNumber List Resource - - IncomingPhoneNumber List Resource URI - /2012-04-24/Accounts/{AccountSid}/IncomingPhoneNumbers - -
- Supported Operations - - HTTP GET - Returns the list representation of all the - IncomingPhoneNumber resources for this Account, including the - properties above. - - - HTTP POST - Creates a new IncomingPhoneNumber and returns the - representation of the resource, including - the properties above. - Below you will find a list of required and - optional parameters. - - - Request Parameters - - - - - - Parameter - Description - - - - - PhoneNumber(Required) - The phone number you want to provision. - - - FriendlyName - A formatted version of this phone number. - - - ApiVersion - Calls to this phone number will create a new RCML session - with this API version. - - - VoiceCallerIdLookup - Look up the caller's caller-ID name. Either true or - false. - - - VoiceUrl - The URL RestComm will request when this phone number - receives a call. - - - VoiceMethod - The HTTP method RestComm will use when requesting the - above Url. Either GET or POST. - - - VoiceFallbackUrl - The URL that RestComm will request if execution of - VoiceUrl fails for any reason. - - - VoiceFallbackMethod - The HTTP method RestComm will use when requesting the - VoiceFallbackUrl. Either GET or POST. - - - StatusCallback - The URL that RestComm will request to pass status - parameters (such as the call state) to your - application. - - - StatusCallbackMethod - The HTTP method RestComm will use to make requests to the - StatusCallback URL. Either GET or POST. - - - SmsUrl - The URL that RestComm will request when receiving an - incoming SMS message to this number. - This may not be supported. - Please consult with your DID provider. - - - SmsMethod - The HTTP method RestComm will use when making requests to - the SmsUrl. Either GET or POST. - - - SmsFallbackUrl - The URL that RestComm will request if SmsUrl fail for any - reason. Please see SmsUrl - as this feature may not be supported. - - - SmsFallbackMethod - The HTTP method RestComm will use when making requests to - SmsFallbackUrl. Either GET or POST. - - - -
- - - Attach a phone number to an application - This one uses the default application - - - - - - - - - - Delete a phone number - You have to get the SID of the phone and use curl to delete - as follows - - - - - - - List of phone numbers - Gets all numbers created using IncomingPhoneNumbers.json - - - - - -
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- - - -
- Calls - A Call represents a connection between a phone or user agent and - RestComm. This may be inbound - or outbound. The Calls list resource - represents the set of phone calls - originated and terminated from - an - account. - - Call Resource URI - /2012-04-24/Accounts/{AccountSid}/Calls/{CallSid} - - - Resource Properties - - - - - - Property - Description - - - - - Sid - A string that uniquely identifies this call. - - - ParentCallSid - A string that uniquely identifies the call that created - this leg. - - - DateCreated - The date that this call was created. - - - DateUpdated - The date that this call was last updated. - - - AccountSid - The unique id of the Account that created this call. - - - To - The phone number or identifier that will be the recipient - of this call. - - - From - The phone number or identifier that originated this call. - - - PhoneNumberSid - If the call was inbound, this is the Sid of the - IncomingPhoneNumber that received the call. - - - Status - A string representing the status of the call. Possible - values are queued, ringing, in-progress, completed, - failed, busy - and no-answer. - - - StartTime - The start time of the call. Empty if the call has not yet - been started. - - - EndTime - The end time of the call. Empty if the call has not ended.. - - - Duration - The length of the call in seconds. - - - - Price - The charge for this call, in the currency associated with - the account. Populated after the call is completed. - - - - - Direction - A string describing the direction of the call. Possible - values are inbound, outbound-api, - and outbound-dial - - - - AnsweredBy - If this call was initiated with answering machine - detection, either human or machine. Empty otherwise. - - - - ApiVersion - Displays the current API version - - - - ForwardFrom - If this call was an incoming call forwarded from another - number, the forwarding phone number (depends on carrier - supporting forwarding). Empty otherwise. - - - - - CallerName - If this call was an incoming call, the caller's name. Empty - otherwise. - - - Uri - The URI for this account, relative to - http://localhost:port/restcomm. - - - -
- - -
- Supported Operations - - HTTP GET - Returns the representation of a Call resource, including the - properties above. - -
- - -
- Call List Resource - - Call List Resource URI - /2012-04-24/Accounts/{AccountSid}/Calls - -
- -
- Supported Operations - - HTTP GET - Returns the list representation of all the Call resources for - this Account, including the - properties above. - - - HTTP POST - Makes a new Call and returns the representation of the Call - resource, including - the properties above. Below you will find a list - of required and optional parameters. - - - Request Parameters - - - - - - Parameter - Description - - - - - From(Required) - The phone number to use as the caller id. - - - To(Required) - The phone number to call. - - - Url(Required) - The fully qualified URL that should be executed when the - call connects. - - - Method - The HTTP method RestComm should use when making its - request to the above Url. Defaults to POST. - - - FallbackUrl - The URL that RestComm will request if execution of Url - fails for any reason. - - - FallbackMethod - The HTTP method that RestComm should use to request the - FallbackUrl. Must be either GET or POST. - Defaults to POST. - - - StatusCallback - A URL that RestComm will request when the call ends to - notify your app. - - - StatusCallbackMethod - The HTTP method RestComm should use when requesting the - above StatusCallback. Defaults to POST. - - - Timeout - The number of seconds that RestComm should allow the phone - to ring before - assuming there is no answer. The default is 60 - seconds. - - - -
- - - - Making a call to a SIP account - - Restcomm will make a call to any SIP account that is reachable. - It the example below, the SIP account is listening on port 5060. - When you make the call, the SIP phone on which Alice is registered - will ring and the hello-play.xml file will be played. - - - - - - - Making a call to a Restcomm client - - You must first create a RestComm client. In the example below, - the Restcomm client created is called Alice. When you make the call, - the SIP phone on which Alice is registered will ring and the - hello-play.xml file will be played. - - - - - - - - - - - - Calling a DID number - - The above example shows how to make a call to a SIP number. If you - want to make a call to a DID number, you must can connect Restcomm - to a DID provisioning service provider. The quickest way is to use - RestComm AMI for Voice Innovation - - - - - Get a list of all available calls - This will return all the available calls linked to the account - SID - - - - Working on a production server - Using filter is a good practice on a server with thousands or - millions of calls - - - - - - If the system hasn't received any calls, you will see the the - output below - - - - - -]]> - -
- - -
- REST API: Modifying Live Calls - - - Realtime call modification allows you to interrupt an - in-progress call and - terminate it or have it begin processing RCML - from a new URL. This - is useful for any application where you want to - asynchronously - change the behavior of a running call. For example: - hold music, call - queues, transferring calls, forcing hangup, etc. - - -
- HTTP POST to a Call - - Client List Resource URI - - - To redirect or terminate a live call, you make an HTTP POST - request - to an in-progress Call instance resource URI: - - /2010-04-01/Accounts/{AccountSid}/Calls/{CallSid} - -
- - - POST Parameters - - - The following parameters are available for you to POST when - modifying a - phone call: - - - - Request Parameters - - - - - - Parameter - Description - - - - - Url - A valid URL that returns TwiML. Twilio will immediately - redirect the call to the new TwiML. - - - Method - The HTTP method Twilio should use when requesting the - above URL. Defaults to POST. - - - Status - Either canceled or completed. Specifying canceled will - attempt to hangup calls that are queued or ringing but not - affect calls already in progress. Specifying completed will - attempt to hang up a call even if it's already in progress. - - - - -
- - Call in-Progress - - Note that any call which is currently ringing within a Dial - verb is - in-progress from the point of view of Restcomm, and thus you - must - use 'Status=completed' to cancel it. - - - - - - Optional Parameters - - You may POST the following parameters: - - - - - - Request Parameters - - - - - - Parameter - Description - - - - - FallbackUrl - A URL that Twilio will request if an error occurs - requesting or executing the TwiML at Url. - - - FallbackMethod - The HTTP method that Twilio should use to request the - FallbackUrl. Must be either GET or POST. Defaults to POST. - - - StatusCallback - A URL that Twilio will request when the call ends to - notify your app. - - - StatusCallbackMethod - The HTTP method Twilio should use when requesting the - above URL. Defaults to POST. - - - - -
- - -
- Modifying Live Calls - Example - - - - - - - - CAfa51b104354440b09213d04752f50271 - 2013-11-01T03:41:14.488-06:00 - 2013-11-01T03:41:14.488-06:00 - - ACae6e420f425248d6a26948c17a9e2acf - alice - +16175551212 - - ..... TRUNCATED - - -You can still redirect the current call back to the previous application - - -curl -X POST http://ACae6e420f425248d6f8c12cc7b8f8423e5c38b035249166@127.0.0.1:8080/restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Calls/CAfa51b104354440b09213d04752f50271 -d "Url=http://127.0.0.1:8080/restcomm/demos/hello-play.xml" - -The output showing the same Call ID - - - - CAfa51b104354440b09213d04752f50271 - 2013-11-01T03:41:14.488-06:00 - 2013-11-01T03:41:14.488-06:00 - - ACae6e420f425248d6a26948c17a9e2acf - alice - +16175551212 - - ..... TRUNCATED - - -You can end the call using the Status=completed command as shown below - -curl -X POST http://ACae6e420f425248d6f8c12cc7b8f8423e5c38b035249166@127.0.0.1:8080/restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Calls/CAfa51b104354440b09213d04752f50271 -d "Status=completed" - -The output showing the same Call ID - - - - CAfa51b104354440b09213d04752f50271 - 2013-11-01T03:41:14.488-06:00 - 2013-11-01T03:41:14.488-06:00 - - ACae6e420f425248d6a26948c17a9e2acf - alice - +16175551212 - - ..... TRUNCATED - -]]> - - -
- - -
- - - - -
- - List Filter - - - HTTP GET - The following GET query string parameters allow you to limit - the list returned. Note, parameters are case-sensitive: - - - - Request Parameters - - - - - - Parameter - Description - - - - - To - Only show calls to this phone number or Client identifier. - - - From - Only show calls from this phone number or Client - identifier. - - - Status - Only show calls currently in this status. May be queued, - ringing, in-progress, canceled, completed, failed, busy, or - no-answer. - - - StartTime - Only show calls that started on this date, given as - YYYY-MM-DD. Also supports inequalities, such as - StartTime=YYYY-MM-DD for calls that started at or before - midnight on a date, and StartTime=YYYY-MM-DD for calls that - started at or after midnight on a date. - - - ParentCallSid - Only show calls spawned by the call with this Sid. - - - - -
- - - - Filter using the From parameter - The example below will only return Calls made from client - Alice - - - - - - - The result will be similar to the one below - - - - - - CAc0fb839632cf444f9066876d5de741e0 - 2013-10-18T04:51:47.643-06:00 - 2013-10-18T04:51:49.174-06:00 - - ACae6e420f425248d6a26948c17a9e2acf - 1234 - alice - - completed - 2013-10-18T04:51:47.671-06:00 - 2013-10-18T04:51:49.174-06:00 - 1 - 0.00 - inbound - - 2012-04-24 - - - /2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Calls/CAc0fb839632cf444f9066876d5de741e0 - - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Calls/CAc0fb839632cf444f9066876d5de741e0/Notifications - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Calls/CAc0fb839632cf444f9066876d5de741e0/Recordings - - - - -]]> - - -
- -
- - Paging Information - - - HTTP GET - The following GET query string parameters allow you to limit - the list returned. Note, parameters are case-sensitive: - - - - - Request Parameters - - - - - - Parameter - Description - - - - - Page - The current page number. Zero-indexed, so the first page - is 0. - - - NumPages - The total number of pages. - - - PageSize - How many items are in each page - - - Total - The total number of items in the list. - - - Start - The position in the overall list of the first item in this - page. - - - End - The position in the overall list of the last item in this - page. - - - -
- - - - - - - Example - The command below will return a single item from the list of - calls using the PageSize parameter - - - - - - The result of the PageSize parameter - - - - - - - CA4049cf008d6b4277b92ab863fd4ec7c8 - 2013-10-18T04:49:45.942-06:00 - 2013-10-18T04:49:46.272-06:00 - - ACae6e420f425248d6a26948c17a9e2acf - 1235 - bob - - completed - 2013-10-18T04:49:45.994-06:00 - 2013-10-18T04:49:46.272-06:00 - 0 - 0.00 - inbound - - 2012-04-24 - - - /2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Calls/CA4049cf008d6b4277b92ab863fd4ec7c8 - - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Calls/CA4049cf008d6b4277b92ab863fd4ec7c8/Notifications - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Calls/CA4049cf008d6b4277b92ab863fd4ec7c8/Recordings - - - - - -]]> - - - - - - - Additional Paging Information - The API returns URIs to the next, previous, first and last - pages of the returned list as shown in the table below: - - - - Request Parameters - - - - - - Parameter - Description - - - - - Uri - The URI of the current page. - - - Firstpageuri - The URI for the first page of this list. - - - Nextpageuri - The URI for the next page of this list. - - - Previouspageuri - The URI for the previous page of this list. - - - Lastpageuri - The URI for the last page of this list. - - - - -
- - - - -
- -
- - - - - - -
- SMS Messages - An SMS Message resource represents an inbound or outbound SMS - message. - - SMS Message Resource URI - /2012-04-24/Accounts/{AccountSid}/SMS/Messages/{SMSMessageSid} - - - Resource Properties - - - - - - Property - Description - - - - - Sid - A string that uniquely identifies this SMS Message. - - - DateCreated - The date that this SMS Message was created. - - - DateUpdated - The date that this SMS Message was last updated. - - - DateSent - The date that the SMS was sent or received by RestComm. - - - AccountSid - The unique id of the Account that sent or received this SMS - message. - - - From - The phone number or short code that initiated the message. - - - To - The phone number or short code that received the message. - - - Body - The text body of the SMS message. Up to 160 characters - long. - - - Status - The status of this SMS message. Possible values are queued, - sending, sent, failed, and received. - - - Direction - The direction of this SMS message. Possible values are - incoming, outbound-api, outbound-call. - - - ApiVersion - The API version RestComm used to handle the SMS message. - - - Uri - The URI for this account, relative to - http://localhost:port/restcomm. - - - -
-
- Supported Operations - - HTTP GET - Returns the representation of an SMS Message resource, - including the properties above. - -
-
- SMS Message List Resource - - SMS Message List Resource URI - /2012-04-24/Accounts/{AccountSid}/SMS/Messages - -
- Supported Operations - - HTTP GET - Returns the list representation of all the Call resources for - this Account, including the - properties above. - - - HTTP POST - Sends a new SMS Message and returns the representation of the - SMS Message resource, including - the properties above. Below you - will find a list of required and - optional parameters. - - - Request Parameters - - - - - - Parameter - Description - - - - - From(Required) - A phone number that is enabled for SMS. - - - To(Required) - The destination phone number in E.164 format. - - - Body(Required) - The text of the message you want to send, limited to 160 - characters. - - - Custom headers X- (optional) - Optionally you can provide as many custom headers as you - wish. The custom headers must start with X-, - for example "X-MyCustom-Header=My custom header value" - - - -
- - - - Using custom headers - - These additional headers will be part of the SIP MESSAGE that Restcomm will create and dispatch to the SMS Aggregator. - - - - - Using SMS and making DID calls - - You need to configure Restcomm to send SMS messages and DID phone - calls to a Service Provider for provisioning. In the - restcomm.xml - file, the outbound-proxy-uri and the SMS outbound-endpoint must - point to the Service Provider IP address. You may also decide to - use - Restcomm AMI. - - - - - - Send SMS Messages - - - Note the encoding used - %2B13216549878 - instead of the - +13216549878 - The + sign is encoded to to send SMS from the command line. - - - - - "From" DID number - - The "From" number should be the DID SMS enabled number from - VoIP - Innovations. - - - From the bash terminal, you can run the command below: - - - - - To send the same SMS but this time also provide some additional headers: - - - - - - Get list of SMS Messages - - This will display list of message sent - From the bash terminal, you can run the command below: - - - - -
-
-
-
- Recordings - Recordings are generated when you use the <Record> verb. - Those recordings are hosted with RestComm for you to retrieve. - The - Recordings list resource represents the set of an account's - recordings. - - Recording Resource URI - /2012-04-24/Accounts/{AccountSid}/Recordings/{RecordingSid} - - To download the audio file just append .wav after the - RecordingSid. - - Resource Properties - - - - - - Property - Description - - - - - Sid - A string that uniquely identifies this recording. - - - DateCreated - The date that this recording was created. - - - DateUpdated - The date that this recording was last updated. - - - AccountSid - The unique id of the Account that created this recording. - - - CallSid - The unique id of the call during which the recording was - made. - - - Duration - The length of the recording, in seconds. - - - ApiVersion - The API version in use during the recording. - - - Uri - The URI for this account, relative to - http://localhost:port/restcomm. - - - -
-
- Supported Operations - - HTTP GET - Returns the representation of a Recording resource, including - the properties above. - - - HTTP DELETE - Removes the recording from the account. - -
-
- Recording List Resource - - Recording List Resource URI - /2012-04-24/Accounts/{AccountSid}/Recordings - -
- Supported Operations - - HTTP GET - Returns the list representation of all the Recording - resources for this Account, including the - properties above. - - - - - How to Record a Message - - Go to the Advanced Chapter under - - section to learn how to record a message. - - - - - - Get List of Recordings - - The list of recorded wav files can be found in the directory - $RESTCOMM_HOME/standalone/deployments/restcomm.war/recordings/ - - From the bash terminal, you can run the command below: - - - - - -
-
-
-
- Transcriptions - A Transcription resource represents a transcription of a - recording. A transcription is a text version of - a recording produced - using automatic speech recognition. - - Transcription Resource URI - /2012-04-24/Accounts/{AccountSid}/Transcriptions/{TranscriptionSid} - - - Resource Properties - - - - - - Property - Description - - - - - Sid - A string that uniquely identifies this transcription. - - - DateCreated - The date that this transcription was created. - - - DateUpdated - The date that this transcription was last updated. - - - AccountSid - The unique id of the Account that created this - transcription. - - - Status - A string representing the status of the transcription. - Possible values are in-progress, completed, - and failed. - - - RecordingSid - The unique id of the Recording this Transcription was made - of. - - - Duration - The duration of the transcribed audio, in seconds. - - - TranscriptionText - The text content of the transcription. - - - Uri - The URI for this account, relative to - http://localhost:port/restcomm. - - - -
-
- Supported Operations - - HTTP GET - Returns the representation of a Transcription resource, - including the properties above. - - - HTTP DELETE - Removes the Transcription from the account. - -
-
- Transcription List Resource - - Transcription List Resource URI - /2012-04-24/Accounts/{AccountSid}/Transcriptions - -
- Supported Operations - - HTTP GET - Returns the list representation of all the Transcription - resources for this Account, including the - properties above. - -
-
-
-
- Notifications - A Notification resource represents a single log entry made by - RestComm while handling your calls or your use of the - Restful APIs. It - is very useful for debugging purposes. The Notifications - list resource - represents the set of notifications - generated for an account. - - Notification Resource URI - /2012-04-24/Accounts/{AccountSid}/Notifications/{NotificationSid} - - - Resource Properties - - - - - - Property - Description - - - - - Sid - A string that uniquely identifies this transcription. - - - DateCreated - The date that this transcription was created. - - - DateUpdated - The date that this transcription was last updated. - - - AccountSid - The unique id of the Account that created this - transcription. - - - CallSid - CallSid is the unique id of the call during which the - notification was generated. Empty if the - notification was - generated by the Restful APIs without regard to a specific - phone - call. - - - ApiVersion - The RestComm API version in use when this notification was - generated. May be empty for events - that don't have a specific API - version. - - - Log - An integer log level corresponding to the type of - notification: 0 is ERROR, 1 is WARNING. - - - ErrorCode - A unique error code for the error condition. You can lookup - errors, in our Error Dictionary. - - - MoreInfo - A URL for more information about the error condition. The - URL is a page in our Error Dictionary. - - - MessageText - The text for the notification. - - - MessageDate - The date the notification was actually generated - - - RequestUrl - The URL of the resource that caused the notification to be - generated. - - - RequestMethod - The HTTP method in use for the request that caused the - notification to be generated. - - - RequestVariables - The HTTP GET or POST variables that RestComm generated and - sent to your server. Also, if the - notification was generated by - the Restful APIs, this field will include any - HTTP POST or PUT - variables - you sent. - - - ResponseHeaders - The HTTP headers returned by your server. - - - ResponseBody - The HTTP body returned by your server. - - - Uri - The URI for this account, relative to - http://localhost:port/restcomm. - - - -
-
- Supported Operations - - HTTP GET - Returns the representation of a Notification resource, - including the properties above. - -
-
- Notification List Resource - - Notification List Resource URI - /2012-04-24/Accounts/{AccountSid}/Notifications - -
- Supported Operations - - HTTP GET - Returns the list representation of all the Notification - resources for this Account, including the - properties above. - -
-
-
-
- OutboundProxy - Using Outbound proxy endpoint you can get the details of Primary - and Backup outbound proxy, - get the current active outbound proxy and - also switch outbound proxy. - - OutboundProxy Resource URI - /2012-04-24//Accounts/{accountSid}/OutboundProxy - - -
- Supported Operations - - HTTP GET - Returns the list of outbound proxies. - - - HTTP GET - /2012-04-24//Accounts/{accountSid}/OutboundProxy/switchProxy - Switch the outbound proxy and returns the proxy in use - - - HTTP GET - /2012-04-24//Accounts/{accountSid}/OutboundProxy/getActiveProxy - Returns the currently active outbound proxy - -
-
- - - - - -
- OutboundProxy - Using Outbound proxy endpoint you can get the details of Primary - and Backup outbound proxy, - get the current active outbound proxy and - also switch outbound proxy. - - OutboundProxy Resource URI - /2012-04-24//Accounts/{accountSid}/OutboundProxy - - -
- Supported Operations - - HTTP GET - Returns the list of outbound proxies. - - - HTTP GET - /2012-04-24//Accounts/{accountSid}/OutboundProxy/switchProxy - Switch the outbound proxy and returns the proxy in use - - - HTTP GET - /2012-04-24//Accounts/{accountSid}/OutboundProxy/getActiveProxy - Returns the currently active outbound proxy - -
-
- -
- UssdPush - A UssdPush resource represents a message sent from Restcomm to a - USSD gateway. - - UssdPush Resource URI - /2012-04-24/Accounts/{AccountSid}/UssdPush - -
- Supported Operations - - HTTP POST - Sends a Ussd message to the configured USSD Gataeway - - -
- - - Example of UssdPush - - - - - - IP_USSD_GW:PORT_NUMBER - - - - - - - - See below a curl example for the USSD Push: - - curl -X POST -H "application/json" http://ACae6e420f425248d6a26948c17a9e2acf:YOUR_PWD@RESTCOMM_IP_ADDRESS:8080/restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a-d "From=Restcomm" -d "To=123" -d "Url=http://RESTCOMM_IP_ADDRESS:8080/restcomm-rvd/services/apps/YOUR_USSD_APP/controller" - ]]> - - - - -
- - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Services_for_SIP_Servlets.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Services_for_SIP_Servlets.xml deleted file mode 100644 index c89b0d7600..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-Services_for_SIP_Servlets.xml +++ /dev/null @@ -1,2140 +0,0 @@ - - -%BOOK_ENTITIES; -]> - -
- Operating the Example Applications -
- The Location Service - The &PLATFORM_NAME; - Location Service contains a list of mappings of - request URIs to destination addresses. When the Location Service receives - a request, it performs a lookup on that mapping and proxies the - request - simultaneously to the destination address (or addresses) associated with - that URI. - - - The Location Service Mappings Cannot Currently Be Configured - - The Location Service currently performs a lookup on a - hard-coded - list of addresses. This model is evolving toward the eventual use of a - database. - - - Regardless of whether you are using the JBoss Application Server - or - the Tomcat Servlet Container as the Servlets Server, the application, - container and Location Service perform the following steps: - - - - - A user—let us call her Alice—makes a call to - sip:receiver@sip-servlets.com - . The - INVITE - is received by the servlet container, which - then starts the Location Service. - - - - - The Location Service, using non-SIP means, determines that the - callee (i.e. the receiver) is registered at two locations, - identified - by the two SIP URIs, - sip:receiver@127.0.0.1:5090 - and - sip:receiver@127.0.0.1:6090 - . - - - - The Location Service proxies to those two destinations in - parallel, without record-routing, and without making use of - supervised - mode. - - - - - One of the destinations returns a - 200 OK - status code; the second proxy is then canceled. - - - - - The - 200 OK - is forwarded to Alice, and call - setup is completed as usual. - - - - Here is the current list of hard-coded contacts and their - location - URIs: - - - sip:receiver@sip-servlets.com - - - sip:receiver@127.0.0.1:5090 - - - - - sip:receiver@127.0.0.1:6090 - - - - -
- The Location Service: Installing, Configuring and Running - -   - - Pre-Install Requirements and Prerequisites - The following requirements must be met before installation can - begin. - - - - Software Prerequisites - - Either - an &SHORT_PLATFORM_NAME; - for JBoss or an &SHORT_PLATFORM_NAME; - for Tomcat Installation - - - The - Location Service requires either an &SHORT_PLATFORM_NAME; - for JBoss or an - &SHORT_PLATFORM_NAME; - for Tomcat binary installation. - - You - can find detailed instructions on installing &SHORT_PLATFORM_NAME; - for - JBoss here: - - . - - You - can find detailed instructions on installing &SHORT_PLATFORM_NAME; - for - Tomcat here: - - . - - - - - - Downloading - - The Location Service is comprised of two archive files, a Web - Archive (WAR) and a Default Application Router (DAR) configuration - file, which you need to add to your installed SIP Servlets Server. - For - more information about WAR files, refer to the - JBoss Application Server Administration and Development Guide - - . For - more information about DAR files, refer to the - JSR 289 spec, Appendix C - . - - - Download - the Location Service's WAR file from here: - - . - - Download - the Location Service's DAR file from here: - - . - - - Installing - - Both the - location-service-1.4.war - WAR file - and the - locationservice-dar.properties - DAR file - that you downloaded should be placed into different directories in - your SIP Servlet Server installation hierarchy. Which directory - depends on whether you are using the Location Service with &SHORT_PLATFORM_NAME; - for - JBoss or with &SHORT_PLATFORM_NAME; - for Tomcat: - - - - - &SHORT_PLATFORM_NAME; - for JBoss - - - - Place - location-service-1.4.war - into the - - JBOSS_HOME - /server/default/deploy/ - - directory, and - locationservice-dar.properties - into the - - JBOSS_HOME - /server/default/conf/dars/ - - directory. - - - - - &SHORT_PLATFORM_NAME; - for Tomcat - - - - Place - location-service-1.4.war - into the - - CATALINA_HOME - /webapps/ - - directory, and - locationservice-dar.properties - into the - - CATALINA_HOME - /conf/dars/ - - directory. - - - - - - Configuring - - The - darConfigurationFileLocation - attribute of - the - Service - element must be set to the value - conf/dars/locationservice-dar.properties - . The - instructions are given below by SIP Servlets Server type: - - - - - &SHORT_PLATFORM_NAME; - for JBoss - - - - Open the - - JBOSS_HOME - /server/default/deploy/jboss-web.deployer/server.xml - - configuration file and find the - Service - element. Add an attribute to it called - darConfigurationFileLocation - , and set it to - conf/dars/locationservice-dar.properties - - : - - - Editing &SHORT_PLATFORM_NAME; - for JBoss's server.xml for the Location Service - - <Service - name="jboss.web" - className="org.mobicents.servlet.sip.startup.SipStandardService" - sipApplicationDispatcherClassName="org.mobicents.servlet.sip.core.SipApplicationDispatcherImpl" - darConfigurationFileLocation="conf/dars/locationservice-dar.properties" - sipStackPropertiesFile="conf/mss-sip-stack.properties"> - - - Make sure that the configuration file only contains one - darConfigurationFileLocation - attribute: your - new one. - - - - - &SHORT_PLATFORM_NAME; - for Tomcat - - - - Open the - - CATALINA_HOME - /conf/server.xml - - configuration file and find the - Service - element. Add an attribute to it called - darConfigurationFileLocation - , and set it to - conf/dars/locationservice-dar.properties - : - - - Editing &SHORT_PLATFORM_NAME; - for Tomcat's server.xml for the Location Service - - <Service - name="Sip-Servlets" - className="org.mobicents.servlet.sip.startup.SipStandardService" - sipApplicationDispatcherClassName="org.mobicents.servlet.sip.core.SipApplicationDispatcherImpl" - darConfigurationFileLocation="conf/dars/locationservice-dar.properties" - sipStackPropertiesFile="conf/mss-sip-stack.properties"> - - - Make sure that the configuration file only contains one - darConfigurationFileLocation - attribute: your - new one. - - - - - - Running - - Once the WAR and DAR files have been placed in the right - directories, and the JBoss Application Server or Tomcat Servlet - Container knows where to find them (which you specified in a - server.xml - file), then you should go ahead and - run the SIP Servlets Server. - - - - To learn how to run the SIP Servlets-enabled JBoss Application - Server, refer to - - . - - - To learn how to run the SIP Servlets-enabled Tomcat Container, - refer to - - . - - - Testing - The following procedure shows how to test the Location - Service. - - - - - <step> - <para> - Start two SIP soft-phones. The first phone should be set up as - <userinput>sip:receiver@sip-servlets.com</userinput> - at the IP address - <userinput>127.0.0.1</userinput> - on port - <userinput>5090</userinput> - . The second phone can be set up in any - way you like. Note that the SIP phones do not have to be - registered. - </para> - </step> - <step> - <para> - Using the second phone, make a call to - <literal>sip:receiver@sip-servlets.com</literal> - . If the Location - Service has been set up correctly and is running, the first phone—as - the receiver or callee—should now be ringing. - </para> - </step> - </procedure> - <!--<formalpara id="sfss-binary-Location_Service-Using"> <title>Using -   --> - - Stopping - - To learn how to stop the SIP Servlets-enabled JBoss Application - Server, refer to - - . - - - - To learn how to stop the SIP Servlets-enabled Tomcat Container, - refer to - - . - - - Uninstalling - - Unless disk space is at a premium, there is usually no need to - uninstall the Location Service. However, if you will not be using - it - again, you may want to unset or reset the - darConfigurationFileLocation - attribute of the - Service - element, which you set in the - server.xml - configuration file in - - . - - - - You may also wish to delete the WAR and DAR files for the Location - Service, which you installed in - - . - -
-
-
- The Diameter Event-Changing Service - The Diameter Event-Changing Service is based on the Location - Service, which performs call-charging at a fixed rate. Upon the - initiation - of a call, a debit of €10.00 occurs. In the cases of a call being - rejected - or the caller disconnecting (hanging up) before an answer is received, - the - caller's account is refunded. - Note - that an &SHORT_PLATFORM_NAME; - for JBoss installation is required to run this - example; it will not work with &SHORT_PLATFORM_NAME; - for Tomcat. - - Provided here is a step-by-step description of the procedure as - performed by the application and container: - - - Diameter Event-Changing Service Step-By-Step - - - - A user, Alice, makes a call to - sip:receiver@sip-servlets.com - . The - INVITE - is received by the servlet container, which - sends a request to debit Alice's account to the Charging Server. - The - servlet container then invokes the location service. - - - - - - The Location Service determines, without using the SIP protocol - itself, where the callee—or receiver—is registered. The callee may - be - registered at two locations identified by two SIP URIs: - sip:receiver@127.0.0.1:5090 - and - sip:receiver@127.0.0.1:6090 - . - - - - - The Location Service proxies to those two destinations - simultaneously, without record-routing and without using the - supervised - mode. - - - - - - One of the destinations returns - 200 (OK) - , and - so the container cancels the other. - - - - - - The - 200 (OK) - is forwarded upstream to Alice, - and the call setup is carried out as usual. - - - - - If none of the registered destinations accepts the - call, a Diameter Accounting-Request for refund is sent to the Diameter - Charging Server in order to debit the already-credited €10.00 - - - - -
- Diameter Event-Changing Service: Installing, Configuring and - Running - Preparing - your &SHORT_PLATFORM_NAME; - for JBoss server to run the Diameter - Event-Changing example requires downloading a WAR archive, a DAR - archive, the Ericsson Charging Emulator, setting an attribute in - JBoss's - server.xml - configuration file, and then running - JBoss AS. Detailed instructions follow. - - - Pre-Install Requirements and Prerequisites - The following requirements must be met before installation can - begin. - - - - Software Prerequisites - - One &SHORT_PLATFORM_NAME; - for JBoss Installation - - - Before - proceeding, you should follow the instructions for - installing, configuring, running and testing &SHORT_PLATFORM_NAME; - for JBoss from - the binary distribution. - - - - - - Downloading - The following procedure describes how to download the required - files. - - - - - First, download the Web Application Archive - ( - WAR - ) file corresponding to this example, the - current version of which is named - diameter-event-charging-1.0.war - , from - - . - - - - - Secondly, download the corresponding Disk Archive - ( - DAR - ) configuration file here: - - . - - - - - Finally, you will need to download the Ericsson Charging - Emulator, version 1.0, from - - . - - - - - Installing - The following procedure describes how to install the - downloaded files. - - - - - Place the - diameter-event-charging-1.0.war - WAR archive into the - $JBOSS_HOME/server/<profile>/deploy - directory. - - - - - Place the - diametereventcharging-dar.properties - DAR file - in your - - $JBOSS_HOME/server/<profile>/conf/dars - - directory. - - - - - Finally, open the terminal, move into the directory to which - you downloaded the Ericsson Charging SDK (for the sake of this - example, we will call this directory - charging_sdk), - and then unzip the downloaded - zip file (you can use Java's - - jar - - - command for this: - - ~]$ cd charging_sdk -charging_sdk]$ jar -xvf ChargingSDK-1_0_D31E.zip - Alternatively, - you can use Linux's - unzip - command to do the dirty work: - - charging_sdk]$ unzip ChargingSDK-1_0_D31E.zip - - - - Configuring - - To configure the server for the Event-Changing example, simply - open the - server.xml - configuration file in your - server's - - $JBOSS_HOME/server/<profile>/deploy/jboss-web.deployer/ - - directory, and edit the value of the - darConfigurationFileLocation - attribute of the - Service - element so that it is - conf/dars/mobicents-dar.properties - . - - - - Editing the darConfigurationFileLocation Attribute of the - Service Tag - ... - -<Service name="jboss.web" - className="org.mobicents.servlet.sip.startup.SipStandardService" - sipApplicationDispatcherClassName="org.mobicents.servlet.sip.core.SipApplicationDispatcherImpl" - sipApplicationRouterClassName="org.mobicents.servlet.sip.router.DefaultApplicationRouter" - darConfigurationFileLocation="conf/dars/mobicents-dar.properties" - sipStackPropertiesFile="conf/mss-sip-stack.properties"> - -... - - - - - Running - The following procedure describes how to run the Diameter - Event-Changing Service. - - - Diameter Event-Changing Service - - First, - you should run your &SHORT_PLATFORM_NAME; - for JBoss server. For - instructions on doing so, refer to - - . - - - - - Then, run the Ericsson Charging Emulator. Open a terminal, - change the working directory to the location of the unzipped - Charging Emulator files (in - ChargingSDK-1_0_D31E - or a similarly-named - directory), and run it with the - - java -jar - PPSDiamEmul.jar - - command: - - ~]$ java -jar PPSDiamEmul.jar - - - - Using - Using the Event-Changing service means, firstly, inserting - some - parameters into the Charging Emulator, and then, by using two SIP - (soft)phones, calling one with the other. The following sequential - instructions show you how. - - - - SIP (Soft)Phone? Which? - The &PLATFORM_NAME; - team recommends one of the following SIP phones, - and has found that they work well: the 3CX Phone, the SJ Phone or the - WengoPhone. - - - - Using the Diameter Event-Changing Service - - Configure the Ericsson SDK Charging Emulator - - Once you have started the Charging Emulator, you should - configure it exactly as portrayed in - - . - -
- Configuring the Charging Emulator - - - - - -
- - - - Set the - Peer ID - to: - aaa://127.0.0.1:21812 - - - - - Set the - Realm - to: - mobicents.org - - - - - Set the - Host IP - to: - 127.0.0.1 - - - - - - - Start two SIP (soft)phones. You should set the first phone up - with the following parameters: - sip:receiver@sip-servlets - on IP address - 127.0.0.1 - on port - 5090 - . The other phone can be set up any way - you like. - - - - - Before making a call, open the - - Config - Options - - dialog window, as shown in the image. - -
- Configuring Accounts in the Charging Emulator - - - - - -
- - - In the - Account Configuration - window of - the Charging Emulator, you can see the user's balances. Select a - user to watch the balance. You can also stretch the window - lengthwise to view the user's transaction history. - - - - - Time to call! From the second, - any-configuration - phone, make a call to - sip:receiver@sip-servlets.com - . Upon doing so, the - other phone should ring or signal that it is being contacted - . - - - - - You should be able to see a request—immediately following the - invite and before the other party (i.e. you) accepts or rejects - the - call—sent to the Charging Emulator. That is when the debit of the - user's account is made. In the case that the call is - rejected, or - the caller gives up, a second, new Diameter request is sent to - refund the initial amount charged by the call. On the other hand, if - the call is accepted, nothing else related to Diameter happens, and - no second request takes place. - - Please note that this is not the correct way to do - charging, as Diameter provides other means, such as unit - reservation. However, for the purpose of a demonstration it is - sufficient to show the debit and follow-up credit working. Also, - this is a fixed-price call, regardless of the duration. Charging - can, of course, be configured so that it is time-based. - - - - -
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- The Call-Blocking Service - The &PLATFORM_NAME; - Call-Blocking Service, upon receiving an - INVITE - request, checks to see whether the sender's - address is a blocked contact. If so, it returns a - FORBIDDEN - reply; otherwise, call setup proceeds as - normal. - - - Blocked Contacts Cannot Currently Be Configured - Blocked contacts are currently hard-coded addresses. This model - is - evolving towards the eventual use of a database. - - - Here is the current hard-coded list of blocked contacts: - - - - sip:blocked-sender@sip-servlets.com - - - - - sip:blocked-sender@127.0.0.1 - - - - -
- The Call-Blocking Service: Installing, Configuring and Running - -   - - Pre-Install Requirements and Prerequisites - The following requirements must be met before installation can - begin. - - - - Software Prerequisites - - Either - an &SHORT_PLATFORM_NAME; - for JBoss or an &SHORT_PLATFORM_NAME; - for Tomcat Installation - - - The - Call-Blocking Service requires either an &SHORT_PLATFORM_NAME; - for JBoss - or an &SHORT_PLATFORM_NAME; - for Tomcat binary installation. - - You - can find detailed instructions on installing &SHORT_PLATFORM_NAME; - for - JBoss here: - - . - - You - can find detailed instructions on installing &SHORT_PLATFORM_NAME; - for - Tomcat here: - - . - - - - - - Downloading - - The Call-Blocking Service is comprised of two archive files, a - Web Archive (WAR) and a Default Application Router (DAR) configuration - file, which you need to add to your installed SIP Servlets Server. - For - more information about WAR files, refer to the - JBoss Application Server Administration and Development Guide - - . For - more information about DAR files, refer to the - JSR 289 spec, Appendix C - . - - - Download - the Call-Blocking Service's WAR file from here: - - . - - Download - the Call-Blocking Service's DAR file from here: - - . - - - Installing - - Both the - call-blocking-1.4.war - WAR file and - the - call-blocking-servlet-dar.properties - DAR file - that you downloaded should be placed into different directories in - your SIP Servlet Server installation hierarchy. Which directory depends - on whether you are using the Call-Blocking Service with &SHORT_PLATFORM_NAME; - for JBoss - or with &SHORT_PLATFORM_NAME; - for Tomcat: - - - - - &SHORT_PLATFORM_NAME; - for JBoss - - - - Place - call-blocking-1.4.war - into the - - JBOSS_HOME - /server/default/deploy/ - - directory, and - call-blocking-servlet-dar.properties - into the - - JBOSS_HOME - /server/default/conf/dars/ - - directory. - - - - - &SHORT_PLATFORM_NAME; - for Tomcat - - - - Place - call-blocking-servlet-dar.properties - into the - - CATALINA_HOME - /webapps/ - - directory, and - call-blocking-servlet-dar.properties - into the - - CATALINA_HOME - /conf/dars/ - - directory. - - - - - - Configuring - - The - darConfigurationFileLocation - attribute of - the - Service - element must be set to the value - conf/dars/call-blocking-servlet-dar.properties - - . - The instructions for doing so are given below by SIP Servlets Server - type: - - - - - &SHORT_PLATFORM_NAME; - for JBoss - - - - Open the - - JBOSS_HOME - /server/default/deploy/jboss-web.deployer/server.xml - - configuration file and find the - Service - element. Add an attribute to it called - darConfigurationFileLocation - , and set it to - conf/dars/call-blocking-servlet-dar.properties - - : - - - Editing &SHORT_PLATFORM_NAME; - for JBoss's server.xml for the Call-Blocking Service - - <Service - name="jboss.web" - className="org.mobicents.servlet.sip.startup.SipStandardService" - sipApplicationDispatcherClassName="org.mobicents.servlet.sip.core.SipApplicationDispatcherImpl" - darConfigurationFileLocation="conf/dars/call-blocking-servlet-dar.properties" - sipStackPropertiesFile="conf/mss-sip-stack.properties"> - - - Make sure that the configuration file only contains one - darConfigurationFileLocation - attribute: your - new one. - - - - - &SHORT_PLATFORM_NAME; - for Tomcat - - - - Open the - - CATALINA_HOME - /conf/server.xml - - configuration file and find the - Service - element. Add an attribute to it called - darConfigurationFileLocation - , and set it to - conf/dars/call-blocking-servlet-dar.properties - - : - - - Editing &SHORT_PLATFORM_NAME; - for Tomcat's server.xml for the Call-Blocking Service - - <Service - name="Sip-Servlets" - className="org.mobicents.servlet.sip.startup.SipStandardService" - sipApplicationDispatcherClassName="org.mobicents.servlet.sip.core.SipApplicationDispatcherImpl" - darConfigurationFileLocation="conf/dars/call-blocking-servlet-dar.properties" - sipStackPropertiesFile="conf/mss-sip-stack.properties"> - - - Make sure that the configuration file only contains one - darConfigurationFileLocation - attribute: your - new one. - - - - - - Running - - Once the WAR and DAR files have been placed in the right - directories, and the JBoss Application Server or Tomcat Servlet - Container knows where to find them (which you specified in a - server.xml - file), then you should go ahead and - run the SIP Servlets Server. - - - - To learn how to run the SIP Servlets-enabled JBoss Application - Server, refer to - - . - - - To learn how to run the SIP Servlets-enabled Tomcat Container, - refer to - - . - - - Testing - The following procedure shows how to test the Call-Blocking - Service. - - - - Testing the Call Blocking Service - - - Start a SIP softphone of your choice. The account name should - be - blocked-sender - . The - From Header - should list one of the following addresses: - sip:blocked-sender@sip-servlets.com - or - sip:blocked-sender@127.0.0.1 - . The SIP - softphone does not need to be registered. - - - - - Make a call to any address, and you should receive a - FORBIDDEN - response. - - - - - - Stopping - - To learn how to stop the SIP Servlets-enabled JBoss Application - Server, refer to - - . - - - - To learn how to stop the SIP Servlets-enabled Tomcat Container, - refer to - - . - - - Uninstalling - - Unless disk space is at a premium, there is usually no need to - uninstall the Call-Blocking Service. However, if you will not be - using - it again, you may want to unset or reset the - darConfigurationFileLocation - attribute of the - Service - element, which you set in the - server.xml - configuration file in - - . - - - - You may also wish to delete the WAR and DAR files for the - Call-Blocking Service, which you installed in - - . - -
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- The Call-Forwarding Service - The &PLATFORM_NAME; - Call-Forwarding Service, upon receiving an - INVITE - request, checks to see whether the sender's - address is among those in a list of addresses which need to be forwarded. - If so, then the Call-Forwarding Service acts as a Back-to-Back User - Agent - (B2BUA), and creates a new call leg to the destination. When the response - is received from the new call leg, it sends it an acknowledgment - ( - ACK - ) and then responds to the original caller. If, on - the other hand, the server does not receive an - ACK - , - then it tears down the new call leg with a - BYE - . Once - the - BYE - is received, then it answers - OK - directly and sends the - BYE - to - the new call leg. - - - Contacts to Forward Cannot Currently Be Configured - Contacts to forward are currently hard-coded addresses. This - model - is evolving toward the eventual use of a database. - - - Here is the current hard-coded list of contacts to forward: - - - - - sip:receiver@sip-servlets.com - - - - - sip:receiver@127.0.0.1 - - - - -
- The Call-Forwarding Service: Installing, Configuring and - Running -   - - Pre-Install Requirements and Prerequisites - The following requirements must be met before installation can - begin. - - - - Software Prerequisites - - Either - an &SHORT_PLATFORM_NAME; - for JBoss or an &SHORT_PLATFORM_NAME; - for Tomcat Installation - - - The - Call-Forwarding Service requires either an &SHORT_PLATFORM_NAME; - for JBoss - or an &SHORT_PLATFORM_NAME; - for Tomcat binary installation. - - You - can find detailed instructions on installing &SHORT_PLATFORM_NAME; - for - JBoss here: - - . - - You - can find detailed instructions on installing &SHORT_PLATFORM_NAME; - for - Tomcat here: - - . - - - - - - Downloading - - The Call-Forwarding Service is comprised of two archive files, a - Web Archive (WAR) and a Data Archive (DAR), which you need to add - to - your installed SIP Servlets Server. For more information about WAR and - DAR files, refer to the - JBoss Application Server Administration and Development Guide - - . - - - Download - the Call-Forwarding Service's WAR file from here: - - . - - Download - the Call-Forwarding Service's DAR file from here: - - . - - - Installing - - Both the - call-forwarding-1.4.war - WAR file - and the - call-forwarding-servlet-dar.properties - DAR file that you downloaded should be placed into different - directories in your SIP Servlet Server installation hierarchy. - Which - directory depends on whether you are using the Call-Forwarding Service - with &SHORT_PLATFORM_NAME; - for JBoss or with &SHORT_PLATFORM_NAME; - for Tomcat: - - - - - &SHORT_PLATFORM_NAME; - for JBoss - - - - Place - call-forwarding-1.4.war - into the - - JBOSS_HOME - /server/default/deploy/ - - directory, and - call-forwarding-servlet-dar.properties - into - the - - JBOSS_HOME - /server/default/conf/dars/ - - directory. - - - - - &SHORT_PLATFORM_NAME; - for Tomcat - - - - Place - call-forwarding-1.4.war - into the - - CATALINA_HOME - /webapps/ - - directory, and - call-forwarding-servlet-dar.properties - into - the - - CATALINA_HOME - /conf/dars/ - - directory. - - - - - - Configuring - - The - darConfigurationFileLocation - attribute of - the - Service - element must be set to the value - conf/dars/call-forwarding-b2bua-servlet-dar.properties - - . - The instructions for doing so are given below by SIP Servlets Server - type: - - - - - &SHORT_PLATFORM_NAME; - for JBoss - - - - Open the - - JBOSS_HOME - /server/default/deploy/jboss-web.deployer/server.xml - - configuration file and find the - Service - element. Add an attribute to it called - darConfigurationFileLocation - , and set it to - conf/dars/call-forwarding-b2bua-servlet-dar.properties - - : - - - Editing &SHORT_PLATFORM_NAME; - for JBoss's server.xml for the Call-Forwarding Service - - <Service - name="jboss.web" - className="org.mobicents.servlet.sip.startup.SipStandardService" - sipApplicationDispatcherClassName="org.mobicents.servlet.sip.core.SipApplicationDispatcherImpl" - darConfigurationFileLocation="conf/dars/call-forwarding-b2bua-servlet-dar.properties" - sipStackPropertiesFile="conf/mss-sip-stack.properties"> - - - Make sure that the configuration file only contains one - darConfigurationFileLocation - attribute: your - new one. - - - - - &SHORT_PLATFORM_NAME; - for Tomcat - - - - Open the - - CATALINA_HOME - /conf/server.xml - - configuration file and find the - Service - element. Add an attribute to it called - darConfigurationFileLocation - , and set it to - conf/dars/call-forwarding-b2bua-servlet-dar.properties - - : - - - Editing &SHORT_PLATFORM_NAME; - for Tomcat's server.xml for the Call-Forwarding Service - - <Service - name="Sip-Servlets" - className="org.mobicents.servlet.sip.startup.SipStandardService" - sipApplicationDispatcherClassName="org.mobicents.servlet.sip.core.SipApplicationDispatcherImpl" - darConfigurationFileLocation="conf/dars/call-forwarding-b2bua-servlet-dar.properties" - sipStackPropertiesFile="conf/mss-sip-stack.properties"> - - - Make sure that the configuration file only contains one - darConfigurationFileLocation - attribute: your - new one. - - - - - - Running - - Once the WAR and DAR files have been placed in the right - directories, and the JBoss Application Server or Tomcat Servlet - Container knows where to find them (which you specified in a - server.xml - file), then you should go ahead and - run the SIP Servlets Server. - - - - To learn how to run the SIP Servlets-enabled JBoss Application - Server, refer to - - . - - - To learn how to run the SIP Servlets-enabled Tomcat Container, - refer to - - . - - - Testing - The following procedure shows how to test the Call-Forwarding - Service. - - - - - <step> - <para>Start two SIP soft-phones of your choice. Set the account - settings of the first SIP softphone to: - </para> - <itemizedlist> - <listitem> - <para> - Account name: - <userinput>forward-receiver</userinput> - </para> - </listitem> - <listitem> - <para> - IP address: - <userinput>127.0.0.1</userinput> - </para> - </listitem> - <listitem> - <para> - Port: - <userinput>5090</userinput> - </para> - </listitem> - </itemizedlist> - <para>Neither of the SIP soft-phones needs to be registered.</para> - </step> - <step> - <para> - From the second phone, make a call to - <literal>sip:receiver@sip-servlets.com</literal>. - The first phone, - "forward-receiver", should now be ringing. - </para> - </step> - </procedure> - <!--<formalpara id="sfss-binary-Call-Forwarding_Service-Using"> <title>Using -   --> - - Stopping - - To learn how to stop the SIP Servlets-enabled JBoss Application - Server, refer to - - . - - - - To learn how to stop the SIP Servlets-enabled Tomcat Container, - refer to - - . - - - Uninstalling - - Unless disk space is at a premium, there is usually no need to - uninstall the Call-Forwarding Service. However, if you will not be - using it again, you may want to unset or reset the - darConfigurationFileLocation - attribute of the - Service - element, which you set in the - server.xml - configuration file in - - . - - - - You may also wish to delete the WAR and DAR files for the - Call-Forwarding Service, which you installed in - - . - -
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- The Call-Controller Service - The Call-Controller service is a composition of two other - services: - Call-Blocking and Call-Forwarding. Essentially, it performs the services - of both call-forwarding and call-blocking. - - - - - To learn about how the Call-Blocking service works, refer to - - . - - - - - To learn about how the Call-Forwarding service works, refer to - - . - - - - - Blocked Contacts and Contacts to Forward Cannot Currently Be - Configured - Both the list of blocked contacts and the list of contacts to - forward are currently both hard-coded. However, both of those models - are - evolving toward the eventual use of databases. - - - -
- The Call-Controller Service: Installing, Configuring and - Running - The - Call-Controller service requires the two WAR files for the - Call-Blocking and Call-Forwarding services to be placed in the - correct - directory inside your &PLATFORM_NAME; - SIP Servlets Server binary installation. - However, the Call-Controller service does - not - require their corresponding DAR files: you need only to download and - install a DAR file customized for the Call-Controller service. The - instructions below show you how to do precisely this; there is no - need, - therefore, to first install either the Call-Blocking or the - Call-Forwarding services, though it is helpful to at least be familiar - with them. - - - Pre-Install Requirements and Prerequisites - The following requirements must be met before installation can - begin. - - - - Software Prerequisites - - Either - an &SHORT_PLATFORM_NAME; - for JBoss or an &SHORT_PLATFORM_NAME; - for Tomcat Installation - - - The - Call-Controller Service requires either an &SHORT_PLATFORM_NAME; - for JBoss - or an &SHORT_PLATFORM_NAME; - for Tomcat binary installation. - - You - can find detailed instructions on installing &SHORT_PLATFORM_NAME; - for - JBoss here: - - . - - You - can find detailed instructions on installing &SHORT_PLATFORM_NAME; - for - Tomcat here: - - . - - - - - - Downloading - - The Call-Controller Service is comprised of two WAR files, one - for the Call-Forwarding service and one for Call-Blocking, and a - customized Call-Controller DAR file. You do not need to install the - DAR files for the Call-Forwarding or the Call-Blocking services. - For - more information about WAR files, refer to the - JBoss Application Server Administration and Development Guide - - . For - more information about DAR files, refer to the - JSR 289 spec, Appendix C - - - Download - the Call-Blocking Service's WAR file from here: - - . - - Download - the Call-Forwarding Service's WAR file from here: - - . - - Download - the Call-Controller Service's DAR file from here: - - . - - - Installing - - The - call-blocking-1.4.war - , - call-forwarding-1.4.war - and - call-controller-servlet-dar.properties - archive - files that you downloaded should be placed into different directories - in your SIP Servlet Server installation hierarchy. Which directory - depends on whether you are using the Call-Controller Service with &SHORT_PLATFORM_NAME; - for JBoss or with &SHORT_PLATFORM_NAME; - for Tomcat: - - - - - &SHORT_PLATFORM_NAME; - for JBoss - - - - Place - call-blocking-1.4.war - and - call-forwarding-1.4.war - into the - - JBOSS_HOME - /server/default/deploy/ - - directory, and - call-controller-servlet-dar.properties - into - the - - JBOSS_HOME - /server/default/conf/dars/ - - directory. - - - - - &SHORT_PLATFORM_NAME; - for Tomcat - - - - Place - call-blocking-1.4.war - and - call-forwarding-1.4.war - into the - - CATALINA_HOME - /webapps/ - - directory, and - call-controller-servlet-dar.properties - into - the - - CATALINA_HOME - /conf/dars/ - - directory. - - - - - - Configuring - - The - darConfigurationFileLocation - attribute of - the - Service - element must be set to the value - conf/dars/call-controller-servlet-dar.properties - - . - Instructions for doing so are given below by SIP Servlets Server - type: - - - - - &SHORT_PLATFORM_NAME; - for JBoss - - - - Open the - - JBOSS_HOME - /server/default/deploy/jboss-web.deployer/server.xml - - configuration file and find the - Service - element. Add an attribute to it called - darConfigurationFileLocation - , and set it to - conf/dars/call-controller-servlet-dar.properties - - : - - - Editing &SHORT_PLATFORM_NAME; - for JBoss's server.xml for the Call-Controller Service - - <Service - name="jboss.web" - className="org.mobicents.servlet.sip.startup.SipStandardService" - sipApplicationDispatcherClassName="org.mobicents.servlet.sip.core.SipApplicationDispatcherImpl" - darConfigurationFileLocation="conf/dars/call-controller-servlet-dar.properties " - sipStackPropertiesFile="conf/mss-sip-stack.properties"> - - - Make sure that the configuration file only contains one - darConfigurationFileLocation - attribute: your - new one. - - - - - &SHORT_PLATFORM_NAME; - for Tomcat - - - - Open the - - CATALINA_HOME - /conf/server.xml - - configuration file and find the - Service - element. Add an attribute to it called - darConfigurationFileLocation - , and set it to - conf/dars/call-controller-servlet-dar.properties - - : - - - Editing &SHORT_PLATFORM_NAME; - for Tomcat's server.xml for the Call-Controller Service - - <Service - name="Sip-Servlets" - className="org.mobicents.servlet.sip.startup.SipStandardService" - sipApplicationDispatcherClassName="org.mobicents.servlet.sip.core.SipApplicationDispatcherImpl" - darConfigurationFileLocation="conf/dars/call-controller-servlet-dar.properties " - sipStackPropertiesFile="conf/mss-sip-stack.properties"> - - - Make sure that the configuration file only contains one - darConfigurationFileLocation - attribute: your - new one. - - - - - - Running - - Once the WAR and DAR files have been placed in the right - directories, and the JBoss Application Server or Tomcat Servlet - Container knows where to find them (which you specified in a - server.xml - file), then you should go ahead and - run the SIP Servlets Server. - - - - To learn how to run the SIP Servlets-enabled JBoss Application - Server, refer to - - . - - - To learn how to run the SIP Servlets-enabled Tomcat Container, - refer to - - . - - - Testing - Two use-cases can be distinguished for the Call-Controller - service: one in which a call is blocked, and another in which a - call - is forwarded. Therefore, we have two cases for which we can test the - Call-Controller. - - - - Blocking a Call with Call-Controller - - Start two SIP soft-phones of your choice. Set the account - settings of the SIP soft-phones to: - - - Relevant First Softphone Settings - - - Account name: - forward-receiver - - - - - IP address: - 127.0.0.1 - - - - - Port: - 5090 - - - - - Relevant Second Softphone Settings - - - Account name: - blocked-sender - - - - Neither of the SIP soft-phones needs to be registered. - - - - From the second phone, - blocked-sender - , make - a call to - sip:receiver@sip-servlets.com - . You - should receive a - FORBIDDEN - response. - - - - - Forwarding a Call with Call-Controller - - Start two SIP soft-phones of your choice. Set the account - settings of the SIP soft-phones to: - - - Relevant First Softphone Settings - - - Account name: - forward-receiver - - - - - IP address: - 127.0.0.1 - - - - - Port: - 5090 - - - - - Relevant Second Softphone Settings - - - Account name: - forward-sender - - - - Neither of the SIP soft-phones needs to be registered. - - - - From the second softphone, - forward-sender - , - make a call to - sip:receiver@sip-servlets.com - . The - first phone, - forward-receiver - , should now be - ringing. - - - - - - Stopping - - To learn how to stop the SIP Servlets-enabled JBoss Application - Server, refer to - - . - - - - To learn how to stop the SIP Servlets-enabled Tomcat Container, - refer to - - . - - - Uninstalling - - Unless disk space is at a premium, there is usually no need to - uninstall the Call-Controller Service. However, if you will not be - using it again, you may want to unset or reset the - darConfigurationFileLocation - attribute of the - Service - element, which you set in the - server.xml - configuration file in - - . - - - - You may also wish to delete the WAR and DAR files for the - Call-Controller Service, which you installed in - - . - - - - - provides more information about other service examples - available. - - -
-
- -
diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-scopes_and_contexts.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-scopes_and_contexts.xml deleted file mode 100644 index 1f5c67e08c..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/concept-chapter-scopes_and_contexts.xml +++ /dev/null @@ -1,10 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - Scopes and Contexts of the RestComm - - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/images/RestComm-ekiga-sip-phone-demo.png b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/images/RestComm-ekiga-sip-phone-demo.png deleted file mode 100644 index 6b2a4f5174..0000000000 Binary files a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/images/RestComm-ekiga-sip-phone-demo.png and /dev/null differ diff --git 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b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/images/sflphone-config-basic.png deleted file mode 100644 index 36c218a603..0000000000 Binary files a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/images/sflphone-config-basic.png and /dev/null differ diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/task-chapter-examples.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/task-chapter-examples.xml deleted file mode 100644 index a163bbbcf1..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/task-chapter-examples.xml +++ /dev/null @@ -1,94 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - Click2Call Example - With the advent of MSS 2.x, the Mobicents team made available - asynchronous processing of requests feature in the world of - SipServlets and demonstrated this feature with a new version of the - traditional Click2Call example. - Now its time to get Click2Call one step further and demonstrate - the CTF features with it. - In this new version of the Click2Call, the CDI programming model - makes the application significant simpler by using beans as services. - Redundant is the use of the ServletContext to store attributes such as - registered users or calls. This is now being served by the - CallStatusContainer and the SipRegistar modules. Several functions of - the application are being delegated to beans also. Methods that need - to fire an event are using the CDI event producer/observer model - instead of a queue that was placed in the ServletContext and the - SimpleSipServlet class is using the SipRegistar module to register - users. - If you check out the Click2Call example you will notice the - following major changes: - - - - SimpleSipServlet is nothing more than a simple class. No more extend - SipServlet and - @Override - doRequest(SipServletRequest req) and such. All the methods of the - class that needs to get notified for a SIP request or response have - @Observe @SipEvent - and that enough to handle SIP messages. - - - - - Everywhere in the project, when there is a need for the SipFactory - we don’t take it using JNDI anymore. A simple - @Inject - SipFactory sipFactory will do the job since CTF makes available - SipFactory for injection. - - - - Several of the application’s functionality have been moved to - stand alone modules. One of them is the SipRegistar module that will - take care of SIP Registration requests. SimpleSipServlet injects - SipRegistar module and delegates registration requests to it. Please - note that these modules could reside in an external archive jar file - placed in the classpath of the application, thus the beginning of - reusable SIP modules. - - - - You will also notice that SipSession or SipApplicationSession in the - SimpleSipServlet wont be retrieved by the request or the response. - If you include the following in the application - @Inject - SipSession sipSession and - @Inject - SipApplicationSession sipAppSession CTF will make sure you will get - the SipSession and SipApplicationSession from the request or the - response accordingly, depending on the SIP event you observe. - - - - - The drawing below will give you the general idea of the - application’s interactions with the serveral modules and the CTF. - -
- Click2Call using CTF - - - - - -
- - - MSS 2.x comes with Click2CallAsync predeployed, make sure you - undeploy this application and apply the correct dar configuration - before you deploy the new version of Click2Call. - - In the examples of the framework you will find a version of - Click2Call with CTF, suitable for servlet 2.5 containers such as - Tomcat 6.0.29 and JBoss AS5. - - diff --git a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/task-chapter-getting_started.xml b/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/task-chapter-getting_started.xml deleted file mode 100644 index f710d0f8b0..0000000000 --- a/restcomm/restcomm.docs/sources-mobicents/src/main/resources/en-US/task-chapter-getting_started.xml +++ /dev/null @@ -1,284 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - Getting started with CTF - In order to get started with CTF, the only thing you have to do - is to define few maven dependencies in your project and register the - CTF listener in the sip.xml or web.xml or your application. - First dependency is Weld, the CDI reference implementation by - JBoss and more specific the weld-servlet module. - - - org.jboss.weld.servlet - weld-servlet - 1.1.2.Final - - ]]> - - Second is the framework itself: - - - org.mobicents.servlet.sip.ctf - ctf-core - ]]>&CTF_VERSION; - - ]]> - - - - Make sure you use the JBoss public maven repository: - - - jboss-public-repository-group - JBoss Public Maven Repository Group - https://repository.jboss.org/nexus/content/groups/public - default - - true - never - - - true - never - - - ]]> - - - - - The CTF listener that you have to register in your application - is: - - - - org.mobicents.servlet.sip.ctf.environment.servlet.SipServletsListener - - - ]]> - - - To get the source code along with the examples: - - - checkout project from svn repository: svn co - https://mobicents.googlecode.com/svn/trunk/frameworks/sip-servlets/cdi-telco-framework - - - - In the folder core/ctf-core/ you will find the project - responsible to bootstrap the framework, in order to build it: mvn - clean compile install - - - In the folder core/ctf-core-extension/ you will find the Sip - Servlets extensions to CDI, such as CTF annotations, in order to - build it: mvn clean compile install - - - - In the folder examples/ you will find the examples of the framework, - build according to the notes of every example. The examples contain - the following: - - - click2call-servlet3.0-cdi . The Click2Call application with - asynchronous processing of request feature and CTF. Deploy to MSS - 2.x Tomcat 7.0.8 only - - - click2call-servlet25-cdi. The Click2Call application with - CTF. Deploy to Servlet 2.5 container such as MSS 1.x Tomcat - 6.0.29. The example can be deployed to JBoss AS5 also, build - using : mvn clean compile package -Pjboss5 - - - - SipRegistration. - - - SipRegistarClient - SIP registration example for java - servlets 3.0. Deploy to MSS 2.x Tomcat 7.0.8 only. - - - SipRegistarClient25 - SIP registration example for java - servlets 2.5. Deploy to MSS 1.x Tomcat 6.0.29. The example can - be deployed to JBoss AS5 also, build using : mvn clean compile - package -Pjboss5 - - - - - - - - - - To build issue mvn clean compile package for each project and - deploy accordingly - - - - For JBoss AS5 special maven dependency needed for MSS 1.x JBoss AS5 - version for the JBoss Weld dependency, that will be: - - - org.jboss.weld.servlet - weld-servlet - runtime - jboss5 - 1.1.2.Final - - - org.mobicents.servlet.sip.ctf - ctf-core - ]]>&CTF_VERSION; - jboss5 - - ]]> - - - - - CTF have been tested and verified working in the following MSS - containers: - - - MSS 1.6 JBoss AS5 - - - MSS 1.6 Tomcat 6.0.29 - - - MSS 2.0-Branch Tomcat 7.0.8 - - - - - Summary of CTF dependencies to use - - - MSS 1.x on Tomcat 6.x - - - CDI - - - org.jboss.weld.servlet - weld-servlet - 1.1.2.Final - - ]]> - - - CTF - - - org.mobicents.servlet.sip.ctf - ctf-core - ]]>&CTF_VERSION; - - ]]> - - - - - - - MSS 1.x on JBoss AS5 - - - CDI - - - org.jboss.weld.servlet - weld-servlet - runtime - jboss5 - 1.1.2.Final - - ]]> - - - CTF - - - org.mobicents.servlet.sip.ctf - ctf-core - ]]>&CTF_VERSION; - jboss5 - - ]]> - - - - - - - MSS 2.x on Tomcat 7.x - - - CDI - - - org.jboss.weld.servlet - weld-servlet - 1.1.2.Final - - ]]> - - - CTF - - - org.mobicents.servlet.sip.ctf - ctf-core - ]]>&CTF_VERSION; - - ]]> - - - - - - - diff --git a/restcomm/restcomm.docs/sources-telscale/.classpath b/restcomm/restcomm.docs/sources-telscale/.classpath deleted file mode 100644 index cb76c82489..0000000000 --- a/restcomm/restcomm.docs/sources-telscale/.classpath +++ /dev/null @@ -1,27 +0,0 @@ - - - - - - - - - - - - - - - - - - - - - - - - - - - diff --git a/restcomm/restcomm.docs/sources-telscale/.gitignore b/restcomm/restcomm.docs/sources-telscale/.gitignore deleted file mode 100644 index ea8c4bf7f3..0000000000 --- a/restcomm/restcomm.docs/sources-telscale/.gitignore +++ /dev/null @@ -1 +0,0 @@ -/target diff --git a/restcomm/restcomm.docs/sources-telscale/.project b/restcomm/restcomm.docs/sources-telscale/.project deleted file mode 100644 index 5746f5f3fa..0000000000 --- a/restcomm/restcomm.docs/sources-telscale/.project +++ /dev/null @@ -1,29 +0,0 @@ - - - restcomm-docs-sources-telscale - A Restful Communications Framework. NO_M2ECLIPSE_SUPPORT: Project files created with the maven-eclipse-plugin are not supported in M2Eclipse. - - - - - org.eclipse.jdt.core.javabuilder - - - - - net.sf.eclipsecs.core.CheckstyleBuilder - - - - - org.eclipse.m2e.core.maven2Builder - - - - - - org.eclipse.m2e.core.maven2Nature - org.eclipse.jdt.core.javanature - net.sf.eclipsecs.core.CheckstyleNature - - diff --git a/restcomm/restcomm.docs/sources-telscale/.settings/org.eclipse.jdt.core.prefs b/restcomm/restcomm.docs/sources-telscale/.settings/org.eclipse.jdt.core.prefs deleted file mode 100644 index ec4300d5d0..0000000000 --- a/restcomm/restcomm.docs/sources-telscale/.settings/org.eclipse.jdt.core.prefs +++ /dev/null @@ -1,5 +0,0 @@ -eclipse.preferences.version=1 -org.eclipse.jdt.core.compiler.codegen.targetPlatform=1.7 -org.eclipse.jdt.core.compiler.compliance=1.7 -org.eclipse.jdt.core.compiler.problem.forbiddenReference=warning -org.eclipse.jdt.core.compiler.source=1.7 diff --git a/restcomm/restcomm.docs/sources-telscale/.settings/org.eclipse.m2e.core.prefs b/restcomm/restcomm.docs/sources-telscale/.settings/org.eclipse.m2e.core.prefs deleted file mode 100644 index 64e89e5224..0000000000 --- a/restcomm/restcomm.docs/sources-telscale/.settings/org.eclipse.m2e.core.prefs +++ /dev/null @@ -1,5 +0,0 @@ -#Wed Jun 13 10:39:31 EDT 2012 -activeProfiles= -eclipse.preferences.version=1 -resolveWorkspaceProjects=true -version=1 diff --git a/restcomm/restcomm.docs/sources-telscale/Makefile b/restcomm/restcomm.docs/sources-telscale/Makefile deleted file mode 100644 index 3f791eb6d7..0000000000 --- a/restcomm/restcomm.docs/sources-telscale/Makefile +++ /dev/null @@ -1,26 +0,0 @@ -#Makefile for Platform_User_Guide - -XML_LANG = en-US -# DOCNAME = Platform_User_Guide -# PRODUCT = FIX_ME! -BRAND = JBoss -# DOC_TYPE = Set - -# Uncomment below 3 lines for a Distributed Set -#BOOKS = Platform_Installation_Guide JAIN_SLEE_Server_User_Guide -#BOOKS = Platform_Installation_Guide JAIN_SLEE_Server_User_Guide SIP_Servlets_Server_User_Guide Media_Server_User_Guide SIP_Presence_Service_User_Guide -#SET_REPO = https://svn.devel.redhat.com/repos/ecs/JBoss_Communications_Platform/1.2.0/ -#SET_REPO_TYPE = SVN - -#OTHER_LANGS = as-IN bn-IN de-DE es-ES fr-FR gu-IN hi-IN it-IT ja-JP kn-IN ko-KR ml-IN mr-IN or-IN pa-IN pt-BR ru-RU si-LK ta-IN te-IN zh-CN zh-TW - -# Extra Parameters start here - -# Extra Parameters stop here -COMMON_CONFIG = /usr/share/publican -include $(COMMON_CONFIG)/make/Makefile.common - -# Uncomment below 2 lines for a Distributed Set -#pre:: get_books set_books -#clean:: clean_set_books - diff --git a/restcomm/restcomm.docs/sources-telscale/README.txt b/restcomm/restcomm.docs/sources-telscale/README.txt deleted file mode 100644 index d0424ade74..0000000000 --- a/restcomm/restcomm.docs/sources-telscale/README.txt +++ /dev/null @@ -1,7 +0,0 @@ -when mvn clean install has been performed in the parent directory, to validate against publican do : - -cd ./src/main/resources -chmod 777 mkbk Makefile -./mkbk sss jbcp html-single-en-US - -that's it diff --git a/restcomm/restcomm.docs/sources-telscale/mkbk b/restcomm/restcomm.docs/sources-telscale/mkbk deleted file mode 100644 index e8f07671d0..0000000000 --- a/restcomm/restcomm.docs/sources-telscale/mkbk +++ /dev/null @@ -1,144 +0,0 @@ -#! /usr/bin/env bash -# :mode=shellscript: - -# short script for building Mobicents docs and ensuring -# necessary files for JDocBook (all-*) are updated -# for usage, call ./build_docs.bash without any arguments - -# EXAMPLES: - -# mkbk test -# ...means: run xmllint on all books, looking for the first that does not build - -# make CONDITION="mob" html-single-en-US -# BECOMES -# mkbk jss mob html-single-en-US - -# make CONDITION="jbcp" html-single-en-US -# BECOMES -# mkbk jss jbcp html-single-en-US - -# JDocBook Reference (ignore this): -# mvn compile -Denv.DOCNAME="SIP_Servlets_Server_User_Guide" -Phtml_single - -echo "Arg 1 [\$BK] (codename of book OR simply 'test'): $1"; BK=$(echo $1 | tr '[:upper:]' '[:lower:]') -echo "Arg 2 [\$COND] (CONDITION): $2"; COND=$(echo $2 | tr '[:upper:]' '[:lower:]') -echo -n "Arg 3 (pass-through arg; 3rd arg to 'make', if exists): "; THIRD="$3"; echo $THIRD -echo -n "Arg 4 (pass-through arg; 4th arg to 'make', if exists): "; FOURTH="$4"; echo $FOURTH -echo -n "Arg 5 (pass-through arg; 5th arg to 'make', if exists): "; FIFTH="$5"; echo $FIFTH -echo "-------------------------------------------------" - -pre="[MKBK] " - -ALL_BKS="jss ms sss sps pig plat" -ALL_CONDS="mob jbcp" - -PLAT="Platform_User_Guide" -PIG="Platform_Installation_Guide" -JSS="JAIN_SLEE_Server_User_Guide" -MS="Media_Server_User_Guide" -SSS="SIP_Servlets_Server_User_Guide" -SPS="SIP_Presence_Service_User_Guide" - -set_docname() { - case $BK in - "plat") DOCNAME="$PLAT";; - "pig" ) DOCNAME="$PIG";; - "jss" ) DOCNAME="$JSS";; - "ms" ) DOCNAME="$MS";; - "sss") DOCNAME="$SSS";; - "sps") DOCNAME="$SPS";; - esac -} - -test() { - for bk in "jss" "ms" "sss" "sps" "pig" "plat" # $PLAT $PIG $JSS $MS $SPS - do - BK=$bk - set_docname - cmd = "mkbk $bk jbcp xml-en-US &>/dev/null" - - echo -n "$pre"; echo "DOCNAME is $DOCNAME" - echo -n "$pre"; echo "Calling:"; echo "" - echo -n "$pre"; echo " $cmd" - - eval $cmd - - if [ $? -ne 0 ]; then - echo -n "$pre"; echo "$bk is failing to build! Run!" - exit 1 - else - echo -n "$pre"; echo "$DOCNAME builds correctly." - echo -n "$pre"; echo "*****************************" - fi - done - echo -n "$pre"; echo "All books build correctly. Throw a party!" - exit 0 -} - -rm_tmp() { - if [[ -e Makefile ]]; then - if [[ -e pom.xml ]]; then - if [[ -e tmp ]]; then - echo -n "$pre"; echo "Removing 'tmp'..." - rm tmp -fr - fi - fi - fi -} - -build_bk() { - rm_tmp - set_docname - cmd="make DOCNAME=$DOCNAME CONDITION=$COND $THIRD $FOURTH $FIFTH" - echo -n "$pre"; echo "DOCNAME is '$DOCNAME'" - echo -n "$pre"; echo "Calling:"; echo "" - echo -n "$pre"; echo " $cmd" - eval $cmd - - if [ $? -ne 0 ]; then - echo -n "$pre"; echo "$DOCNAME did not build!" - exit 1 - else - if [ "$COND" = "mob" ]; then - echo -n "$pre"; echo "Removing 'en-US/all-$DOCNAME.xml'..." - rm "en-US/all-$DOCNAME.xml" - echo -n "$pre"; echo "Copying 'tmp/en-US/xml/$DOCNAME.xml' to 'en-US/all-$DOCNAME.xml'..." - xmllint --postvalid --noent --xinclude "tmp/en-US/xml/$DOCNAME.xml" > "en-US/all-$DOCNAME.xml" - fi - echo -n "$pre"; echo "$DOCNAME built. Exiting successfully." - exit 0 - fi -} - -# TEST ARGUMENTS -chk_BK_arg() { - if [ "$BK" = "" ]; then - echo -n "$pre"; echo "***************************************" - echo -n "$pre"; echo "Error: the BK argument must either represent a book or be 'test'" - echo -n "$pre"; echo " A possible argument for the BK OPTION is one of : $ALL_BKS" - echo -n "$pre"; echo "***************************************" - exit 1 - fi -} - -chk_COND_arg() { - if [ "$COND" != "mob" -a "$COND" != "jbcp" ]; then - echo -n "$pre"; echo "***************************************" - echo -n "$pre"; echo "Error: the second argument must be one of these conditions: $ALL_CONDS" - echo -n "$pre"; echo "***************************************" - exit 1 - fi -} - -dispatch() { - chk_BK_arg - if [ "$BK" = "test" ]; then - test - fi - chk_COND_arg - - build_bk -} - -dispatch diff --git a/restcomm/restcomm.docs/sources-telscale/pom.xml b/restcomm/restcomm.docs/sources-telscale/pom.xml deleted file mode 100644 index 670a7abe3c..0000000000 --- a/restcomm/restcomm.docs/sources-telscale/pom.xml +++ /dev/null @@ -1,93 +0,0 @@ - - - - 4.0.0 - - - org.restcomm - restcomm-connect-docs - 7.6.0-SNAPSHOT - - - restcomm-connect-docs-sources-telscale - - - TelScale - TelScale RestComm - telscale-restcomm - ${project.version} - ivelin.atanasoff.ivanov (at) gmail.com - jean.deruelle (at) gmail.com - quintana.thomas (at) gmail.com - - - - - - ${basedir}/src/main/resources - true - - - - - org.apache.maven.plugins - maven-dependency-plugin - - - unpack - generate-resources - - unpack - - - - - ${pom.groupId} - restcomm-connect-docs-sources - ${pom.version} - jar - true - ${basedir}/src/main/resources - - - - - - - - maven-antrun-plugin - - - install - - run - - - - - - - - - - - - clean-resources - clean - - run - - - - - - - - - - - - - - - diff --git a/restcomm/restcomm.docs/sources-telscale/publican.cfg b/restcomm/restcomm.docs/sources-telscale/publican.cfg deleted file mode 100644 index 48c3d7b9b1..0000000000 --- a/restcomm/restcomm.docs/sources-telscale/publican.cfg +++ /dev/null @@ -1,7 +0,0 @@ -# Config::Simple 4.59 -# Thu Jan 27 14:23:54 2011 - -debug: 1 -xml_lang: en-US -brand: JBoss - diff --git a/restcomm/restcomm.docs/sources/.classpath b/restcomm/restcomm.docs/sources/.classpath deleted file mode 100644 index cb76c82489..0000000000 --- a/restcomm/restcomm.docs/sources/.classpath +++ /dev/null @@ -1,27 +0,0 @@ - - - - - - - - - - - - - - - - - - - - - - - - - - - diff --git a/restcomm/restcomm.docs/sources/.gitignore b/restcomm/restcomm.docs/sources/.gitignore deleted file mode 100644 index ea8c4bf7f3..0000000000 --- a/restcomm/restcomm.docs/sources/.gitignore +++ /dev/null @@ -1 +0,0 @@ -/target diff --git a/restcomm/restcomm.docs/sources/.project b/restcomm/restcomm.docs/sources/.project deleted file mode 100644 index b1c17375fd..0000000000 --- a/restcomm/restcomm.docs/sources/.project +++ /dev/null @@ -1,29 +0,0 @@ - - - restcomm-docs-sources - A Restful Communications Framework. NO_M2ECLIPSE_SUPPORT: Project files created with the maven-eclipse-plugin are not supported in M2Eclipse. - - - - - org.eclipse.jdt.core.javabuilder - - - - - net.sf.eclipsecs.core.CheckstyleBuilder - - - - - org.eclipse.m2e.core.maven2Builder - - - - - - org.eclipse.m2e.core.maven2Nature - org.eclipse.jdt.core.javanature - net.sf.eclipsecs.core.CheckstyleNature - - diff --git a/restcomm/restcomm.docs/sources/.settings/org.eclipse.jdt.core.prefs b/restcomm/restcomm.docs/sources/.settings/org.eclipse.jdt.core.prefs deleted file mode 100644 index ec4300d5d0..0000000000 --- a/restcomm/restcomm.docs/sources/.settings/org.eclipse.jdt.core.prefs +++ /dev/null @@ -1,5 +0,0 @@ -eclipse.preferences.version=1 -org.eclipse.jdt.core.compiler.codegen.targetPlatform=1.7 -org.eclipse.jdt.core.compiler.compliance=1.7 -org.eclipse.jdt.core.compiler.problem.forbiddenReference=warning -org.eclipse.jdt.core.compiler.source=1.7 diff --git a/restcomm/restcomm.docs/sources/.settings/org.eclipse.m2e.core.prefs b/restcomm/restcomm.docs/sources/.settings/org.eclipse.m2e.core.prefs deleted file mode 100644 index 1bafefd517..0000000000 --- a/restcomm/restcomm.docs/sources/.settings/org.eclipse.m2e.core.prefs +++ /dev/null @@ -1,5 +0,0 @@ -#Thu Nov 01 22:16:21 EDT 2012 -activeProfiles= -eclipse.preferences.version=1 -resolveWorkspaceProjects=true -version=1 diff --git a/restcomm/restcomm.docs/sources/pom.xml b/restcomm/restcomm.docs/sources/pom.xml deleted file mode 100644 index 69e0dea800..0000000000 --- a/restcomm/restcomm.docs/sources/pom.xml +++ /dev/null @@ -1,30 +0,0 @@ - - - - 4.0.0 - - - org.restcomm - restcomm-connect-docs - 7.6.0-SNAPSHOT - - - restcomm-connect-docs-sources - - - RestComm - 2011 - RestComm_User_Guide - Mobicents - - - - - - ${basedir}/src/main/resources - true - - - - - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/Author_Group.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/Author_Group.xml deleted file mode 100644 index 870ca675f2..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/Author_Group.xml +++ /dev/null @@ -1,27 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - Jean - Deruelle - jean.deruelle@gmail.com - - - Thomas - Quintana - quintana.thomas@gmail.com - - - Charles - Roufay - charles.roufay@telestax.com - - - George - Vagenas - gvagenas@telestax.com - - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/Book_Info.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/Book_Info.xml deleted file mode 100644 index 43a8fd81f7..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/Book_Info.xml +++ /dev/null @@ -1,44 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - RestComm User Guide - The Guide to the RestComm - Mobicents Platform - 1.0.0 - 1.0 - 1 - - - RestComm is a carrier-grade open source platform that provides developers the tools - to integrate fax, voice, and SMS functionality in to their own applications with ease. RestComm - is designed to have 100% compatibility with Twilio's APIs allowing easy porting between - platforms. - - Furthermore, the RestComm platform is built on top of the industry leading Mobicents - Sip Servlet Container and Mobicents Media Server providing the robustness and performance - these platforms are already known to deliver. - - - - - &YEAR; - - &HOLDER; - - - - - - - - - - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/Common_Content/Conventions.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/Common_Content/Conventions.xml deleted file mode 100644 index 93fb0518a6..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/Common_Content/Conventions.xml +++ /dev/null @@ -1,334 +0,0 @@ - - -%BOOK_ENTITIES; -]> - -
- Document Conventions - - This manual uses several conventions to highlight certain words - and phrases and draw attention to specific pieces of information. - - - In PDF and paper editions, this manual uses typefaces drawn from the - Liberation Fonts - set. The Liberation Fonts set is also used in HTML editions if the set - is installed on your system. If not, alternative but equivalent - typefaces are displayed. Note: Red Hat Enterprise Linux 5 and later - includes the Liberation Fonts set by default. - -
- Typographic Conventions - - Four typographic conventions are used to call attention to - specific words and phrases. These conventions, and the circumstances - they apply to, are as follows. - - - Mono-spaced Bold - - - Used to highlight system input, including shell commands, file - names and paths. Also used to highlight key caps and - key-combinations. For example: - -
- - To see the contents of the file - my_next_bestselling_novel - in your current working directory, enter the - cat my_next_bestselling_novel - command at the shell prompt and press - Enter - to execute the command. - -
- - The above includes a file name, a shell command and a key cap, - all presented in Mono-spaced Bold and all distinguishable thanks to - context. - - - Key-combinations can be distinguished from key caps by the - hyphen connecting each part of a key-combination. For example: - -
- - Press - Enter - to execute the command. - - - Press - - Ctrl - Alt - F1 - - to switch to the first virtual terminal. Press - - Ctrl - Alt - F7 - - to return to your X-Windows session. - -
- - The first sentence highlights the particular key cap to press. - The second highlights two sets of three key caps, each set pressed - simultaneously. - - - If source code is discussed, class names, methods, functions, - variable names and returned values mentioned within a paragraph will - be presented as above, in - Mono-spaced Bold - . For example: - -
- - File-related classes include - filesystem - for file systems, - file - for files, and - dir - for directories. Each class has its own associated set of - permissions. - -
- - Proportional Bold - - - This denotes words or phrases encountered on a system, including - application names; dialogue box text; labelled buttons; check-box and - radio button labels; menu titles and sub-menu titles. For example: - -
- - Choose - System > Preferences > Mouse - from the main menu bar to launch - Mouse Preferences - . In the - Buttons - tab, click the - Left-handed mouse - check box and click - Close - to switch the primary mouse button from the left to the right - (making the mouse suitable for use in the left hand). - - - To insert a special character into a - gedit - file, choose - Applications > Accessories > Character Map - from the main menu bar. Next, choose - Search > Find… - from the - Character Map - menu bar, type the name of the character in the - Search - field and click - Next - . The character you sought will be highlighted in the - Character Table - . Double-click this highlighted character to place it in the - Text to copy - field and then click the - Copy - button. Now switch back to your document and choose - Edit > Paste - from the - gedit - menu bar. - -
- - The above text includes application names; system-wide menu - names and items; application-specific menu names; and buttons and - text found within a GUI interface, all presented in Proportional Bold - and all distinguishable by context. - - - Note the - > - shorthand used to indicate traversal through a menu and its - sub-menus. This is to avoid the difficult-to-follow 'Select - Mouse - from the - Preferences - sub-menu in the - System - menu of the main menu bar' approach. - - - - Mono-spaced Bold Italic - - or - - Proportional Bold Italic - - - - Whether Mono-spaced Bold or Proportional Bold, the addition of - Italics indicates replaceable or variable text. Italics denotes text - you do not input literally or displayed text that changes depending - on circumstance. For example: - -
- - To connect to a remote machine using ssh, type - - ssh - username - @ - domain.name - - at a shell prompt. If the remote machine is - example.com - and your username on that machine is john, type - ssh john@example.com - . - - - The - - mount -o remount - file-system - - command remounts the named file system. For example, to remount the - /home - file system, the command is - mount -o remount /home - . - - - To see the version of a currently installed package, use the - - rpm -q - package - - command. It will return a result as follows: - - package-version-release - - . - -
- - Note the words in bold italics above — username, - domain.name, file-system, package, version and release. Each word is - a placeholder, either for text you enter when issuing a command or - for text displayed by the system. - - - Aside from standard usage for presenting the title of a work, - italics denotes the first use of a new and important term. For - example: - -
- - When the Apache HTTP Server accepts requests, it dispatches child - processes or threads to handle them. This group of child processes - or threads is known as a - server-pool - . Under Apache HTTP Server 2.0, the responsibility for creating and - maintaining these server-pools has been abstracted to a group of - modules called - Multi-Processing Modules - ( - MPMs - ). Unlike other modules, only one module from the MPM group can be - loaded by the Apache HTTP Server. - -
-
- -
- Pull-quote Conventions - - Two, commonly multi-line, data types are set off visually from - the surrounding text. - - - Output sent to a terminal is set in - Mono-spaced Roman - and presented thus: - - - -books Desktop documentation drafts mss photos stuff svn -books_tests Desktop1 downloads images notes scripts svgs - - - Source-code listings are also set in - Mono-spaced Roman - but are presented and highlighted as follows: - - - -package org.jboss.book.jca.ex1; - -import javax.naming.InitialContext; - -public class ExClient -{ - public static void main(String args[]) - throws Exception - { - InitialContext iniCtx = new InitialContext(); - Object ref = iniCtx.lookup("EchoBean"); - EchoHome home = (EchoHome) ref; - Echo echo = home.create(); - - System.out.println("Created Echo"); - - System.out.println("Echo.echo('Hello') = " + echo.echo("Hello")); - } - -} - -
- -
- Notes and Warnings - - Finally, we use three visual styles to draw attention to - information that might otherwise be overlooked. - - - Note - - A note is a tip or shortcut or alternative approach to the task - at hand. Ignoring a note should have no negative consequences, but - you might miss out on a trick that makes your life easier. - - - - Important - - Important boxes detail things that are easily missed: - configuration changes that only apply to the current session, or - services that need restarting before an update will apply. Ignoring - Important boxes won't cause data loss but may cause irritation - and frustration. - - - - Warning - - A Warning should not be ignored. Ignoring warnings will most - likely cause data loss. - - -
- -
- - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/Common_Content/Feedback.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/Common_Content/Feedback.xml deleted file mode 100644 index f294764c2d..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/Common_Content/Feedback.xml +++ /dev/null @@ -1,35 +0,0 @@ - - -%BOOK_ENTITIES; -]> - -
- - Provide feedback to the authors! - - - feedback - - - - If you find a typographical error in this manual, or if you have - thought of a way to make this manual better, we would love to hear - from you! Please submit a report in Bugzilla: - http://http://code.google.com/p/restcomm/issues/ - against the product - ${product.name} - , or contact the authors. - - - When submitting a bug report, be sure to mention the manual's - identifier: &BOOKID; - - If you have a suggestion for improving the documentation, try to - be as specific as possible when describing it. If you have found an - error, please include the section number and some of the surrounding - text so we can find it easily. - -
- - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/Common_Content/Java_Development_Kit-Installing_Configuring_and_Running.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/Common_Content/Java_Development_Kit-Installing_Configuring_and_Running.xml deleted file mode 100644 index 4aec89aadb..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/Common_Content/Java_Development_Kit-Installing_Configuring_and_Running.xml +++ /dev/null @@ -1,449 +0,0 @@ - - -%BOOK_ENTITIES; - -]> -
- - Java Development Kit ( - <acronym>JDK</acronym> - ): Installing, Configuring and Running - - - The - &PLATFORM_NAME; - Platform - - is written in Java; therefore, before running any - &PLATFORM_NAME; - - server, you must have a working Java Runtime Environment ( - JRE - ) or Java Development Kit ( - JDK - ) installed on your system. In addition, the JRE or JDK you are using - to run - &PLATFORM_NAME; - - must be version 5 or higher - - - At this point in time, it is possible to run most - &PLATFORM_NAME; - - servers, such as the JAIN SLEE Server, using a Java 6 JRE or JDK. Be - aware, however, that presently the XML Document Management Server - does not run on Java 6. We suggest checking the &PLATFORM_NAME; - web site, forums or discussion pages if you need to inquire about - the status of running the XML Document Management Server with Java - 6. - - - . - - - Should I Install the JRE or JDK? - - Although you can run - &PLATFORM_NAME; - - servers using the Java Runtime Environment, we assume that most users - are developers interested in developing Java-based, - &PLATFORM_NAME; - - -driven solutions. Therefore, in this guide we take the tact of - showing how to install the full Java Development Kit. - - - - Should I Install the 32-Bit or the 64-Bit JDK, and Does It - Matter? - - Briefly stated: if you are running on a 64-Bit Linux or Windows - platform, you should consider installing and running the 64-bit JDK - over the 32-bit one. Here are some heuristics for determining whether - you would rather run the 64-bit Java Virtual Machine (JVM) over its - 32-bit cousin for your application: - - - - - - Wider datapath: the pipe between RAM and CPU is doubled, which - improves the performance of memory-bound applications when using a - 64-bit JVM. - - - - - 64-bit memory addressing gives virtually unlimited (1 exabyte) - heap allocation. However large heaps affect garbage collection. - - - - - Applications that run with more than 1.5 GB of RAM (including - free space for garbage collection optimization) should utilize the - 64-bit JVM. - - - - - Applications that run on a 32-bit JVM and do not require more - than minimal heap sizes will gain nothing from a 64-bit JVM. Barring - memory issues, 64-bit hardware with the same relative clock speed - and architecture is not likely to run Java applications faster than - their 32-bit cousin. - - - - - Note that the following instructions detail how to download and - install the 32-bit JDK, although the steps are nearly identical for - installing the 64-bit version. - - - - - Downloading - - You can download the Sun JDK 5.0 (Java 2 Development Kit) from - Sun's website: - - . Click on the - Download - link next to "JDK 5.0 Update - <x> - " (where - <x> - is the latest minor version release number). On the next page, select - your language and platform (both architecture—whether 32- or - 64-bit—and operating system), read and agree to the - Java Development Kit 5.0 License Agreement - , and proceed to the download page. - - - - The Sun website will present two download alternatives to you: one is - an RPM inside a self-extracting file (for example, - jdk-1_5_0_16-linux-i586-rpm.bin - ), and the other is merely a self-extracting file (e.g. - jdk-1_5_0_16-linux-i586.bin - ). If you are installing the JDK on Red Hat Enterprise Linux, Fedora, - or another RPM-based Linux system, we suggest that you download the - self-extracting file containing the RPM package, which will set up and - use the SysV service scripts in addition to installing the JDK. We - also suggest installing the self-extracting RPM file if you will be - running - &PLATFORM_NAME; - - in a production environment. - - - - Installing - - The following procedures detail how to install the Java - Development Kit on both Linux and Windows. - - - - Installing the JDK on Linux - - - Regardless of which file you downloaded, you can install it on - Linux by simply making sure the file is executable and then running - it: - - - ~]$ chmod +x "jdk-1_5_0_<minor_version>-linux-<architecture>-rpm.bin" -~]$ ./"jdk-1_5_0_<minor_version>-linux-<architecture>-rpm.bin" - - - - - You Installed Using the Non-RPM Installer, but Want the SysV - Service Scripts - - If you download the non-RPM self-extracting file (and installed it), - and you are running on an RPM-based system, you can still set up the - SysV service scripts by downloading and installing one of the - -compat - packages from the JPackage project. Remember to download the - -compat - package which corresponds correctly to the minor release number of - the JDK you installed. The compat packages are available from - - . - - - - - You do not need to install a - -compat - package in addition to the JDK if you installed the self-extracting - RPM file! The - -compat - package merely performs the same SysV service script set up that the - RPM version of the JDK installer does. - - - - Installing the JDK on Windows - - - Using Explorer, simply double-click the downloaded - self-extracting installer and follow the instructions to install the - JDK. - - - - - - - Configuring - - Configuring your system for the JDK consists in two tasks: setting - the - JAVA_HOME - environment variable, and ensuring that the system is using the - proper JDK (or JRE) using the - alternatives - command. Setting - JAVA_HOME - usually overrides the values for - java - , - javac - and - java_sdk_1.5.0 - in - alternatives - , but we will set them all just to be safe and consistent. - - - - - - Setting the - JAVA_HOME - Environment Variable on Generic Linux - - - - After installing the JDK, you must ensure that the - JAVA_HOME - environment variable exists and points to the location of your JDK - installation. - - - - Setting the - <envar>JAVA_HOME</envar> - Environment Variable on Linux - - - You can determine whether - JAVA_HOME - is set on your system by - echo - ing it on the command line: - - - - ~]$ echo $JAVA_HOME - - If - JAVA_HOME - is not set already, then you must set its value to the location of - the JDK installation on your system. You can do this by adding two - lines to your personal - ~/.bashrc - configuration file. Open - ~/.bashrc - (or create it if it doesn't exist) and add a line similar to - the following one anywhere inside the file: - - - export JAVA_HOME="/usr/lib/jvm/jdk1.5.0_<version>" - - - You should also set this environment variable for any other users - who will be running - &PLATFORM_NAME; - - (any environment variables - export - ed from - ~/.bashrc - files are local to that user). - - - - - - Setting - java - , - javac - and - java_sdk_1.5.0 - Using the - alternatives - command - - - - - Selecting the Correct System JVM on Linux using - <command>alternatives</command> - - - On systems with the - alternatives - command, including Red Hat Enterprise Linux and Fedora, you can - easily choose which JDK (or JRE) installation you wish to use, as - well as which - java - and - javac - executables should be run when called. - - - - As the root user - , call - /usr/sbin/alternatives - with the - - option to select between JDKs and JREs installed on your system: - - - root@localhost ~]$ /usr/sbin/alternatives --config java - -There are 3 programs which provide 'java'. - - Selection Command ------------------------------------------------ - 1 /usr/lib/jvm/jre-1.5.0-gcj/bin/java - 2 /usr/lib/jvm/jre-1.6.0-sun/bin/java -*+ 3 /usr/lib/jvm/jre-1.5.0-sun/bin/java - - -Enter to keep the current selection[+], or type selection number: - - - In our case, we want to use the Sun JDK, version 5, that we - downloaded and installed, to run the - java - executable. In the - alternatives - information printout above, a plus ( - + - ) next to a number indicates the one currently being used. As per - alternatives - ' instructions, pressing - Enter - will simply keep the current JVM, or you can enter the number - corresponding to the JVM you would prefer to use. - - - Repeat the procedure above for the - javac - command and the - java_sdk_1.5.0 - environment variable, - as the root user - : - - - ~]$ /usr/sbin/alternatives --config javac - - ~]$ /usr/sbin/alternatives --config java_sdk_1.5.0 - - - - - Setting the - JAVA_HOME - Environment Variable on Windows - - - - For information on how to set environment variables in Windows, - refer to - - . - - - - - - - Testing - - Finally, to make sure that you are using the correct JDK or Java - version (5 or higher), and that the java executable is in your - PATH - , run the - - java - - - command in the terminal from your home directory: - - - - ~]$ java -version -java version "1.5.0_16" -Java(TM) 2 Runtime Environment, Standard Edition (build 1.5.0_16-b03) -Java HotSpot(TM) Client VM (build 1.5.0_16-b03, mixed mode, sharing) - - - - Uninstalling - - There is usually no reason (other than space concerns) to remove a - particular JDK from your system, given that you can switch between - JDKs and JREs easily using - alternatives - , and/or by setting - JAVA_HOME - . - - - - Uninstalling the JDK on Linux - - On RPM-based systems, you can uninstall the JDK using the - - yum - - - command. - - - - Uninstalling the JDK on Windows - - On Windows systems, check the JDK entry in the - Start - menu for an uninstall command, or use - Add/Remove Programs - . - - -
- - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/Common_Content/Preface.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/Common_Content/Preface.xml deleted file mode 100644 index 7bbe6b2560..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/Common_Content/Preface.xml +++ /dev/null @@ -1,12 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - Preface - - - - - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/Common_Content/Setting_the_JBOSS_HOME_Environment_Variable.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/Common_Content/Setting_the_JBOSS_HOME_Environment_Variable.xml deleted file mode 100644 index 7b9ff427ad..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/Common_Content/Setting_the_JBOSS_HOME_Environment_Variable.xml +++ /dev/null @@ -1,353 +0,0 @@ - - -%BOOK_ENTITIES; - -]> -
- Setting the JBOSS_HOME Environment Variable - - The - &PLATFORM_NAME; - Platform - - ( - &PLATFORM_NAME; - - ) is built on top of the - JBoss Application Server - ( - JBoss AS - ). You do not need to set the - JBOSS_HOME - environment variable to run any of the - &PLATFORM_NAME; - Platform - - servers - unless - JBOSS_HOME - is - already - set. - - - The best way to know for sure whether - JBOSS_HOME - was set previously or not is to perform a simple check which may save - you time and frustration. - - - Checking to See If JBOSS_HOME is Set on Unix - - At the command line, - echo - $JBOSS_HOME - to see if it is currently defined in your environment: - - - - ~]$ echo $JBOSS_HOME - - The - &PLATFORM_NAME; - Platform - - and most &PLATFORM_NAME; - servers are built on top of the - JBoss Application Server - ( - JBoss AS - ). When the - &PLATFORM_NAME; - Platform - - or &PLATFORM_NAME; - servers are built - from source - , then - JBOSS_HOME - must - be set, because the &PLATFORM_NAME; - files are installed into (or - over top of - if you prefer) a clean - JBoss AS - installation, and the build process assumes that the location pointed - to by the - JBOSS_HOME - environment variable at the time of building is the - JBoss AS - installation into which you want it to install the &PLATFORM_NAME; - files. - - - This guide does not detail building the - &PLATFORM_NAME; - Platform - - or any &PLATFORM_NAME; - servers from source. It is nevertheless useful to understand the role - played by - JBoss AS - and - JBOSS_HOME - in the &PLATFORM_NAME; - ecosystem. - - - The immediately-following section considers whether you need to set - JBOSS_HOME - at all and, if so, when. The subsequent sections detail how to set - JBOSS_HOME - on Unix and Windows - - - - Even if you fall into the category below of - not needing - to set - JBOSS_HOME - , you may want to for various reasons anyway. Also, even if you are - instructed that you do - not need - to set - JBOSS_HOME - , it is good practice nonetheless to check and make sure that - JBOSS_HOME - actually - isn't - set or defined on your system for some reason. This can save you both - time and frustration. - - - - You - DO NOT NEED - to set - JBOSS_HOME - if... - - - - - ...you have installed the - &PLATFORM_NAME; - Platform - - binary distribution. - - - - - ...you have installed a &PLATFORM_NAME; - server binary distribution - - which bundles - JBoss AS - . - - - - - - You - MUST - set - JBOSS_HOME - if... - - - - - ...you are installing the - &PLATFORM_NAME; - Platform - - or any of the &PLATFORM_NAME; - servers - from source - . - - - - - ...you are installing the - &PLATFORM_NAME; - Platform - - binary distribution, or one of the &PLATFORM_NAME; - server binary distributions, which - do not - bundle - JBoss AS - . - - - - - Naturally, if you installed the - &PLATFORM_NAME; - Platform - - or one of the &PLATFORM_NAME; - server binary releases which - do not - bundle - JBoss AS - , yet requires it to run, then you should - - install - JBoss AS - - before setting - JBOSS_HOME - or proceeding with anything else. - - - Setting the JBOSS_HOME Environment Variable on Unix - - The - JBOSS_HOME - environment variable must point to the directory which contains all - of the files for the - - &PLATFORM_NAME; - Platform - - or individual &PLATFORM_NAME; - server - - that you installed. As another hint, this topmost directory contains - a - bin - subdirectory. - - - - Setting - JBOSS_HOME - in your personal - ~/.bashrc - startup script carries the advantage of retaining effect over reboots. - Each time you log in, the environment variable is sure to be set for - you, as a user. On Unix, it is possible to set - JBOSS_HOME - as a system-wide environment variable, by defining it in - /etc/bashrc - , but this method is neither recommended nor detailed in these - instructions. - - - To Set JBOSS_HOME on Unix... - - - Open the - ~/.bashrc - startup script, which is a hidden file in your home directory, in a - text editor, and insert the following line on its own line while - substituting for the actual install location on your system: - - - export JBOSS_HOME="/home/<username>/<path>/<to>/<install_directory>" - - - - Save and close the - .bashrc - startup script. - - - - - You should - source - the - .bashrc - script to force your change to take effect, so that - JBOSS_HOME - becomes set for the current session - - - Note that any other terminals which were opened prior to your - having altered - .bashrc - will need to - source - ~/.bashrc - as well should they require access to - JBOSS_HOME - . - - - . - - - ~]$ source ~/.bashrc - - - - Finally, ensure that - JBOSS_HOME - is set in the current session, and actually points to the correct - location: - - - - The command line usage below is based upon a binary installation of - the - &PLATFORM_NAME; - Platform - - . In this sample output, - JBOSS_HOME - has been set correctly to the - topmost_directory - of the - &PLATFORM_NAME; - - installation. Note that if you are installing one of the standalone - &PLATFORM_NAME; - - servers (with - JBoss AS - bundled!), then - JBOSS_HOME - would point to the - topmost_directory - of your server installation. - - - - ~]$ echo $JBOSS_HOME -/home/silas/&HOME_LIN; - - - - - Setting the JBOSS_HOME Environment Variable on Windows - - The - JBOSS_HOME - environment variable must point to the directory which contains all - of the files for the - &PLATFORM_NAME; - Platform or individual &PLATFORM_NAME; - server - - that you installed. As another hint, this topmost directory contains - a - bin - subdirectory. - - - - For information on how to set environment variables in recent versions - of Windows, refer to - - . - -
- - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/Preface.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/Preface.xml deleted file mode 100644 index 88c4ed5563..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/Preface.xml +++ /dev/null @@ -1,12 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - Preface - - - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/RestComm_User_Guide.ent b/restcomm/restcomm.docs/sources/src/main/resources/en-US/RestComm_User_Guide.ent deleted file mode 100644 index f2d9d7350f..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/RestComm_User_Guide.ent +++ /dev/null @@ -1,19 +0,0 @@ - - - - - - - - - - - - - - - - - - - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/RestComm_User_Guide.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/RestComm_User_Guide.xml deleted file mode 100644 index 9c5645fb24..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/RestComm_User_Guide.xml +++ /dev/null @@ -1,28 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - - - - - - - - - - - - - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/Revision_History.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/Revision_History.xml deleted file mode 100644 index 8df7b8d7e4..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/Revision_History.xml +++ /dev/null @@ -1,74 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - Revision History - - - - - 2.1 - Tuesday 27 2014 - - Charles - Roufay - charles.roufay@telestax.com - - - - Update doc with RestComm USSD Integration and new RVD features - - - - - - 2.0 - Friday Sep 2013 - - Charles - Roufay - charles.roufay@telestax.com - - - - Update doc with RestComm JBoss AS7 and Advanced Examples - - - - - 1.0 - Mon Apr 18 2011 - - George - Vagenas - gvagenas@gmail.com - - - - First release CDI-Telco-Framework user guide - - - - - - 1.0 - Mon Apr 18 2011 - - George - Vagenas - gvagenas@gmail.com - - - - Changes to reflect changes made regarding JBoss AS5 and - the refactoring made to the project. - - - - - - - - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Admin-UI.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Admin-UI.xml deleted file mode 100644 index 8985ddddff..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Admin-UI.xml +++ /dev/null @@ -1,462 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - Restcomm Admin User Interface - -The admin user interface is a great way to perform repetive Restcomm task in a orderly manner. Care has been taken to make the user experience intuitive for those new to the platform. In this chapter, you will learn how to use some of the features available to create Clients, Phone Numbers, Check Call Logs, Get Speech-to-Text Transcription and many more - - -Please also visit the official Telestax documentation page for more usage tutorials:http://www.telestax.com/docs - - -
-Login Interface - -You need to make sure Restcomm is running before you can use the Admin User Interface. When you are running on a local install, open your web browser and go to this url - http://127.0.0.1:8080/restcomm-management . You will see a screenshot similar to the one below - - - - -Please see the for more information about changing the default password. - - - - -
- -
-Passwords, Sub Accounts Settings - -When you click on the Administrators Account at the top right corner of the window, as shown in the screenshot below, you will be able to change your Administrators passwords and create Sub Accounts. - - - - - - - - - - Account Settings - - - - - - Number - Description - - - - - 1 - Shows the current Acount profile - - - 2 - You can create new Sub Accounts - - - 3 - Descriptive name of the user account - - - 4 - New password of the current logged in account - - - 5 - You can activate or suspend an account - - - 6 - There are three options, Trial, Basic and Full(default) - - - 7 - List of Sub Accounts - - - -
- - -
- - - -
-Dashboard - -This is the default page where you can get an overview of your Restcomm Installation. - - - - - - - - - Dashboard - - - - - - Number - Description - - - - - 1 - Account SID is the default used by the Administrator's account - - - 2 - The Auth Token is the password that is required for any Restcomm operation. You can click on the hidden button to reveal the hashed password. - - - 3 - The Debugger lets you troubleshoot any issues you might encounter using Restcomm - - - 4 - A quick way to get all the API exposed by Restcomm - - - 5 - Additional information about Calls parsed by Restcomm - - - 6 - Additional information about SMS parsed by Restcomm - - - - 7 - This will take you to the Restcomm Visual Designer (RVD) page located a - - - - -
- - -
- -
-Restcomm Numbers - -This will show the default demo applications that come with Restcomm. When you start creating applications and attaching numbers, they will also be displayed on this page. - - - - - - - - - Restcomm Numbers - - - - - - Number - Description - - - - - 1 - The number 1235 is attached to the hello-world.xml application. You must have configured VoiceRSS text-to-speech to use this application - - - 2 - The number 1236 is attached to a the Gather verb. It will ask you to enter a number and you can hear the number you enter. You must have configured VoiceRSS text-to-speech to use this application - - - 3 - The number 1234 plays a pre-recorded file. - - - 4 - This icon lets you edit the Name of the entry to a more descriptive one. - - - 5 - This button lets you create a new number that can be attached to a RCML - - - - - -
- - - -
-Register Number - - -When you click on the Register Number button, you will see a screenshot similar to the one below. This will allow you to create a new phone number that can be attached to a Restcomm application. - - - - - - - - - Restcomm Numbers - - - - - - Number - Description - - - - - 1 - The field to enter the phone number. - - - 2 - This button will show the Area Code. You must have Voip Innovations API account configured in the restcomm.xml file to use this feature. It will search for DID in the area code specified. At the moment, only US DIDs are available. - - - - 3 - This button will show advanced options if you want to add more features to the phone number like the VoiceUrl - - - 4 - See the Rest API for more information - - - 5 - This button will show available Restcomm Visual Designer (RVD) applications that can be linked to a phone number. - - - 6 - See the Rest API for more information - - - - - - -
-
-
-Edit Update Number - - -Editing a phone number can be done by clicking on the number, the screenshot below shows how you can edit the number 1235. You can change the VoiceUrl to which the number is attached. - - - - - - - - - Edit Update Numbers - - - - - - Number - Description - - - - - 1 - You can link the phone number to a VoiceUrl application. See the REST API for more details. - - - 2 - You can link the phone number to a SMS application. See the REST API for more details - - - 3 - You can link the phone number to a USSD application. See the REST API for more details - - - 4 - Caller Id lookup requires a CNAM provier - - - 5 - You can save your changes or press close to discard the changes - - - - - -
- -
-
- - -
-RestComm Clients - -RestComm Client is a feature that allows you to create a Restcomm profile that can be used for P2P or B2BUA calls. This will be empty until you create a new client. You can create a new client by clicking on the Resgister RestComm Client button. - - - - - - - - - RestComm Clients - - - - - - Number - Description - - - - - 1 - The client name. (ex. alice or bob) - - - 2 - The password that will be used to when you want to register the client with restcomm - - - 3 - Use to open optional parameters windows - - - 4 - This can be the full name of the client (ex.Alice Wilkinson) or any descriptive name - - - 5 - This is where you specify the VoiceUrl that is automatically invoked when the client is called. See the REST API for more details. - - - 6 - See the REST API for more details. - - - 7 - This will validate your changes and create the client. - - - - -
- - - -
-
-Outgoing CallerId - -Will be available in future release - -
- -
-Logs - -The log section gives you an overview of current Restcomm system information. - - - -
-Logs - Calls - - - -A list of all calls that have been processed by Restcomm - - - - - - - -
- -
-Logs - Messages - - - -A list of all SMS messages that have been processed by Restcomm - - - - - - - - -
- -
-Logs - Recordings - - -A list of all Recordings (using the Record Verb) that have been processed by Restcomm - - - - - - - -
- -
-Logs - Transcriptions - - -A list of all Transcriptions that have (using the Transcribe parameters of the Record Verb) that have been processed by Restcomm. See the - - - - - - - -
- -
-Logs - Notifications - - -A list of all Notifications received by Restcomm. - - - - - - - -
- -
- - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Advanced_Examples.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Advanced_Examples.xml deleted file mode 100644 index b8ea2abc32..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Advanced_Examples.xml +++ /dev/null @@ -1,317 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - -Advanced RestComm Examples - - -In this chapter, you will learn how to use more advanced features of Restcomm verbs to create communications applications - - -
-Activating Text-to-Speech (TTS) - - -You must get an API key from http://www.voicerss.org in order to proceed with this section. You can register for a free account and an API key will be emailed to you. Once you have the API key, open the $RESTCOMM_HOME/standalone/deployments/restcomm.war/WEB-INF/conf/restcomm.xml file and find the speech-synthesizer VoiceRSS section. Add your API key as shown below and restart RestComm - - - - - http://api.voicerss.org - 2901c0aXXXXXXXXXXXXXX -]]> - - - - -Once you have done that, you are ready to use Text-to-Speech feature of RestComm. - - -
-Say Verb -Create a file called test.xml in the $RESTCOMM_HOME/standalone/deployments/restcomm.war/demos/ directory and copy the content of the application below into the test.xml file and save it - - - - - - Welcome to RestComm, you have successfully tested the Say Verb - -]]> - - - -From a command terminal run the Curl command below to bind the 5555 phone number to the test.xml file. - - - - - -If the command is successful, you will see an output similar to the following: - - - - -Next, you have to configure your SIP phone and make a call to the test.xml dialing number 5555 - - -Known Issues with some SIP Phones - -Some SIP phones have been known to have codec problems and do not correctly render the application as desired. In this example, we shall be using Ekiga. - - - - -Start the SIP phone Ekiga (see below) and dial 5555@127.0.0.1:5080. You will hear the content of the Say Verb in the test.xml file. - - - - - - - - -
- -
-Record Verb -Create a file called test.xml in the $RESTCOMM_HOME/standalone/deployments/restcomm.war/demos/ directory and copy the content of the application below into the test.xml file and save it. Dial the number 5555 and when you hear the beep sound, leave a message and hangup when you are finished. - - - - - - Welcome to RestComm, leave a message after the beep - - -]]> - - - - -A wave file will be recorded and saved in the $RESTCOMM_HOME/standalone/deployments/restcomm.war/recordings/ directory. You can use any media player to listen to the recorded voice message. - -
- -
-Dial Verb and Client - -In order to use the Dial verb, you will need two users and register them to RestComm. - -Restcomm has already two users created for you for testing purposes, alice and bob - - -Alice Restcomm client -Username = alice / Password = 1234 - - - -Bob Restcomm client -Username = bob / Password = 1234 - - - - - -Register Alice and Bob using softphone Sflphone - - - - -In order to start two instances of Sflphone on the same computer, you need to start the second instance using sudo Sflphone. - - - -When Sflphone is started, go to the the menu Edit->Accounts, then press the Add button - - - -Fill out the configuration for Bob as shown in the screenshot below: - - -Sflphone Basic Configuration - - - - - - - - - -Sflphone Advanced Configuration -Make sure the port number is set to 5061 - - - - - - - - - -In the second instance of Sflphone, register user Alice following the same procedure used for Bob. In the Advanced settings, make sure the port number for user Alice is set to 5060 - - - -Copy the content of the application below into the test.xml file and save. - - - - - - Welcome to RestComm, you are being forwarded to Alice - - alice - - -]]> - -From the Sflphone registered with user Bob, make a call to Alice. Enter the name alice and make the call - -
- -
-Dial Verb and Uri - -Copy the Dial Uri application below to the test.xml. Remember that the port number for user Alice is 5060 and that is the port that is used in the application. If you want to use a different port, you also have to change the SIP Uri to reflect that. You can then make a call from user Bob to Alice. - - - - - Welcome to RestCom, you are being forward to Alice - - sip:alice@127.0.0.1:5060 - - - -]]> - -
- -
-Dial Verb and Conference - -Copy the Dial Conference application below to the test.xml and save the file. You can dial any number like 4321 and RestComm will read the Say verb and make a beep sound when you join the conference room. You can use another SIP phone to join the same conference. It works better if you are on a different computer as this reduces the echo generated from the microphones. - - - - - Welcome to RestCom, you are now joining the conference room - - test-conference - - -]]> - -
- -
-Gather Verb - -This verb is used to get user input and instruct RestComm to perform a specific task. This example is a little bit more elaborate and it will require the creation of a php file. You also must host the php file on a web server like Apache. Copy the Gather application below into the test.xml and save the file. - - - - - - Welcome to Telestax RestCom. - For opening hours, press 1. - to talk to Alice, press 2. - - - Sorry, I didn't get your response. - http://127.0.0.1:8080/restcomm/demos/test.xml - -]]> - - - -Create a php file in the Apache /var/www/html directory - -You can use any web server of your choice, in this example, we shall be using Apache on a Linux computer -Start the httpd server as follows sudo service httpd start -Create a new file called test-user-input.php in the /var/www/html directory -Copy the php application below into the test-user-input.php and save - - - -'; - - echo ''; - - $user_input = (int) ($_POST['Digits']); - - if ($user_input == 1) - - { - echo 'Our Opening hours are 24 hours 7 Days a week '; - echo 'Telestax appreciates your business'; - echo 'You may Hang up or wait to be redirected to the main menu'; - echo 'http://127.0.0.1:8080/restcomm/demos/test.xml'; - - } elseif($user_input == 2) { - - echo 'You are being forwarded to Alice'; - echo ''; - echo 'alice'; - echo ''; - - } - echo ''; -?> -]]> - - -This example welcomes the user and offers two options. If the user presses 1, he hears the openining hours message. If the user presses 2, he will be redirected to user Alice. - -To test the application using the Gather verb, dial any number from user Bob and follow the application instruction. - - -Using the Fax, SMS and other features of RestComm - -Please note that Telestax also offers TelScale RestComm hosted on Amazon AWS. If you want to learn more about how to use DID provisioning and more, please visit www.telestax.com - - -
- -
- - - - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Advantages.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Advantages.xml deleted file mode 100644 index bc6f75951d..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Advantages.xml +++ /dev/null @@ -1,21 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - Advantages of the RestComm - Loose coupling and reusability of SIP Components across different - applications are the most obvious advantages over the current SIP - Servlets 1.1 specification. In addition to that, you get better - integration with Java EE 6. - - In the future we will try to explore further usage of this - component-based approach to allow application developers to compose - their applications in a new way and benefits from Modularization - within a single application which should give additional performance - improvements over SIP Servlets Application Routing and Chaining while - you can still use the latter. - - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Conclusion.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Conclusion.xml deleted file mode 100644 index a6c6c703c3..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Conclusion.xml +++ /dev/null @@ -1,18 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - Conclusion - CTF provides an extensive set of features and advantages to the - developer but we believe there are improvements to be done. Developers - hold a vital role for the further improvement of the framework. - Please give it a try and provide early feedback ! - This is a perfect time to offer any ideas for or contribute Telco - modules/extensions that you would find useful in your applications. - - Leave us your comments on the discussion started at mobicents - google group - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Configuration.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Configuration.xml deleted file mode 100644 index 210a8e328b..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Configuration.xml +++ /dev/null @@ -1,1585 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - -Getting Started and General Configuration - -
-Quick Start Guide - -RestComm is a robust, powerful platform that will facilitate building comprehensive real-time communication solutions. The steps below will help you get started with ease. - - - - -On a production system, please take into consideration the following: - - -MMS can get into a bad state and report "Too many open files" error because of Linux default value on the total number of file descriptors. - - -You can go to the following sites to see how to increase the maximum number of open files in your server: - - - - - - - - - - - - - -Running RestComm on JBoss - -JBoss Security - -Running a secure application is dependent on multiple factors. Restcomm runs on JBoss which implies that system security implemention can be handled at JBoss level. Please see the link below on how you can make your server environment more secure. - - - - - -Download TelScale-Restcomm-JBoss-AS7-XX.XX.GA.zip - - - - -Using a terminal of your choice, extract the content of TelScale-Restcomm-XX.XX.GA.zip to a local directory on your computer. The root directoy into which you extract the content of the .zip file will be referred to as $RESTCOMM_HOME. - - - - -Go to $RESTCOMM_HOME/telscale-media/telscale-media-server/bin - - - - -change the permission of the run.sh as follows - - - - -sudo chmod +x ./run.sh - - - - -then start the TelScale Media Server as follows - - - - -sudo ./run.sh - - - - -If all is correctly started you will see the following at the end of the bash terminal window - -[MainDeployer] [[[[[[[[[ Mobicents Media Server: release.version=3.0.0.FINAL Started ]]]]]]]]] - - - - - - -Open another terminal and proceed as follows: - - - - - -Go to $RESTCOMM_HOME/bin - - - - -change the permissions of all the .sh files in the bin directory as follows - - - - -sudo chmod +x ./*.sh - - - - -Start RestComm by running the following command - - - - -sudo ./standalone.sh -c standalone-sip.xml - - - - - -If RestComm is correctly started you will see the following at the end of the terminal - - - -INFO [Version] TelScale Sip Servlets 6.1.3.GA-TelScale (build: Git Hash=r8947f2732ee64c76566ed6c0b236204c048538e1 date=201306131639) Started. -17:30:05,854 INFO [Version] -============================================================================== -== == -== Thank you for running TelScale == -== Carrier Grade Communications Platform by the creators of Mobicents == -== Copyright 2011-2013 Telestax, Inc. == -== http://www.telestax.com/ == -== == -============================================================================== - - - - - - - - - -Running RestComm on Tomcat - - - -Download TelScale-Restcomm-Tomcat-XX.XX.GA.zip - - - - -Using a terminal of your choice, extract the content of TelScale-Restcomm-Tomcat-XX.XX.GA.zip to a local directory on your computer. The root directoy into which you extract the content of the .zip file will be referred to as $RESTCOMM_HOME. - - - - -Go to $RESTCOMM_HOME/telscale-media/telscale-media-server/bin - - - - -change the permission of the run.sh as follows - - - - -sudo chmod +x ./run.sh - - - - -then start the TelScale Media Server as follows - - - - -sudo ./run.sh - - - - -If all is correctly started you will see the following at the end of the bash terminal window - -[MainDeployer] [[[[[[[[[ Mobicents Media Server: release.version=3.0.0.FINAL Started ]]]]]]]]] - - - - - - -Open another terminal and proceed as follows: - - - - - -Go to $RESTCOMM_HOME/bin - - - - -change the permissions of all the .sh files in the bin directory as follows - - - - -sudo chmod +x ./*.sh - - - -The content of the bin directory should be similar to the list below - -bootstrap.jar digest.sh tomcat-juli.jar -catalina.bat setclasspath.bat tomcat-native.tar.gz -catalina.sh setclasspath.sh tool-wrapper.bat -catalina-tasks.xml shutdown.bat tool-wrapper.sh -commons-daemon.jar shutdown.sh version.bat -commons-daemon-native.tar.gz startup.bat version.sh -cpappend.bat startup.sh -digest.bat telestax-license.xml - - - -Start RestComm by running the following command - - - - - sudo ./catalina.sh run - - - -If RestComm is correctly started on Tomcat, you will see an output similar to the one below - -Started. -2013-08-15 19:02:46,903 INFO [Version] (main) -============================================================================== -== == -== Thank you for running TelScale == -== Carrier Grade Communications Platform by the creators of Mobicents == -== Copyright 2011-2013 Telestax, Inc. == -== http://www.telestax.com/ == -== == -============================================================================== -Aug 15, 2013 7:02:50 PM org.apache.catalina.startup.Catalina start -INFO: Server startup in 14090 ms - - - - - - - - - - - - - - -
- -
-Login into Restcomm and Changing the Default Password - -Before you start using Restcomm, you must change the default password. Please follow the steps below - - - - -Open your browser and go to http://127.0.0.1:8080/restcomm-management - -Enter the following information: - - Email/Account SID = administrator@company.com - - Password = RestComm - - Press the Login button - - - - -You will see a screen similar to the one below - - - - - - - - - - - -In the screen below, change the password and update. - - - - - - - - - - -Once the password has been changed, you can now log into the Admin Management interface and start using Restcomm. - - -
- - -
-Testing the Demo Applications - - - -Restcomm comes prepackaged with multiple example applications designed to help you quickly get started. For more demos and configuration details please visit www.telestax.com/doc - - - - - -
-Demo 1 - Testing the Play Verb - - -Start a SIP phone (see below) and dial 1234@127.0.0.1:5080. You will hear a welcome message. - - - - - - - - -Some SIP phones have codec incompatibility issues, in the above example, we used Ekiga. You may also try Jitsi or Sflphone. - - - - - Getting Started and General Configuration - -
- Quick Start Guide - - RestComm is a robust, powerful platform that will facilitate building - comprehensive real-time communication solutions. The steps below will - help you get started with ease. - - - - - On a production system, please take into consideration the following: - - - MMS can get into a bad state and report "Too many open files" error - because of Linux default value on the total number of file - descriptors. - - - You can go to the following sites to see how to increase the maximum - number of open files in your server: - - - - - - - - - - - - - - Running RestComm on JBoss - - JBoss Security - - Running a secure application is dependent on multiple factors. - Restcomm runs on JBoss which implies that system security - implemention can be handled at JBoss level. Please see the link - below on how you can make your server environment more secure. - - - - - - - Download - TelScale-Restcomm-JBoss-AS7-XX.XX.GA.zip - - - - - Using a terminal of your choice, extract the content of - TelScale-Restcomm-XX.XX.GA.zip - to a local directory on your computer. The root directoy into which - you extract the content of the .zip file will be referred to as - $RESTCOMM_HOME. - - - - - Go to - $RESTCOMM_HOME/telscale-media/telscale-media-server/bin - - - - - change the permission of the - run.sh - as follows - - - - - sudo chmod +x ./run.sh - - - - - then start the TelScale Media Server as follows - - - - - sudo ./run.sh - - - - - If all is correctly started you will see the following at the end - of the bash terminal window - - [MainDeployer] [[[[[[[[[ Mobicents Media Server: release.version=3.0.0.FINAL - Started ]]]]]]]]] - - - - - - - Open another terminal and proceed as follows: - - - - - - Go to - $RESTCOMM_HOME/bin - - - - - change the permissions of all the - .sh - files in the bin directory as follows - - - - - sudo chmod +x ./*.sh - - - - - Start RestComm by running the following command - - - - - sudo ./standalone.sh -c standalone-sip.xml - - - - - - If RestComm is correctly started you will see the following at the - end of the terminal - - - - INFO [Version] TelScale Sip Servlets 6.1.3.GA-TelScale (build: Git - Hash=r8947f2732ee64c76566ed6c0b236204c048538e1 date=201306131639) - Started. - 17:30:05,854 INFO [Version] - ============================================================================== - == == - == Thank you for running TelScale == - == Carrier Grade Communications Platform by the creators of Mobicents - == - == Copyright 2011-2013 Telestax, Inc. == - == http://www.telestax.com/ == - == == - ============================================================================== - - - - - - - - - - Running RestComm on Tomcat - - - - Download - TelScale-Restcomm-Tomcat-XX.XX.GA.zip - - - - - Using a terminal of your choice, extract the content of - TelScale-Restcomm-Tomcat-XX.XX.GA.zip - to a local directory on your computer. The root directoy into which - you extract the content of the .zip file will be referred to as - $RESTCOMM_HOME. - - - - - Go to - $RESTCOMM_HOME/telscale-media/telscale-media-server/bin - - - - - change the permission of the - run.sh - as follows - - - - - sudo chmod +x ./run.sh - - - - - then start the TelScale Media Server as follows - - - - - sudo ./run.sh - - - - - If all is correctly started you will see the following at the end - of the bash terminal window - - [MainDeployer] [[[[[[[[[ Mobicents Media Server: release.version=3.0.0.FINAL - Started ]]]]]]]]] - - - - - - - Open another terminal and proceed as follows: - - - - - - Go to - $RESTCOMM_HOME/bin - - - - - change the permissions of all the - .sh - files in the bin directory as follows - - - - - sudo chmod +x ./*.sh - - - - The content of the bin directory should be similar to the list - below - - bootstrap.jar digest.sh tomcat-juli.jar - catalina.bat setclasspath.bat tomcat-native.tar.gz - catalina.sh setclasspath.sh tool-wrapper.bat - catalina-tasks.xml shutdown.bat tool-wrapper.sh - commons-daemon.jar shutdown.sh version.bat - commons-daemon-native.tar.gz startup.bat version.sh - cpappend.bat startup.sh - digest.bat telestax-license.xml - - - - Start RestComm by running the following command - - - - - sudo ./catalina.sh run - - - - If RestComm is correctly started on Tomcat, you will see an - output similar to the one below - - Started. - 2013-08-15 19:02:46,903 INFO [Version] (main) - ============================================================================== - == == - == Thank you for running TelScale == - == Carrier Grade Communications Platform by the creators of Mobicents - == - == Copyright 2011-2013 Telestax, Inc. == - == http://www.telestax.com/ == - == == - ============================================================================== - Aug 15, 2013 7:02:50 PM org.apache.catalina.startup.Catalina start - INFO: Server startup in 14090 ms - - - - - - - - - - - - - - -
-
- Testing the Demo Applications - - - - Restcomm comes prepackaged with multiple example applications designed to - help you quickly get started. - - -
- Demo 1 - Testing the Play Verb - - - Start a SIP phone (see below) and dial - 1234@127.0.0.1:5080 - . You will hear a welcome message. - - - - - - - - - Some SIP phones have codec incompatibility issues, in the above example, - we used Ekiga. You may also try Jitsi or Sflphone. - - - - -$RESTCOMM_HOME/standalone/deployments/restcomm.war/demos/hello-play.xml. -]]> - - - - -
-
- Demo 2 - Testing Say Verb - - - You must first activate Text-to-Speech before you can proceed. - - - You must get an API key from - http://www.voicerss.org - in order to proceed with this section. You can register for a free - account and an API key will be emailed to you. Once you have the API - key, open the - $RESTCOMM_HOME/standalone/deployments/restcomm.war/WEB-INF/conf/restcomm.xml - file and find the speech-synthesizer VoiceRSS section. Add your API - key as shown below and restart RestComm - - - - - http://api.voicerss.org - 2901c0aXXXXXXXXXXXXXX -]]> - - - - Start a SIP phone dial - 1235@127.0.0.1:5080 - . You will hear a welcome message in multiple languages. - - - - - - - - - -
- -
- Demo 3 - Testing Gather Verb - This demo creates a simple IVR system - - - - Activate DTMF using Ekiga - Make sure your DTMF setting in Ekiga is set to RFC2833. In - order to set it correctly, go to the menu - Edit->Preference->Protocols->SIP Settings->DTMF Mode - You may also use SFLPHONE instead of Ekiga - - - - - Start a SIP phone dial - 1236@127.0.0.1:5080 - . You will hear a message asking you to enter a digit and press - star. If the digit is correctly received, Restcomm will replay the - number you entered. - - - - -$RESTCOMM_HOME/standalone/deployments/restcomm.war/demos/gather/hello-gather.xml. -and -$RESTCOMM_HOME/standalone/deployments/restcomm.war/demos/gather/gather.jsp. -]]> - - -
- -
- Demo 4 - Testing the Dial Sip Noun - - This demo makes a call from one SIP phone to another. The Demo uses - the SIP noun. You can calll any SIP account. All you have to do is - change the content of the - dial-sip.xml - - - In order to use this demo, you may use the default accounts, Alice and - Bob, and register them on two separate SIP phones. Start both SIP - phones and make sure Alice and Bob are registerd with the password - (1234). These users come pre-configured with Restcomm for test - purposes. - - - - Start Two SIP Phone Sessions - If you are using the SIP sflphone here is what to do: - Start first instance ex. - sflphone - Start second instance ex. - sudo sflphone - - - - In the application - dial-sip.xml - you can change the default to - sip:alice@127.0.0.1:5061? - - - This will allow you to make a call to Alice. Note that Alice must be - registered on port 5061 for the call to succeed. - - - - From the the phone on which Bob is registerd, dial the number - 1237 - . The phone on which Alice is registered will ring and the - connection will be made when you answer the call. - - - - - - - - - -
- -
- Demo 5 - Testing the Client Noun - This demo makes a call from one SIP Client to Another. The demo - uses the Client noun - - - In order to use this demo, you must have user Alice and Bob registered - on two separate SIP phones. Start both SIP phones and make sure - Alice and Bob are registerd with the password (1234). These users - come pre-configured with Restcomm for test purposes. - - - - Start Two SIP Phone Sessions - - You can start the second instance of your SIP phones by prepending - the executable with - sudo - . If you are using the SIP sflphone here is what to do: - - Start first instance ex. - sflphone - Start second instance ex. - sudo sflphone - - - - - From the phone on which Bob is registerd, dial the number - 1238 - . The phone on which Alice is registered will ring and the - connection will be made when you answer the call. - - - - -$RESTCOMM_HOME/standalone/deployments/restcomm.war/demos/dial/client/dial-client.xml. - -]]> - - - - - Advanced Examples - - Please go to this chapter for more examples - - . - - - -
- - -
- Demo 6 - Testing Conference Noun - - This demo Lets a user join a conference as a moderator and the other - user as a participant. The participant will dial - 1310 - and will hear a hold music. The moderator will dial - 1311 - and the hold music will stop and the conference will be started. - - - - Most SIP phones will require you to register before you can make a call. - You can use the default accounts, Alice and Bob with password - (1234)to register. - - - - Start Two SIP Phone Sessions - - You can start the second instance of your SIP phones by prepending - the executable with - sudo - . If you are using the SIP sflphone here is what to do: - - Start first instance ex. - sflphone - Start second instance ex. - sudo sflphone - - - - - From the phone on which Bob is registerd, dial the number - 1310 - . From the phone on which Alice is registered, dial - 1311 - - - - - - - - - - Advanced Examples - - Please go to this chapter for more examples - - . - - - -
- - -
-Demo 7 - RVD Say Verb - -This example is built using the Restcomm Visual Designer. You can register a SIP phone with the user bob or alice and password (1234). Then dial 1239. You will hear a welcome message. - - - - - - -
- -
-Demo 8 - RVD Gather Verb - -This demo uses the Gather verb to get user input and map it to the corresponding modules. You can register a SIP phone with the user bob or alice and password (1234). Then, dial 1240. You will hear a welcome message asking you to press 1 for Sales or 2 for Technical Support. - - - - - - -
- - -
-Demo 9 - RVD External Services - -This demo uses the External Services to Restcomm to an external program for more complex solutions. There are two parts to this demo application, the RVD app and the external servlet http://127.0.0.1:8080/RvdExternalServicesDemo/rvdESdemo which manages the logic. You can register a SIP phone with the user bob or alice and password (1234). Then, dial 1241. You will be asked to enter a customer ID. You can enter number 11 or 12. You will hear a corresponding welcome message depending on the customer ID you entered. - - - - - - -
- - -
- -
- - - - - -
- - Detailed Configuration - - - In order for TelScale RestComm to function properly it must first be - configured. In this chapter we will cover - the settings for TelScale RestComm and it's respective plug-ins. In - order to make TelScale RestComm easy to manage - all of the configuration is done in only one file. The file is located - at - RESTCOMM_HOME/WEB-INF/webapps/restcomm/conf/restcomm.xml - and is composed of the sections that follow. - - - TelScale RestComm provides a default set of plug-ins for the - storage engine, fax service, SMS aggregator, - automatic speech recognition, and speech synthesis. All these services are - pluggable and TelScale RestComm is not - limited to any service provider or software platformz. - -
-
- Runtime Settings - - The runtime-settings are used by TelScale RestComm at runtime to - customize it's behavior. A list of the runtime settings - and a description is provided below. - - - - Runtime Settings - - - - - - Element - Description - - - - - api-version - The version of the TelScale RestComm API that we will be - executing by default. - - - prompts-uri - The location where the audio prompts are located. - - - conference-entry-audio - This audio will be played when a participant joins the conference - - - conference-exit-audio - This audio will be played when a participant leaves the conference - - - cache-path - The local path the cache folder. - - - cache-uri - The HTTP URI to the cache folder. - - - recordings-path - The local path to the folder containing the recordings. - - - recordings-uri - The HTTP URI to the folder containing the recordings. - - - error-dictionary-uri - The HTTP URI to the TelScale RestComm error dictionary. - - - external-ip - - The IP to use for out-bound SIP REGISTER requests. This is useful - when you - want to report a different IP than the one TelScale RestComm picked - by default. - - - - use-to - - If set to true TelScale RestComm will use the To header to - determine the destination. - If set to false TelScale RestComm will use the Request URI to - determine the destination. - - - - outbound-proxy-user - - The user name used to authenticate with the outbound proxy. - (Optional) - - - - outbound-proxy-password - - The password used to authenticated with the outbound proxy. - (Optional) - - - - outbound-proxy-uri - - The SIP URI to the outbound proxy. Note: Do not prepend 'sip:' to - the proxy uri. - If necessary the port can be included, for example - alice@localhost:5080 - - - - fallback-outbound-proxy-user - - The user name used to authenticate with the fallback outbound proxy. - (Optional) - - - - fallback-outbound-proxy-password - - The password used to authenticated with the fallback outbound proxy. - (Optional) - - - - fallback-outbound-proxy-uri - - The SIP URI to the fallback outbound proxy. Note: Do not prepend 'sip:' to - the proxy uri. - If necessary the port can be included, for example - alice@localhost:5080 - - - - allow-fallback - - Set this to true so Restcomm will fallback to the backup proxy in - case of failed calls - - - - max-failed-calls - - Maximum number of failed call before switching from primary to fallback - outbound proxy - - - - allow-fallback-to-primary - - Allow fallback from Backup to Primary proxy in case fail called at - backup proxy. - - - - -
- - TelScale Restcomm Resource Security - The security model is based on role based access control. - TelScale RestComm defines a set of permissions - that can be defined for each role. There are two predefined roles - Administrator and Developer. The Developer role - can be configured or removed all together depending on your needs but - the Administrator role can not be changed. - The Administrator role can not be modified or removed and it has - access to every resource. - - First we will define the list of permissions and what they - mean. - - TelScale RestComm Permissions - - - - - - Permission - Description - - - - - Create - The role has access to create a type resource. - - - Read - The role has access to read a type of resource. - - - Modify - The role has access to modify a type of resource. - - - Delete - The role has access to delete a type of resource. - - - -
- These permissions apply to every resource exposed by the - Restful APIs. Once a role is defined it can - be specified for new Account resources that are created. - - If no role is specified when creating a new Account resource - then the new Account will inherit - the security role of the account that created it. If this role is the - Administrator role the system - may become compromised. - - - Wildcard Permission - The asterisk '*' is a wildcard that means grant all - permissions and can be used in place of - typing out all the permissions. - - - To see how this all comes together please check out the - restcomm.xml - configuration - file. - -
-
- VoIP Innovations Restful API Access - The default auto provisioning API used by TelScale RestComm. - - - VoIP Innovations Settings - - - - - - Element - Description - - - - - login - The user name configured in the VoIP Innovations Portal - under API users. - - - password - The password configured in the VoIP Innovations Portal - under API users. - - - endpoint - The end point ID of the end point that will be used by - TelScale RestComm - - - uri - The path to the VoIP Innovations service end point. - - - -
-
-
- Dao Manager - The data access object manager is used to gain access to the - data store that TelScale RestComm will use. - - The default data access object manager is based on MyBatis and - provides access to RDBM systems. - - - MyBatis Settings - - - - - - Element - Description - - - - - configuration-file - The path to the mybatis.xml configuration file. - - - data-files - The path to the data files used to store the database - tables. - - - sql-files - The path to the XML files containing the SQL statements - that will be executed by the database. - - - -
-
-
- Media Server Manager - - - The media server manager is responsible for all the media servers - managed by TelScale RestComm. - - - Media Server Manager Settings - - - - - - Property - Description - - - - - local-address - The local IP address for the MGCP stack. - - - local-port - The local port for the MGCP stack. - - - remote-address - The IP address for the media server. - - - remote-port - The port for the media server. - - - response-timeout - - In milliseconds the maximum amount of time to wait for a response - from - the media server before abandoning the request. This does NOT apply - to - RQNT/NOTIFY request/response. - - - - external-address - - Sometimes there is interest to use a different address in the SDP than - the IP address the media server is reporting. This parameter if - set tells TelScale RestComm to - patch the Connection attribute in the SDP on behalf of the media - server to the specified IP address. - Note: TelScale RestComm will only do NAT resolution when necessary so - if your server already has a - routable IP address setting this parameter will have no effect. - - - - -
-
-
- SMS Aggregator - The SMS aggregator is responsible for the handling of SMS - messages on behalf of TelScale RestComm. - - - Requirements - - The default SMS aggregator is SMS over IP as used by VoIP - Innovations. Therefore, an account must be - created @ - http://www.voipinnovations.com - before any text - messages can be sent. - - - - Below is a list of settings that must be configured to send - text messages. - - - SMS Aggregator Settings - - - - - - Element - Description - - - - - outbound-prepend-string - - A string that you would like to prepend to the destination - address. For example, - to use service from VoIP Innovations a '#' must be prepended to - every destination - address. - - - - outbound-endpoint - - The SIP endpoint to which outbound SMS messages should be sent. - Note: Do not prepend sip: to the endpoint. - - - - -
-
-
- Fax Service - The fax service sends faxes out on behalf of TelScale RestComm. - - - Requirements - - The default fax service is provided by Interfax. Therefore, an - account must be created @ - http://www.interfax.net - before any faxes can be - sent. - - - - - Fax Service Settings - - - - - - Element - Description - - - - - user - The user name used to authenticate with Interfax. - - - password - The password used to authenticate with Interfax. - - - -
-
-
- Speech Recognizer - This speech recognizer turns speech into text on behalf of - TelScale Restcomm. - - - Requirements - - The default speech recognizer uses the ASR service provided by - iSpeech. Therefore, an account must be created @ - http://www.ispeech.org - before any speech can be converted to text. - - - - - Speech Recognizer Settings - - - - - - Element - Description - - - - - api-key - The Web API key provided by iSpeech. This is used as an - authentication token for the service. - - - -
-
- - -
- Speech Synthesizer - The speech synthesizer turns text to speech on behalf of - TelScale RestComm. - - - Requirements - - The default speech synthesizer uses the TTS service provided by - VoiceRSS. Therefore, an account must be created @ - http://www.voicerss.org - before any text can be converted to - speech. - - - - - VoiceRSS Speech Synthesizer Settings - - - - - - Element - Description - - - - - service-root - The VoiceRSS service root URI. - - - - apikey - The API key used to authenticate with the VoiceRSS. - - - languages - - A list of male and female speakers for different languages. These - can be replaced - with other alternatives provided by VoiceRSS. - - - - -
-
- - - -
- - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Introduction.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Introduction.xml deleted file mode 100644 index f9ded1aa37..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Introduction.xml +++ /dev/null @@ -1,59 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - Introduction to &SHORT_PLATFORM_NAME; - &SHORT_PLATFORM_NAME; is a turnkey Cloud Communications platform based on Open Source building blocks from the team behind Mobicents. Restcomm offers a clean room implementation of the Twilio.com APIs and much more. - - - -Restcomm can be integrated with VoIP and legacy SS7 network providers either in the cloud or via on-premise Resource Adaptors. - -
- How it Works - In order to demonstrate how &SHORT_PLATFORM_NAME; works we will go over an application that - instructs &SHORT_PLATFORM_NAME; to answer a phone call, say "Hello World" and finally hang up the - call. - - A more thorough explanation of the &SHORT_PLATFORM_NAME; Markup Language is available in Chapter 3. &SHORT_PLATFORM_NAME; - Markup Language - - In this particular example, the first thing that must happen is that TelScale RestComm receives a call. - - - - - - After TelScale RestComm can confirm that the call is destined for an application that it handles. - TelScale RestComm calls out to your application for instructions on how to handle the new incoming call. - Keep in mind that when TelScale RestComm calls out to your application it provides useful information such - as from what number the call was dialed, to what number the call is destined, caller ID information, etc. - - - - - - Once your application responds with instructions on how to handle the call, TelScale RestComm gets busy executing the - provided instruction set. - - <Response> - <Say>Hello World!</Say> - </Response> - - - - - - - In this case TelScale RestComm synthesizes the text to speech, says "Hello World!" to the caller and hangs up. - - - - - -
- Keep in mind that before you continue you should configure your ASR, Fax, SMS, and TTS plugins in order to have - everything function as you would expect it to. - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-RVD.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-RVD.xml deleted file mode 100644 index a851947de4..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-RVD.xml +++ /dev/null @@ -1,177 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - Restcomm Visual Designer - -Restcomm Visual Designer is a practical feature that lets you create applications easily using a GUI based interface. - - - -Please also visit the official Telestax documentation page for more usage tutorials:http://www.telestax.com/docs - -
-Getting Started - -In order to access the designer, open your web browser and go to this url http://127.0.0.1:8080/restcomm-rvd. This page lets you select which type of project you will like to create. There are three options, Voice, USSD and SMS. - - - - - - - - -
- -
-Voice Projects - -In the screenshot below, some demo voice projects and you are also able to create a new project. - - - - - - - - - - -In the screenshot below, you can see the main dashboard of the RVD voice project. It has a drag and drop feature. - - - - - - - - -Please also visit the official Telestax documentation page for more RVD usage tutorials:http://www.telestax.com/docs -
- - - -
-USSD Projects - - -The main page of the USSD Restcomm integration is similar to the screenshot below. You can create new projects or access existing projects from this window. - - - - - - - - - - -In the screenshot below, you can see the main dashboard of the USSD project. It has a drag and drop feature. - - - - - - - - -Please also visit the official Telestax documentation page for more RVD USSD usage tutorials:http://www.telestax.com/docs -
- - - -
-SMS Projects - - -The main page of the SMS Restcomm integration is similar to the screenshot below. You can create new projects or access existing projects from this window. - - - - - - - - - - -In the screenshot below, you can see the main dashboard of the SMS project. It has a drag and drop feature. - - - - - - - - -Please also visit the official Telestax documentation page for more RVD SMS usage tutorials:http://www.telestax.com/docs -
- - - - -
-Demo Application - -In this section, you will learn how to build a sample application. - - -Create a new project called Demo. The result will be similar to the screenshot below. - - - - - - - -Press the save button and click on the Startup Url to make sure the project is correctly saved. The result will be similar to the screenshot below. - - - - - - - -Next you need to go to the Restcomm Admin User Interface http://127.0.0.1:8080/restcomm-management/ . Login and press the menu "Numbers", then press the button, "Register Number". You will see a window similar to the screenshot below. - - - -Leave Account SID as default. Enter a "Phone Number" you want to use. In the demo we shall be using 9999. - - -Under "Voice" -> Voice Request URL, click on the List icon to reveal the Demo application you created using the Restcomm Visual Designer. See screenshot below. - - - -Press "Register" to save your work and return to the list of available phone numbers. - - - - - - - - - -You have just created a new phone number "9999" and have linked it to your Demo application. See below. - - - - - - - - - -You can then start a SIP phone to test your Demo application. Note that you must have enabled Text-to-speech for this application to work. See the Getting started chapter to learn how to configure VoiceRSS. - - - -
- - - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-RestComm_Markup_Language.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-RestComm_Markup_Language.xml deleted file mode 100644 index b286734fe0..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-RestComm_Markup_Language.xml +++ /dev/null @@ -1,1794 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - RestComm Markup Language - The RestComm Markup Language (RCML) is composed of a set of XML tags that can be used to instruct - RestComm on how to handle an on-going phone call. RestComm applications are built by combining these - XML verbs and nouns in a way that is sensible for a given set of application requirements. In this - chapter we will discuss what a request from RestComm to your application looks like as well as what the - response from your application should look like. Then, we will dive in to how each verb and noun in the - XML instruction set is used. -
- RestComm Request - - How RestComm Passes Data to Your Application - The way RestComm passes data to you application depends on the request method for the given - URI. If the request method is GET then the data is passed in the query string or the part after - the question mark. If the request method is POST then the data is sent a multi-part form data just - like when a browser submits a form. - - - RestComm Voice Request - Any time RestComm makes a request to you applications it will include the following data - as request parameters. - - - Request Parameters - - - - - - Parameter - Description - - - - - CallSid - The unique identifier for this call. - - - AccountSid - Your account id. - - - From - The phone number of the originator of the call. - - - To - The phone number of the call recipient. - - - CallStatus - The status of the call. The possible values are queued, ringing, in-progress, completed, busy, failed, and no-answer. - - - ApiVersion - The version of the RestComm API used to handle this call. - - - Direction - The direction of the call. The possible values are inbound and outbound-dial. - - - CallerName - The caller ID for the caller in the case of inbound calls. - - - -
-
-
- Your Response - In your response to the request from RestComm you want to provide RCML that will instruct RestComm on how - to handle the current call. - - MIME Types - RestComm supports the MIME types described in the table below. - - - Supported MIME Types - - - - - - Parameter - Description - - - - - text/xml, - application/xml - RestComm interprets the returned document as an XML instruction set. - - - -
- - When your application returns the RCML document the root element of the document must - always be <Response> or the parser will complain. - -
-
- Say - The <Say> verb is used to synthesize text to speech and play it to the remote party. - The voices supported depends on the TTS Service provider plug-in. Below are the voices for our default - TTS service provider plug-in which uses Acapela Voice and VoiceRSS (default) as a Service. - - Say Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - voice - man, woman - man - - - language - bf, bp, en, en-gb,cf, cs, dan, fi es, fr, de, el, - it, nl, no, pl, pt, ru, ar, ca, sv, tr - en - - - loop - integer > 1 - 1 - - - -
- - voice - The 'voice' attribute allows you to select the gender of the voice used to synthesize - the text to speech for playback. - - - language - The 'language' attribute allows you pick a specific language for speech synthesis. - RestComm currently supports languages 'bf' (Belgium-French), 'bp' (Brazilian-Portugues), 'en' (English), - 'en-gb' (British-English), 'cf' (Canadian-French), 'cs' (Czech), 'dan' (Dannish), 'fi' (Finnish), 'es' (Spanish), - 'fr' (French), 'de' (German), 'el' (Greek), 'it' (Italian), 'nl' (Netherlands-Dutch), 'no' (Norwegian), - 'pl' (Polish), 'pt' (Portuguese), 'ru' (Russian), 'ar' (Saudi-Arabia Arabic), 'ca' (Spain Catalan), - 'sv' (Swedish), and 'tr' (Turkish). - - - loop - The 'loop' attribute specifies how many times you'd like the text repeated. Specifying '0' will - cause the the <Say> verb to loop until the call is hung up. - - - Nesting - The <Say> verb can not have any other verbs or nouns nested. Only text. - - - Examples - For an example of how to use the <Say> verb see below. - - - <Response> - <Say>Hello World</Say> - </Response> - - -The example below shows how you can set the language, voice and loop parameters of the Say Verb. - - - - Bienvenue à RestComm un projet parrainé par TeleStax - - - ]]> - - - - - Usefull Information - - - - When translating text to speech, the Say verb will make assumptions about how to pronounce numbers, dates, times, amounts of money and other abbreviations. - - - - - - - When saying numbers, '12345' will be spoken as "twelve thousand three hundred forty-five." Whereas '1 2 3 4 5' will be spoken as "one two three four five." - - - - - - - - -
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- Play - The <Play> verb is used to play an audio file to the remote party. - - Play Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - loop - integer > 1 - 1 - - - -
- - loop - The 'loop' attribute specifies how many times you'd like the audio file to be repeated. Specifying '0' will - cause the the <Play> verb to loop until the call is hung up. - - - Supported Audio Formats - - - - - - MIME type - Description - - - - - audio/wav - wav format audio - - - audio/wave - wav format audio - - - audio/x-wav - wav format audio - - - -
- -Media Server Audio File Format - -The recommended audio file format is linear( 8000,16 bit , Mono ). - - - - - Nesting - The <Play> verb can not have any other verbs or nouns nested. Only a URL. - - - - - Basic example - For an example of how to use the <Play> verb see below. - - - <Response> - <Play>http://foobar.com/demo.wav</Play> - </Response> - - - -
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- Gather - The <Gather> verb "gathers" digits that a caller enters into his or her telephone keypad. When the caller - is done entering digits, RestComm submits that digits to the provided 'action' URL in an HTTP GET or POST request. - If no input is received before timeout, <Gather> falls through to the next verb in the RestComm document. - You may optionally nest <Say>, <Play>, and <Pause> verbs within a <Gather> verb while - waiting for input. This allows you to read menu options to the caller while letting her enter a menu selection - at any time. After the first digit is received the audio will stop playing. - - Gather Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - action - relative or absolute URL - current document URL - - - method - GET, POST - POST - - - timeout - positive integer - 5 seconds - - - finishOnKey - any digit, #, * - # - - - numDigits - integer >= 1 - unlimited - - - -
- - action - The 'action' attribute takes an absolute or relative URL as a value. When the caller has finished - entering digits RestComm will make a GET or POST request to this URL including the parameters below. - If no 'action' is provided, RestComm will by default make a POST request to the current document's URL. - - - Request Parameters - - - - - - Parameter - Description - - - - - Digits - The digits the caller pressed, excluding the finishOnKey digit. - - - -
- - method - The 'method' attribute takes the value 'GET' or 'POST'. This tells RestComm whether to request the 'action' URL - via HTTP GET or POST. - - - timeout - The 'timeout' attribute sets the limit in seconds that RestComm will wait for the caller to press another digit before - moving on and making a request to the 'action' URL. For example, if 'timeout' is '10', RestComm will wait ten seconds for - the caller to press another key before submitting the previously entered digits to the 'action' URL. RestComm waits until - completing the execution of all nested verbs before beginning the timeout period. - - - finishOnKey - The 'finishOnKey' attribute lets you choose one value that submits the received data when entered. For example, if you - set 'finishOnKey' to '#' and the user enters '1234#', RestComm will immediately stop waiting for more input when the '#' is - received and will submit "Digits=1234" to the 'action' URL. Note that the 'finishOnKey' value is not sent. The allowed - values are the digits 0-9, '#' , '*' and the empty string (set 'finishOnKey' to ''). If the empty string is used, - <Gather> captures all input and no key will end the <Gather> when pressed. In this case RestComm will submit - the entered digits to the 'action' URL only after the timeout has been reached. The value can only be a single character. - - - numDigits - The 'numDigits' attribute lets you set the number of digits you are expecting, and submits the data to the 'action' URL - once the caller enters that number of digits. - - - Nesting - You can nest the following verbs within <Gather>: <Say>, <Play>, <Pause> - - - Examples - For an example of how to use the <Gather> verb see below. - - - <Response> - <Gather action="handle-user-input.php" numDigits="1"> - <Say>Welcome to TPS.</Say> - <Say>For store hours, press 1.</Say> - <Say>To speak to an agent, press 2.</Say> - <Say>To check your package status, press 3.</Say> - </Gather> - <!-- If customer doesn't input anything, prompt and try again. --> - <Say>Sorry, I didn't get your response.</Say> - <Redirect>handle-incoming-call.xml</Redirect> - </Response> - -
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- Record - The <Record> verb records the caller's voice and returns to you the URL of a file containing the audio recording. - - Record Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - action - relative or absolute URL - current document URL - - - method - GET, POST - POST - - - timeout - positive integer - 5 - - - finishOnKey - any digit, #, * - 1234567890*# - - - maxLength - integer greater than 1 with the number of seconds to wait - 3600 (1 hour) - - - transcribe - true, false - false - - - transcribeCallback - relative or absolute URL - none - - - playBeep - true, false - true - - - -
- - action - The 'action' attribute takes an absolute or relative URL as a value. When recording is finished - RestComm will make a GET or POST request to this URL including the parameters below. If no 'action' is - provided, <Record> will default to requesting the current document's URL. After making this request, - RestComm will continue the current call using the RCML received in your response. Any RCML verbs occuring - after a <Record> are unreachable. There is one exception: if RestComm receives an empty recording, - it will not make a request to the 'action' URL. The current call flow will continue with the next verb in - the current RCML document. - - - Request Parameters - - - - - - Parameter - Description - - - - - RecordingUrl - The URL of the recorded audio. - - - RecordingDuration - The time duration of the recorded audio. - - - Digits - The digits the caller pressed, excluding the finishOnKey digit. - - - -
- - method - The 'method' attribute takes the value 'GET' or 'POST'. This tells RestComm whether to - request the URL via HTTP GET or POST. - - - timeout - The 'timeout' attribute tells RestComm to end the recording after a number of seconds of silence has passed. - - - finishOnKey - The 'finishOnKey' attribute lets you choose a set of digits that end the recording when entered. For example, if you - set 'finishOnKey' to '#' and the caller presses '#', RestComm will immediately stop recording and submit 'RecordingUrl', - 'RecordingDuration', and the '#' as parameters in a request to the 'action' URL. The allowed values are the digits 0-9, '#' - and '*'. Unlike <Gather>, you may specify more than one character as a 'finishOnKey' value. - - - maxLength - The 'maxLength' attribute lets you set the maximum length for the recording in seconds. - - - transcribe - The 'transcribe' attribute tells RestComm that you would like a text representation of the audio of the recording. - - - transcribeCallback - The 'transcribeCallback' attribute is used in conjunction with the 'transcribe' attribute. It allows you to specify a URL - to which RestComm will make an asynchronous POST request when the transcription is complete. This is not a request for RCML - and the response will not change call flow, but the request will contain the standard RCML request parameters as well as - 'TranscriptionStatus', 'TranscriptionText', 'TranscriptionUrl' and 'RecordingUrl'. If 'transcribeCallback' is specified, - then there is no need to specify 'transcribe=true'. It is implied. If you specify 'transcribe=true' without a 'transcribeCallback', - the completed transcription will be stored for you to retrieve later (see the REST API Transcriptions section), but RestComm will - not asynchronously notify your application. - - - Request Parameters - - - - - - Parameter - Description - - - - - TranscriptionText - Contains the text of the transcription. - - - TranscriptionStatus - The status of the transcription attempt: either 'completed' or 'failed'. - - - TranscriptionUrl - The URL for the transcription's REST API resource. - - - RecordingUrl - The URL for the transcription's source recording resource. - - - -
- - playBeep - The 'playBeep' attribute allows you to toggle between playing a sound before the start of a recording. - - - Nesting - The <Record> verb can not have any other verbs or nouns nested. - - - Examples - For an example of how to use the <Record> verb see below. - - - <Response> - <Record maxLength="30"/> - </Response> - -
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- Fax - The <Fax> verb sends a fax to some a fax machine. - - Fax Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - to - phone number - see below - - - from - phone number - see below - - - action - relative or absolute URL - none - - - method - GET, POST - POST - - - statusCallback - relative or absolute URL - none - - - -
- - to - The 'to' attribute takes a valid E.164 phone number as a value. RestComm will send a fax to this number. - When sending a fax during an incoming call, 'to' defaults to the caller. When sending an fax during an outgoing call, - 'to' defaults to the called party. The value of 'to' must be a valid phone number. - - - from - The 'from' attribute takes a valid E.164 phone number as an argument. When sending a fax during an incoming call, - 'from' defaults to the calling party. When sending a fax during an outgoing call, 'from' defaults to the called party. - The value of 'from' must be a valid phone number. - - - action - The 'action' attribute takes a URL as an argument. After processing the <Fax> verb, RestComm will make a GET - or POST request to this URL with the form parameters 'FaxStatus' and 'FaxSid'. Using an 'action' URL, your application - can receive synchronous notification that the message was successfully enqueued. If you provide an 'action' URL, RestComm - will use the RCML received in your response to the 'action' URL request to continue the current call. Any RCML verbs - occuring after a <Fax> which specifies an 'action' attribute are unreachable. If no 'action' is provided, <Fax> - will finish and RestComm will move on to the next RCML verb in the document. If there is no next verb, RestComm will end the - phone call. - - - Request Parameters - - - - - - Parameter - Description - - - - - FaxSid - The Sid for the Sms message. - - - FaxStatus - The current status of the Sms message. Either 'sent' or 'failed'. - - - -
- - method - The 'method' attribute takes the value 'GET' or 'POST'. This tells RestComm whether to request the 'action' URL via HTTP GET - or POST. This attribute is modeled after the HTML form 'method' attribute. - - - statusCallback - The 'statusCallback' attribute takes a URL as an argument. When the fax is actually sent, or if sending fails, - RestComm will make an asynchronous POST request to this URL with the parameters 'FaxStatus' and 'FaxSid'. - Note, 'statusCallback' always uses HTTP POST to request the given url. - - - Request Parameters - - - - - - Parameter - Description - - - - - FaxSid - The Sid for the fax message. - - - FaxStatus - The current status of the fax. Either 'sent' or 'failed'. - - - -
- - Nesting - The <Fax> verb can not have any other verbs or nouns nested. - - - Examples - For an example of how to use the <Fax> verb see below. - - - <Response> - <Fax>This is a test fax.</Fax> - </Response> - -
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- Sms - The <Sms> verb sends an SMS message to a phone number during a phone call. - - Sms Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - to - phone number - see below - - - from - phone number - see below - - - action - relative or absolute URL - none - - - method - GET, POST - POST - - - statusCallback - relative or absolute URL - none - - - -
- - to - The 'to' attribute takes a valid E.164 phone number as a value. RestComm will send an SMS message to this number. - When sending an SMS during an incoming call, 'to' defaults to the caller. When sending an SMS during an outgoing call, - 'to' defaults to the called party. The value of 'to' must be a valid phone number. - - - from - The 'from' attribute takes a valid E.164 phone number as an argument. When sending an SMS during an incoming call, - 'from' defaults to the calling party. When sending an SMS during an outgoing call, 'from' defaults to the called party. - - - action - The 'action' attribute takes a URL as an argument. After processing the <Sms> verb, RestComm will make a GET - or POST request to this URL with the form parameters 'SmsStatus' and 'SmsSid'. Using an 'action' URL, your application - can receive synchronous notification that the message was successfully enqueued. If you provide an 'action' URL, RestComm - will use the RCML received in your response to the 'action' URL request to continue the current call. Any RCML verbs - occuring after an <Sms> which specifies an 'action' attribute are unreachable. If no 'action' is provided, <Sms> - will finish and RestComm will move on to the next RCML verb in the document. If there is no next verb, RestComm will end the - phone call. - - - method - The 'method' attribute takes the value 'GET' or 'POST'. This tells RestComm whether to request the 'action' URL via HTTP GET - or POST. This attribute is modeled after the HTML form 'method' attribute. - - - statusCallback - The 'statusCallback' attribute takes a URL as an argument. When the SMS message is actually sent, or if sending fails, - RestComm will make an asynchronous POST request to this URL with the parameters 'SmsStatus' and 'SmsSid'. - Note, 'statusCallback' always uses HTTP POST to request the given url. - - - Request Parameters - - - - - - Parameter - Description - - - - - SmsSid - The Sid for the Sms message. - - - SmsStatus - The current status of the Sms message. Either 'sent' or 'failed'. - - - -
- - Nesting - The <Sms> verb can not have any other verbs or nouns nested. - - - Examples - For an example of how to use the <Sms> verb see below. - - - <Response> - <Sms>Hello World!</Sms> - </Response> - - - - Sms Custom headers - - Restcomm will check for any custom headers and make use of them. - - - In case of an incoming SMS (incoming SIP MESSAGE) Restcomm will scan the message for any headers starting with X- and if any found will be send to - the application server with the request (part of the query string in the case of HTTP GET or part of the message body in the case of HTTP POST) - - - The application server can also provide some custom http headers (again starting with X-) along with the RCML response, that Restcomm will store and use them - when creating the outgoing SMS message (in the case of RCML response is Sms). - - - -
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- Dial - The <Dial> verb connects the current caller to another phone. If the called party picks up, the two parties are - connected and can communicate until one hangs up. If the called party does not pick up, if a busy signal is received, or if - the number doesn't exist, the dial verb will finish. - - Dial Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - action - relative or absolute URL - no default for <Dial> - - - method - GET, POST - POST - - - timeout - positive integer in seconds - 30 seconds - - - timeLimit - positive integer (seconds) - 14400 seconds (4 hours) - - - callerId - a valid phone number, or client identifier if you are dialing a <Client>. - Caller's callerId - - - record - true, false - false - - - -
- - action - The 'action' attribute takes a URL as an argument. When the dialed call ends, RestComm will make a GET or POST - request to this URL including the parameters below. If you provide an 'action' URL, RestComm will continue the current - call after the dialed party has hung up, using the RCML received in your response to the 'action' URL request. - Any RCML verbs occuring after a <Dial> which specifies an 'action' attribute are unreachable. If no 'action' - is provided, <Dial> will finish and RestComm will move on to the next RCML verb in the document. - If there is no next verb, RestComm will end the phone call. - - - Request Parameters - - - - - - Parameter - Description - - - - - DialCallStatus - The outcome of the <Dial> attempt. See the DialCallStatus section below for details. - - - DialCallSid - The call sid of the new call leg. This parameter is not sent after dialing a conference. - - - DialCallDuration - The duration in seconds of the dialed call. This parameter is not sent after dialing a conference. - - - - RecordingUrl - The URL of the recorded audio. This parameter is only sent if record="true" is set on the Dial verb, and does not include recordings from the Record verb or Record=True on REST API calls.. - - - - -
- - DialCallStatus Values - - - - - - Parameter - Description - - - - - completed - The called party answered the call and was connected to the caller. - - - busy - RestComm received a busy signal when trying to connect to the called party. - - - no-answer - The called party did not pick up before the timeout period passed. - - - failed - RestComm was unable to route to the given phone number. This is frequently caused by dialing a - properly formated but non-existent phone number. - - - canceled - The call was canceled via the REST API before it was answered. - - - -
- - method - The 'method' attribute takes the value 'GET' or 'POST'. This tells RestComm whether to request the 'action' - URL via HTTP GET or POST. This attribute is modeled after the HTML form 'method' attribute. - - - timeout - The 'timeout' attribute sets the limit in seconds that <Dial> waits for the called party to answer the call. - - - timelimit - The 'timeLimit' attribute sets the maximum duration of the <Dial> in seconds. - - - callerId - The 'callerId' attribute lets you specify the caller ID that will appear to the called party when RestComm calls. - By default, when you put a <Dial> in your RCML response to RestComm's inbound call request, the caller ID that - the dialed party sees is the inbound caller's caller ID. If you are dialing to a <Client>, you can set a client - identifier as the callerId attribute. For instance, if you've set up a client for incoming calls and you are dialing to - it, you could set the callerId attribute to client:thomas. - - - record - The 'record' attribute lets you specify whether the call will be recorded or not. - By default, the call is not recorded. If you set the attribute to 'true' Restcomm will start recording when the two calls are bridged - - -Record attribute for Dial Conference - -The current release of Restcomm doesn't support recording for Dial Conference. This feature will be available in the next release - - - - Nesting - You can nest the following nouns within <Dial>: <Number>, <Client>, <Conference> - - - Examples - For examples of how to use the <Dial> verb see below. - - - <Response> - <Dial>1-444-555-666</Dial> - </Response> - - - <Response> - <Dial record="true">1-444-555-666</Dial> - </Response> - - - <Response> - <Dial callerId="1555666777" record="true">1-444-555-666</Dial> - </Response> - -
- Number - The <Number> noun specifies a phone number to dial. You can use multiple <Number> nouns within a - <Dial> verb to simultaneously call all of them at once. The first call to pick up is connected to the current call and - the rest are hung up. - - Number Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - sendDigits - any digits - none - - - url - any url - none - - - -
- - sendDigits - The 'sendDigits' attribute tells RestComm to play DTMF tones when the call is answered. - This is useful when dialing a phone number and an extension. RestComm will dial the number, and - when the automated system picks up, send the DTMF tones to connect to the extension. - - - url - The 'url' attribute allows you to specify a url for a RCML document that will run on the - called party's end, after he/she answers, but before the parties are connected. You can use this RCML - to privately play or say information to the called party, or provide a chance to decline the phone call - using <Gather> and <Hangup>. The current caller will continue to hear ringing while the - RCML document executes on the other end. RCML documents executed in this manner are not allowed to contain - the <Dial> verb. - - - Examples - For an example of how to use the <Number> noun see below. - - - <Response> - <Dial> - <Number sendDigits="wwww1234">1-444-555-6666</Number> - </Dial> - </Response> - -
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- Client - The <Client> noun specifies a client identifier to dial. You can use multiple <Client> nouns within - a <Dial> verb to simultaneously attempt a connection with many clients at once. The first client to accept the - incoming connection is connected to the call and the other connection attempts are canceled. - - Examples - For an example of how to use the <Client> none see below. - - - <Response> - <Dial> - <Client>thomas</Client> - </Dial> - </Response> - -
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- Conference - The <Conference> noun allows you to connect to a named conference room and talk with the other callers - who have also connected to that room. The name of the room is up to you and is namespaced to your account. - - Conference Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - muted - true, false - false - - - beep - true, false - true - - - startConferenceOnEnter - true, false - true - - - endConferenceOnExit - true, false - false - - - waitUrl - RCML url, empty string - default RestComm hold music - - - waitMethod - GET or POST - POST - - - maxParticipants - positive integer <= 40 - 40 - - - -
- - muted - The 'muted' attribute lets you specify whether a participant can speak in the conference. - If this attribute is set to 'true', the participant will only be able to listen to people in the conference. - - - beep - The 'beep' attribute lets you specify whether a notification beep is played to the conference when a participant - joins or leaves the conference. - - - startConferenceOnEnter - This attribute tells a conference to start when this participant joins the conference, if it is not already started. - If this is false and the participant joins a conference that has not started, they are muted and hear background music - until a participant joins where startConferenceOnEnter is true. This is useful for implementing moderated conferences. - - - endConferenceOnExit - If a participant has this attribute set to 'true', then when that participant leaves, the conference ends and all - other participants drop out. This is useful for implementing moderated conferences that bridge two calls and allow either - call leg to continue executing RCML if the other hangs up. - - - waitUrl - The 'waitUrl' attribute lets you specify a URL for music that plays before the conference has started. The URL may - be a WAV or a RCML document that uses <Play> or <Say> for content. This defaults to a selection of Creative - Commons licensed background music, but you can replace it with your own music and messages. If the 'waitUrl' responds with - RCML, RestComm will only process <Play>, <Say>, and <Redirect> verbs. If you do not wish anything to play - while waiting for the conference to start, specify the empty string (set 'waitUrl' to ''). - - - waitMethod - This attribute indicates which HTTP method to use when requesting 'waitUrl'. It defaults to 'POST'. Be sure to use - 'GET' if you are directly requesting static audio files such as WAV files so that RestComm properly caches the files. - - - maxParticipants - This attribute indicates the maximum number of participants you want to allow within a named conference room. The default - maximum number of participants is 40. The value must be a positive integer less than or equal to 100. - - - Examples - For an example of how to use the <Conference> noun see below. - - - <Response> - <Dial> - <Conference>1234</Conference> - </Dial> - </Response> - -
- - - - Uri noun deprecated - The Uri noun has been deprecated and will no longer be available in future releases. Please use the SIP noun instead. - - -
- Sip - The <Sip> noun specifies a SIP URI to dial. You can use multiple <Sip> nouns within - a <Dial> verb to simultaneously attempt a connection with many user agents at once. The first - user agent to accept the incoming connection is connected to the call and the other connection attempts - are canceled. - - The Dial verb's Sip noun lets you set up VoIP sessions by using SIP -- Session Initiation Protocol. With this feature, you can send a call to any SIP endpoint. Set up your RCML to use the Sip noun within the Dial verb. - - -Currently, only one Sip noun may be specified per Dial, and the INVITE message may be sent to only one SIP endpoint. Also, you cannot add any other nouns (eg Number, Client) in the same Dial as the SIP. If you want to use another noun, set up a callback on the Dial to use alternate methods. - - - Examples - For an example of how to use the <Sip> noun see below. - - - <Response> - <Dial> - <Sip>sip:alice@127.0.0.1:5080</Sip> - </Dial> - </Response> - - - -Authentication - - - -Send username and password attributes for authentication to your SIP infrastructure as attributes on the Sip noun. - - - - Request Parameters - - - - - - Attribute Name - Values - - - - - username - Username for SIP authentication. - - - password - Password for SIP authentication - - - - -
- - -Authentication Example - - - - - - - sip:alice@example.com - - - -]]> - - - - -Custom headers - - - -Send custom headers by appending them to the SIP URI -- just as you'd pass headers in a URI over HTTP. For example: - - - - - - - - sip:alice@example.com?mycustomheader=tata&myotherheader=toto - - - - -]]> - - - -Character Limit -While the SIP URI itself must be under 255 chars, the headers must be under 1024 characters. - - - - -Transport - - - -Set a parameter on your SIP URI to specify what transport protocol you want to use. Currently, this is limited to TCP and UDP. By default, Restcomm sends your SIP INVITE over UDP. Change this by using the transport parameter: - - - - - - - - - - sip:alice@example.com;transport=tcp - - - - -]]> - - - - -Attributes - - - - - Request Parameters - - - - - - - Attribute Name - Allowed Values - Default Value - - - - - url - call screening url. - none. - - - method - GET, POST - POST - - - - -
- - - - -The url attribute allows you to specify a url for a RCML document that runs on the called party's end, after they answer, but before the two parties are connected. You can use this RCML to privately Play or Say information to the called party, or provide a chance to decline the phone call using Gather and Hangup. The current caller continues to hear ringing while the RCML document executes on the other end. RCML documents executed in this manner cannot contain the Dial verb. - - - -method - - - -The method attribute allows you to specify which HTTP method Restcomm should use when requesting the URL specified in the url attribute. The default is POST. - - - - -Call Screening HTTP parameters - - - - -When a call is answered, Restcomm passes the following parameters with its request to your screening URL (in addition to the standard RCML Voice request parameters): - - - - - - Request Parameters - - - - - - Attribute Name - Values - - - - - - SipCallId - The SIP call ID header of the request made to the remote SIP infrastructure. - - - SipHeader - The name/value of any X-headers returned in the 200 response to the SIP INVITE request. - - - - -
- - - -Dial Action HTTP parameters - - - -Use the action callback parameters to modify your application based on the results of the SIP dial attempt: - - - - - Request Parameters - - - - - - Attribute Name - Values - - - - - - DialSipCallId - The SIP call ID header of the request made to the remote SIP infrastructure. - - - DialSipResponseCode - The SIP response code as a result of the INVITE attempt. - - - - DialSipHeader_ - The name/value of any X-headers returned in the final response to the SIP INVITE request. - - - - -
- - -Dial with Multiple Examples - - - -A more complex Dial, specifying custom settings as attributes on Dial, including call screening and setting the protocol to TCP. - - - - - - - - sip:alice@example.com?customheader=foo - - - - -]]> - - - -
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- Hangup - The <Hangup> verb ends a call. - - Nesting - The <Hangup> verb can not have any other verbs or nouns nested. - - - Examples - For an example of how to use the <Hangup> verb see below. - - - <Response> - <Hangup/> - </Response> - -
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- Redirect - The <Redirect> verb transfers control of a call to the RCML at a different URL. - All verbs after <Redirect> are unreachable and ignored. - - Redirect Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - method - GET, POST - POST - - - -
- - method - The 'method' attribute takes the value 'GET' or 'POST'. This tells RestComm whether to - request the URL via HTTP GET or POST. - - - Nesting - The <Redirect> verb can not have any other verbs or nouns nested. - - - Examples - For an example of how to use the <Redirect> verb see below. - - - <Response> - <Redirect>http://foobar.com/instructions</Redirect> - </Response> - -
-
- Reject - The <Reject> verb rejects an incoming call to your RestComm endpoint. This is useful for blocking unwanted calls. - If the first verb in a RCML response is <Reject>, RestComm will not pick up the call. The call ends with a status of - 'busy' or 'no-answer', depending on the verb's 'reason' attribute. Any verbs after <Reject> are unreachable and ignored. - - Reject Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - reason - rejected, busy - rejected - - - -
- - reason - The reason attribute takes the values "rejected" and "busy." This tells RestComm what message to play when - rejecting a call. Selecting "busy" will play a busy signal to the caller, while selecting "rejected" will - play a standard not-in-service response. - - - Nesting - The <Reject> verb can not have any other verbs or nouns nested. - - - Examples - For an example of how to use the <Reject> verb see below. - - - <Response> - <Reject reason="busy"/> - </Response> - -
-
- Pause - The <Pause> verb waits silently for a specific number of seconds. If <Pause> is the first verb - in a RCML response, RestComm will wait the specified number of seconds before picking up the call. - - Pause Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - length - integer > 0 - 1 second - - - -
- - length - The 'length' attribute specifies how many seconds RestComm will wait silently before continuing on. - - - Nesting - The <Pause> verb can not have any other verbs or nouns nested. - - - Examples - For an example of how to use the <Pause> verb see below. - - - <Response> - <Pause length="5"/> - </Response> - -
- - - -
- UssdCollect - The UssdCollect verb is a USSD integration with Restcomm resource. When fully configured with a USSD gateway, Restcomm will send all UssdMessage to the configured gateway - - - - Example - For an example of how to use the USSDCollect verb see below. - - - - - 1 for first option - 2 for first option - 3 for first option - - - ]]> - - - -
-UssdMessage - - - -Before you can send an USSD message you must first configure the restcomm.xml with the USSD gateway information. See for more information. - - - The UssdMessage verb contains the message that Restcomm will send to the USSD gateway - - - Nesting - The UssdMessage can be nested inside of the USSDCollect verb. - - - - Examples - For an example of how to use the USSDCollect verb see below. - - - - Test USSD Message - - ]]> - - -
- -
-Language - The language noun let you specify the language of the USSD message that is sent to the USSD gateway - - - - - Language Attributes - - - - - - - Name - Allowed Values - Default Value - - - - - Language - bf, bp, en, en-gb,cf, cs, dan, fi es, fr, de, el, it, nl, no, pl, pt, ru, ar, ca, sv, th, tr - en - - - -
- - - Nesting - The Language noun cannot be nested. If not specified, English language is default - - - Example - For an example of how to use the USSD Language noun see below. - - - - fr - Test USSD Message - - ]]> - - -
- -
- - - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Restful-API.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Restful-API.xml deleted file mode 100644 index ef98657df4..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Restful-API.xml +++ /dev/null @@ -1,2901 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - Restful APIs - The RestComm Restufl APIs are very similar to Twilio's and allow - you to query meta-data about your account, phone numbers, - calls, text - messages, and recordings. Through the Restful API you can also - start - outbound calls and send text messages. - - Resource Encoding - When an HTTP client makes a request to RestComm for a resource - the HTTP client has the option to pick which encoding - is used for the - response. Currently, RestComm supports XML and JSON - encoding. The - default encoding is XML but - to receive the resource in JSON just - append .json to the end of the - resource name. - - - - User Authentication - Before accepting any Restful API call, Restcomm will - authenticate the request source using either a valid Account SID or - an Email Account combined with an Authentication Token. - - See below for the default Administrator account. - - - - - - - - or - - - - - - - - -
- Accounts - Accounts and sub-accounts are useful for things like segmenting - phone numbers and usage data for your users and - controlling access to - data. - - Account Resource URI - /2012-04-24/Accounts/{AccountSid} - - - Resource Properties - - - - - - Property - Description - - - - - Sid - A string that uniquely identifies this account. - - - DateCreated - The date that this account was created. - - - DateUpdated - The date that this account was last updated. - - - FriendlyName - A description of this account, up to 64 characters long. By - default - the FriendlyName is your email address. - - - Status - The status of this account. Possible values are active, - suspended, and closed. - - - AuthToken - The authorization token for this account. This should not - be shared. - - - Uri - The URI for this account, relative to - http://localhost:port/restcomm. - - - -
-
- Supported Operations - - HTTP GET - Returns the representation of an Account resource, including - the properties above. - - - - - - - Account Resource URI - /2012-04-24/Accounts/{EmailAddress} - - - HTTP POST/PUT - Modifies an Account resource and returns the representation, - including the properties above. Below - you will find a list of - optional parameters. - - - Request Parameters - - - - - - Parameter - Description - - - - - FriendlyName - A description of this account, up to 64 characters long. - - - Status - The status of this account. Possible values are active, - suspended, and closed. - - - Password - A password that will be used to generate the AuthToken for - the new Account resource. - - - -
- - - Get information about the default account - - - - - - - - Change default account password(AuthToken) - - - - - - - Command line versus Browser - - The above command uses the Account SID and the one below uses the - Email Account. Note the - administrator%40company.com - is used instead of - administrator@company.com - . This is because using curl on the bash terminal doesn't parse the - @ correctlyl. If you were to running on a browser, you can safely - use the @ as the web browser will correctly handle it. - - - - - - - The above commands will print an output similar to the one - below: - - - - ACae6e420f425248d6a26948c17a9e2acf - Default Administrator Account - active - Full - 2012-04-24T00:00:00.000-06:00 - 2012-04-24T00:00:00.000-06:00 - 77f8c12cc7b8f8423e5c38b035249166 - /2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf - - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/AvailablePhoneNumbers - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Calls - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Conferences - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/IncomingPhoneNumbers - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Notifications - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/OutgoingCallerIds - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Recordings - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Sandbox - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/SMS/Messages - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Transcriptions - - - -]]> - - - -
-
- Account List Resource - - Account List Resource URI - /2012-04-24/Accounts - -
- Supported Operations - - HTTP GET - Returns the list representation of all the Sub-Account - resources for this Account, including the - properties above. - - - - - - - - - HTTP POST - Creates a new Sub-Account and returns the representation of - the Sub-Account resource, including - the properties above. Below you - will find a list of required and - optional parameters. - - - - - - Request Parameters - - - - - - Parameter - Description - - - - - EmailAddress(Required) - The email address to use for this account. - - - FriendlyName - A description of this account, up to 64 characters long. - Default, is your email address. - - - Status - The status of this account. Default is active, possible - values are active, suspended, and closed. - - - Password(Required) - A password that will be used to generate the AuthToken - for the new Account resource. - - - Role(Required) - The security role that this Account resource will use. If - no role is provided then - the role of the account resource - creating this will be inherited - to the new Account resource - and - may compromise the system. - - - -
- - Here is an example of how to createa a sub-account. The - sub-account will inherit the same permissions has the - Administrator's account. - - - - - - About Sub-accounts - Note that the SID, Email and the AuthToken (see output below) - of the sub-account can now be used instead of the Administrator's - account - - - - - AC3b8f0dd2e5026abde018446cbb3b185d - MySubAccount - active - Full - 2013-10-16T09:22:28.708-06:00 - 2013-10-16T09:22:28.712-06:00 - 53134d7a9914e2b47c8435ebdb50ded3 - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d - - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/AvailablePhoneNumbers - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/Calls - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/Conferences - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/IncomingPhoneNumbers - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/Notifications - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/OutgoingCallerIds - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/Recordings - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/Sandbox - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/SMS/Messages - /restcomm/2012-04-24/Accounts/AC3b8f0dd2e5026abde018446cbb3b185d/Transcriptions - - - -]]> - - - - -
-
-
-
- AvailablePhoneNumbers - The AvailablePhoneNumbers subresources let you search for - incoming local and toll-free phone numbers that are available - for you - to purchase from a TeleStax partner. - - AvailablePhoneNumbers List Resource URI - /2012-04-24/Accounts/{AccountSid}/AvailablePhoneNumbers/US/Local - - - Searching For Numbers. - When using RestComm the way to search for new phone numbers is - by searching - the AvailablePhoneNumbers list resource and providing - the desired area - code as a filter. - - - Resource Properties - - - - - - Property - Description - - - - - FriendlyName - A friendly version of the phone number. - - - PhoneNumber - The phone number, in E.164 format. - - - Lata - The LATA for this phone number. - - - RateCenter - The rate center for this phone number. - - - Latitude - The latitude coordinate for this phone number. - - - Longitude - The longitude coordinate for this phone number. - - - Region - The two-letter state or province abbreviation for this - phone number. - - - PostalCode - The zip code for this phone number. - - - IsoCountry - The ISO country code for this phone number.. - - - -
- - - - -
- Supported Operations - - HTTP GET - Returns the representation of an AvailablePhoneNumber - resource, including the properties above. - - - Request Parameters - - - - - - Property - Description - - - - - AreaCode - A three digit area code inside the U.S. - - - -
- - - Querying Available Phone Numbers - - You need to be using RestComm for VoIP Innovations in order to be - able to use this feature. See here for more details - RestComm AMI for VoIP Innovation - - - - - Here is an example, the AreaCode is any valid United States - Code - -:8080/restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/AvailablePhoneNumbers/US/Local -d "AreaCode=305" - -]]> - - - -
-
- -
- Gateways - The Gateways subresources let you create sip accounts that Restcomm will use to register itself to the Gateway and receive incoming traffic. - - - Gatewat List Resource URI - /2012-04-24/Accounts/{AccountSid}/Management/Gateways - - - Register Restcomm instance to a SIP Gateway. - You might need to register Restcomm instnace to a SIP Gateway and receive incoming traffic from that gateway. - For that you need to use Gateway REST endpoint - - - Resource Properties - - - - - - Property - Description - - - - - FriendlyName - A friendly version of the gateway. - - - UserName - The username that will be used to register to this gateway - - - Password - The password that will be used to register to this gateway - - - Proxy - The proxy address of the gateway - - - Register - Boolean flag to register or not the gateway - - - TTL - Time to live for the Register - - - -
- -
- Supported Operations - - HTTP GET - Returns the representation of a Gateway - resource, including the properties above. - Resource URI: - /2010-04-01/Accounts/{AccountSid}/Management/Gateways/{GatewaySid} - - - HTTP POST - Creates a new Gateway - resource and returns the representation of the resource, including the properties above. - Resource URI: - /2010-04-01/Accounts/{AccountSid}/Management/Gateways - - - HTTP POST/PUT - Update a Gateway - resource and returns the representation of the resource, including the properties above. - Resource URI: - /2010-04-01/Accounts/{AccountSid}/Management/Gateways/{GatewaySid} - - - HTTP DELETE - Deletes a Gateway resource. - Resource URI: - /2010-04-01/Accounts/{AccountSid}/Management/Gateways/{GatewaySid} - -
- -
- Examples - - Create a new Gateway - -:8080/restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Management/Gateways -d "FriendlyName=mygateway" -d "UserName=username" -d "Password=password" -d "Proxy=my.gateway.com" -d "Register=true" -d "TTL=3600" - -]]> - - - - - Get a list of available Gateways - -:8080/restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Management/Gateways - -]]> - - - - - Update an existing Gateway - -:8080/restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Management/Gateways/GW106bc6f34bd24790a435eaeccc1aed72 -d "FriendlyName=MyGatewayNewName" -d "UserName=newUserName" - -]]> - - - - - Delete an existing Gateway - -:8080/restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Management/Gateways/GW1cffb069192a45f2b5f5af2e76489550 - -]]> - - - -
-
- -
- Clients - An Client instance resource represents a user agent registered - with RestComm. - - Client Resource URI - /2012-04-24/Accounts/{AccountSid}/Clients/{ClientSid} - - - Using SIP User Agents - When using RestComm to handle SIP user agent you have to create - a new Client resource, - this resource acts as an account for your user - agent and also dictates - how calls made by the - user agent should be - handled. - - - - Client without VoiceUrl - Restcomm has a new implied behavior when VoiceUrl is not - provided for a Client account. Restcomm will proxy calls from such - Clients to the destination Client (only if registered) or to the - destination Application DID. - - - Only registered Clients are allowed to use the B2BUA/P2P/Proxy - functionalities of Restcomm. Proxying and P2P calls are only - allowed between registered(authenticated) Clients. - - - - Resource Properties - - - - - - Property - Description - - - - - Sid - A string that uniquely identifies this client. - - - DateCreated - The date that this client was created. - - - DateUpdated - The date that this clientr was last updated. - - - FriendlyName - A friendly name for this client. - - - AccountSid - The unique id of the Account that owns this phone number. - - - ApiVersion - Calls to this phone number will create a new RCML session - with this API version. - - - Login - The name that is used inside the <Client> noun. This - is also used by - the user agent as the user name used for - registration and outbound - dialing. - - - Password - The password used by the user agent during registration and - outbound dialing. - - - Status - The client status the possible values are 0 for disabled - and 1 for enabled. - - - VoiceUrl - The URL RestComm will request when this client makes an - outbound call. - - - VoiceMethod - The HTTP method RestComm will use when requesting the above - Url. Either GET or POST. - - - VoiceFallbackUrl - The URL that RestComm will request if execution of VoiceUrl - fails for any reason. - - - VoiceFallbackMethod - The HTTP method RestComm will use when requesting the - VoiceFallbackUrl. Either GET or POST. - - - VoiceApplicationSid - If this entry contains an Sid to a voice application then - RestComm will ignore these voice URLs - and use the voice URLs - specified by the voice application. - - - StatusCallback - The URL that RestComm will request to pass status - parameters (such as the call state) to your - application. - - - StatusCallbackMethod - The HTTP method RestComm will use to make requests to the - StatusCallback URL. Either GET or POST. - - - Uri - The URI for this Client, relative to - http://localhost:port/restcomm. - - - -
-
- Supported Operations - - HTTP GET - Returns the representation of an Client resource, including - the properties above. - - - HTTP POST/PUT - Modifies a Client resource and returns the representation, - including the properties above. Below - you will find a list of - optional parameters. - - - Request Parameters - - - - - - Parameter - Description - - - - - FriendlyName - A formatted version of this phone number. - - - Password - The password used by the user agent during registration - and outbound dialing. - - - Status - The client status the possible values are 0 for disabled - and 1 for enabled. - - - VoiceUrl - The URL RestComm will request when this phone number - receives a call. - - - VoiceMethod - The HTTP method RestComm will use when requesting the - above Url. Either GET or POST. - - - VoiceFallbackUrl - The URL that RestComm will request if execution of - VoiceUrl fails for any reason. - - - VoiceFallbackMethod - The HTTP method RestComm will use when requesting the - VoiceFallbackUrl. Either GET or POST. - - - VoiceApplicationSid - If this entry contains an Sid to a voice application then - RestComm will ignore these voice URLs - and use the voice URLs - specified by the voice application. - - - StatusCallback - The URL that RestComm will request to pass status - parameters (such as the call state) to your - application. - - - StatusCallbackMethod - The HTTP method RestComm will use to make requests to the - StatusCallback URL. Either GET or POST. - - - -
- - HTTP DELETE - Deletes a Client from the user's Account. - -
-
- Client List Resource - - Client List Resource URI - /2012-04-24/Accounts/{AccountSid}/Clients - -
- Supported Operations - - HTTP GET - Returns the list representation of all the Client resources - for this Account, including the - properties above. - - - HTTP POST - Creates a new Client and returns the representation of the - resource, including - the properties above. Below you will find a - list of required and - optional parameters. - - - Request Parameters - - - - - - Parameter - Description - - - - - FriendlyName - A formatted version of this phone number. - - - Login - The name that is used inside the <Client> noun. - This is also used by - the user agent as the user name used for - registration and outbound - dialing. - - - Password - The password used by the user agent during registration - and outbound dialing. - - - Status - The client status the possible values are 0 for disabled - and 1 for enabled. - - - VoiceUrl - The URL RestComm will request when this phone number - receives a call. - - - VoiceMethod - The HTTP method RestComm will use when requesting the - above Url. Either GET or POST. - - - VoiceFallbackUrl - The URL that RestComm will request if execution of - VoiceUrl fails for any reason. - - - VoiceFallbackMethod - The HTTP method RestComm will use when requesting the - VoiceFallbackUrl. Either GET or POST. - - - VoiceApplicationSid - If this entry contains an Sid to a voice application then - RestComm will ignore these voice URLs - and use the voice URLs - specified by the voice application. - - - StatusCallback - The URL that RestComm will request to pass status - parameters (such as the call state) to your - application. - - - StatusCallbackMethod - The HTTP method RestComm will use to make requests to the - StatusCallback URL. Either GET or POST. - - - -
- - - - Create a Client - The client name will be Alice as shown below - - - - - The output of the command will be similar to the one below - - - - - Delete a Client - You must use the Client SID - - - - - - - - Change Client's Password - You must use the Client SID as shown below: - - - - - - - - Get List of available Clients - The command below shows all Clients created using the default - Admin Account - - - - - -
-
-
-
- IncomingPhoneNumbers - An IncomingPhoneNumber instance resource represents a RestComm - phone number. - - IncomingPhoneNumber Resource URI - /2012-04-24/Accounts/{AccountSid}/IncomingPhoneNumbers/{IncomingPhoneNumberSid} - - - Binding a Phone Number to an Application - When using RestComm the way to bind a phone number to an - application is be creating - a new IncomingPhoneNumber resource and - providing the VoiceUrl to your - application. - - - Resource Properties - - - - - - Property - Description - - - - - Sid - A string that uniquely identifies this incoming phone - number. - - - DateCreated - The date that this incoming phone number was created. - - - DateUpdated - The date that this incoming phone number was last updated. - - - FriendlyName - A formatted version of this phone number. - - - AccountSid - The unique id of the Account that owns this phone number. - - - PhoneNumber - The incoming phone number in E.164 format ex. +2223334444 - - - ApiVersion - Calls to this phone number will create a new RCML session - with this API version. - - - VoiceCallerIdLookup - Look up the caller's caller-ID name. Either true or false. - - - VoiceUrl - The URL RestComm will request when this phone number - receives a call. - - - VoiceMethod - The HTTP method RestComm will use when requesting the above - Url. Either GET or POST. - - - VoiceFallbackUrl - The URL that RestComm will request if execution of VoiceUrl - fails for any reason. - - - VoiceFallbackMethod - The HTTP method RestComm will use when requesting the - VoiceFallbackUrl. Either GET or POST. - - - StatusCallback - The URL that RestComm will request to pass status - parameters (such as the call state) to your - application. - - - StatusCallbackMethod - The HTTP method RestComm will use to make requests to the - StatusCallback URL. Either GET or POST. - - - SmsUrl - The URL that RestComm will request when receiving an - incoming SMS message to this number. - This may not be supported. - Please consult with your DID provider. - - - SmsMethod - The HTTP method RestComm will use when making requests to - the SmsUrl. Either GET or POST. - - - SmsFallbackUrl - The URL that RestComm will request if SmsUrl fail for any - reason. Please see SmsUrl - as this feature may not be supported. - - - SmsFallbackMethod - The HTTP method RestComm will use when making requests to - SmsFallbackUrl. Either GET or POST. - - - Uri - The URI for this incoming phone number, relative to - http://localhost:port/restcomm. - - - -
-
- Supported Operations - - HTTP GET - Returns the representation of an IncomingPhoneNumber resource, - including the properties above. - - - HTTP POST/PUT - Modifies an IncomingPhoneNumber resource and returns the - representation, including the properties above. Below - you will find - a list of optional parameters. - - - Request Parameters - - - - - - Parameter - Description - - - - - FriendlyName - A formatted version of this phone number. - - - ApiVersion - Calls to this phone number will create a new RCML session - with this API version. - - - VoiceCallerIdLookup - Look up the caller's caller-ID name. Either true or false. - - - VoiceUrl - The URL RestComm will request when this phone number - receives a call. - - - VoiceMethod - The HTTP method RestComm will use when requesting the - above Url. Either GET or POST. - - - VoiceFallbackUrl - The URL that RestComm will request if execution of - VoiceUrl fails for any reason. - - - VoiceFallbackMethod - The HTTP method RestComm will use when requesting the - VoiceFallbackUrl. Either GET or POST. - - - StatusCallback - The URL that RestComm will request to pass status - parameters (such as the call state) to your - application. - - - StatusCallbackMethod - The HTTP method RestComm will use to make requests to the - StatusCallback URL. Either GET or POST. - - - SmsUrl - The URL that RestComm will request when receiving an - incoming SMS message to this number. - This may not be supported. - Please consult with your DID provider. - - - SmsMethod - The HTTP method RestComm will use when making requests to - the SmsUrl. Either GET or POST. - - - SmsFallbackUrl - The URL that RestComm will request if SmsUrl fail for any - reason. Please see SmsUrl - as this feature may not be supported. - - - SmsFallbackMethod - The HTTP method RestComm will use when making requests to - SmsFallbackUrl. Either GET or POST. - - - -
- - HTTP DELETE - Releases an IncomingPhoneNumber from the user's Account. - -
-
- IncomingPhoneNumber List Resource - - IncomingPhoneNumber List Resource URI - /2012-04-24/Accounts/{AccountSid}/IncomingPhoneNumbers - -
- Supported Operations - - HTTP GET - Returns the list representation of all the - IncomingPhoneNumber resources for this Account, including the - properties above. - - - HTTP POST - Creates a new IncomingPhoneNumber and returns the - representation of the resource, including - the properties above. - Below you will find a list of required and - optional parameters. - - - Request Parameters - - - - - - Parameter - Description - - - - - PhoneNumber(Required) - The phone number you want to provision. - - - FriendlyName - A formatted version of this phone number. - - - ApiVersion - Calls to this phone number will create a new RCML session - with this API version. - - - VoiceCallerIdLookup - Look up the caller's caller-ID name. Either true or - false. - - - VoiceUrl - The URL RestComm will request when this phone number - receives a call. - - - VoiceMethod - The HTTP method RestComm will use when requesting the - above Url. Either GET or POST. - - - VoiceFallbackUrl - The URL that RestComm will request if execution of - VoiceUrl fails for any reason. - - - VoiceFallbackMethod - The HTTP method RestComm will use when requesting the - VoiceFallbackUrl. Either GET or POST. - - - StatusCallback - The URL that RestComm will request to pass status - parameters (such as the call state) to your - application. - - - StatusCallbackMethod - The HTTP method RestComm will use to make requests to the - StatusCallback URL. Either GET or POST. - - - SmsUrl - The URL that RestComm will request when receiving an - incoming SMS message to this number. - This may not be supported. - Please consult with your DID provider. - - - SmsMethod - The HTTP method RestComm will use when making requests to - the SmsUrl. Either GET or POST. - - - SmsFallbackUrl - The URL that RestComm will request if SmsUrl fail for any - reason. Please see SmsUrl - as this feature may not be supported. - - - SmsFallbackMethod - The HTTP method RestComm will use when making requests to - SmsFallbackUrl. Either GET or POST. - - - -
- - - Attach a phone number to an application - This one uses the default application - - - - - - - - - - Delete a phone number - You have to get the SID of the phone and use curl to delete - as follows - - - - - - - List of phone numbers - Gets all numbers created using IncomingPhoneNumbers.json - - - - - -
-
-
- - - -
- Calls - A Call represents a connection between a phone or user agent and - RestComm. This may be inbound - or outbound. The Calls list resource - represents the set of phone calls - originated and terminated from - an - account. - - Call Resource URI - /2012-04-24/Accounts/{AccountSid}/Calls/{CallSid} - - - Resource Properties - - - - - - Property - Description - - - - - Sid - A string that uniquely identifies this call. - - - ParentCallSid - A string that uniquely identifies the call that created - this leg. - - - DateCreated - The date that this call was created. - - - DateUpdated - The date that this call was last updated. - - - AccountSid - The unique id of the Account that created this call. - - - To - The phone number or identifier that will be the recipient - of this call. - - - From - The phone number or identifier that originated this call. - - - PhoneNumberSid - If the call was inbound, this is the Sid of the - IncomingPhoneNumber that received the call. - - - Status - A string representing the status of the call. Possible - values are queued, ringing, in-progress, completed, - failed, busy - and no-answer. - - - StartTime - The start time of the call. Empty if the call has not yet - been started. - - - EndTime - The end time of the call. Empty if the call has not ended.. - - - Duration - The length of the call in seconds. - - - - Price - The charge for this call, in the currency associated with - the account. Populated after the call is completed. - - - - - Direction - A string describing the direction of the call. Possible - values are inbound, outbound-api, - and outbound-dial - - - - AnsweredBy - If this call was initiated with answering machine - detection, either human or machine. Empty otherwise. - - - - ApiVersion - Displays the current API version - - - - ForwardFrom - If this call was an incoming call forwarded from another - number, the forwarding phone number (depends on carrier - supporting forwarding). Empty otherwise. - - - - - CallerName - If this call was an incoming call, the caller's name. Empty - otherwise. - - - Uri - The URI for this account, relative to - http://localhost:port/restcomm. - - - -
- - -
- Supported Operations - - HTTP GET - Returns the representation of a Call resource, including the - properties above. - -
- - -
- Call List Resource - - Call List Resource URI - /2012-04-24/Accounts/{AccountSid}/Calls - -
- -
- Supported Operations - - HTTP GET - Returns the list representation of all the Call resources for - this Account, including the - properties above. - - - HTTP POST - Makes a new Call and returns the representation of the Call - resource, including - the properties above. Below you will find a list - of required and optional parameters. - - - Request Parameters - - - - - - Parameter - Description - - - - - From(Required) - The phone number to use as the caller id. - - - To(Required) - The phone number to call. - - - Url(Required) - The fully qualified URL that should be executed when the - call connects. - - - Method - The HTTP method RestComm should use when making its - request to the above Url. Defaults to POST. - - - FallbackUrl - The URL that RestComm will request if execution of Url - fails for any reason. - - - FallbackMethod - The HTTP method that RestComm should use to request the - FallbackUrl. Must be either GET or POST. - Defaults to POST. - - - StatusCallback - A URL that RestComm will request when the call ends to - notify your app. - - - StatusCallbackMethod - The HTTP method RestComm should use when requesting the - above StatusCallback. Defaults to POST. - - - Timeout - The number of seconds that RestComm should allow the phone - to ring before - assuming there is no answer. The default is 60 - seconds. - - - -
- - - - Making a call to a SIP account - - Restcomm will make a call to any SIP account that is reachable. - It the example below, the SIP account is listening on port 5060. - When you make the call, the SIP phone on which Alice is registered - will ring and the hello-play.xml file will be played. - - - - - - - Making a call to a Restcomm client - - You must first create a RestComm client. In the example below, - the Restcomm client created is called Alice. When you make the call, - the SIP phone on which Alice is registered will ring and the - hello-play.xml file will be played. - - - - - - - - - - - - Calling a DID number - - The above example shows how to make a call to a SIP number. If you - want to make a call to a DID number, you must can connect Restcomm - to a DID provisioning service provider. The quickest way is to use - RestComm AMI for Voice Innovation - - - - - Get a list of all available calls - This will return all the available calls linked to the account - SID - - - - Working on a production server - Using filter is a good practice on a server with thousands or - millions of calls - - - - - - If the system hasn't received any calls, you will see the the - output below - - - - - -]]> - -
- - -
- REST API: Modifying Live Calls - - - Realtime call modification allows you to interrupt an - in-progress call and - terminate it or have it begin processing RCML - from a new URL. This - is useful for any application where you want to - asynchronously - change the behavior of a running call. For example: - hold music, call - queues, transferring calls, forcing hangup, etc. - - -
- HTTP POST to a Call - - Client List Resource URI - - - To redirect or terminate a live call, you make an HTTP POST - request - to an in-progress Call instance resource URI: - - /2010-04-01/Accounts/{AccountSid}/Calls/{CallSid} - -
- - - POST Parameters - - - The following parameters are available for you to POST when - modifying a - phone call: - - - - Request Parameters - - - - - - Parameter - Description - - - - - Url - A valid URL that returns TwiML. Twilio will immediately - redirect the call to the new TwiML. - - - Method - The HTTP method Twilio should use when requesting the - above URL. Defaults to POST. - - - Status - Either canceled or completed. Specifying canceled will - attempt to hangup calls that are queued or ringing but not - affect calls already in progress. Specifying completed will - attempt to hang up a call even if it's already in progress. - - - - -
- - Call in-Progress - - Note that any call which is currently ringing within a Dial - verb is - in-progress from the point of view of Restcomm, and thus you - must - use 'Status=completed' to cancel it. - - - - - - Optional Parameters - - You may POST the following parameters: - - - - - - Request Parameters - - - - - - Parameter - Description - - - - - FallbackUrl - A URL that Twilio will request if an error occurs - requesting or executing the TwiML at Url. - - - FallbackMethod - The HTTP method that Twilio should use to request the - FallbackUrl. Must be either GET or POST. Defaults to POST. - - - StatusCallback - A URL that Twilio will request when the call ends to - notify your app. - - - StatusCallbackMethod - The HTTP method Twilio should use when requesting the - above URL. Defaults to POST. - - - - -
- - -
- Modifying Live Calls - Example - - - - - - - - CAfa51b104354440b09213d04752f50271 - 2013-11-01T03:41:14.488-06:00 - 2013-11-01T03:41:14.488-06:00 - - ACae6e420f425248d6a26948c17a9e2acf - alice - +16175551212 - - ..... TRUNCATED - - -You can still redirect the current call back to the previous application - - -curl -X POST http://ACae6e420f425248d6f8c12cc7b8f8423e5c38b035249166@127.0.0.1:8080/restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Calls/CAfa51b104354440b09213d04752f50271 -d "Url=http://127.0.0.1:8080/restcomm/demos/hello-play.xml" - -The output showing the same Call ID - - - - CAfa51b104354440b09213d04752f50271 - 2013-11-01T03:41:14.488-06:00 - 2013-11-01T03:41:14.488-06:00 - - ACae6e420f425248d6a26948c17a9e2acf - alice - +16175551212 - - ..... TRUNCATED - - -You can end the call using the Status=completed command as shown below - -curl -X POST http://ACae6e420f425248d6f8c12cc7b8f8423e5c38b035249166@127.0.0.1:8080/restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Calls/CAfa51b104354440b09213d04752f50271 -d "Status=completed" - -The output showing the same Call ID - - - - CAfa51b104354440b09213d04752f50271 - 2013-11-01T03:41:14.488-06:00 - 2013-11-01T03:41:14.488-06:00 - - ACae6e420f425248d6a26948c17a9e2acf - alice - +16175551212 - - ..... TRUNCATED - -]]> - - -
- - -
- - - - -
- - List Filter - - - HTTP GET - The following GET query string parameters allow you to limit - the list returned. Note, parameters are case-sensitive: - - - - Request Parameters - - - - - - Parameter - Description - - - - - To - Only show calls to this phone number or Client identifier. - - - From - Only show calls from this phone number or Client - identifier. - - - Status - Only show calls currently in this status. May be queued, - ringing, in-progress, canceled, completed, failed, busy, or - no-answer. - - - StartTime - Only show calls that started on this date, given as - YYYY-MM-DD. Also supports inequalities, such as - StartTime=YYYY-MM-DD for calls that started at or before - midnight on a date, and StartTime=YYYY-MM-DD for calls that - started at or after midnight on a date. - - - ParentCallSid - Only show calls spawned by the call with this Sid. - - - - -
- - - - Filter using the From parameter - The example below will only return Calls made from client - Alice - - - - - - - The result will be similar to the one below - - - - - - CAc0fb839632cf444f9066876d5de741e0 - 2013-10-18T04:51:47.643-06:00 - 2013-10-18T04:51:49.174-06:00 - - ACae6e420f425248d6a26948c17a9e2acf - 1234 - alice - - completed - 2013-10-18T04:51:47.671-06:00 - 2013-10-18T04:51:49.174-06:00 - 1 - 0.00 - inbound - - 2012-04-24 - - - /2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Calls/CAc0fb839632cf444f9066876d5de741e0 - - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Calls/CAc0fb839632cf444f9066876d5de741e0/Notifications - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Calls/CAc0fb839632cf444f9066876d5de741e0/Recordings - - - - -]]> - - -
- -
- - Paging Information - - - HTTP GET - The following GET query string parameters allow you to limit - the list returned. Note, parameters are case-sensitive: - - - - - Request Parameters - - - - - - Parameter - Description - - - - - Page - The current page number. Zero-indexed, so the first page - is 0. - - - NumPages - The total number of pages. - - - PageSize - How many items are in each page - - - Total - The total number of items in the list. - - - Start - The position in the overall list of the first item in this - page. - - - End - The position in the overall list of the last item in this - page. - - - -
- - - - - - - Example - The command below will return a single item from the list of - calls using the PageSize parameter - - - - - - The result of the PageSize parameter - - - - - - - CA4049cf008d6b4277b92ab863fd4ec7c8 - 2013-10-18T04:49:45.942-06:00 - 2013-10-18T04:49:46.272-06:00 - - ACae6e420f425248d6a26948c17a9e2acf - 1235 - bob - - completed - 2013-10-18T04:49:45.994-06:00 - 2013-10-18T04:49:46.272-06:00 - 0 - 0.00 - inbound - - 2012-04-24 - - - /2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Calls/CA4049cf008d6b4277b92ab863fd4ec7c8 - - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Calls/CA4049cf008d6b4277b92ab863fd4ec7c8/Notifications - /restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a9e2acf/Calls/CA4049cf008d6b4277b92ab863fd4ec7c8/Recordings - - - - - -]]> - - - - - - - Additional Paging Information - The API returns URIs to the next, previous, first and last - pages of the returned list as shown in the table below: - - - - Request Parameters - - - - - - Parameter - Description - - - - - Uri - The URI of the current page. - - - Firstpageuri - The URI for the first page of this list. - - - Nextpageuri - The URI for the next page of this list. - - - Previouspageuri - The URI for the previous page of this list. - - - Lastpageuri - The URI for the last page of this list. - - - - -
- - - - -
- -
- - - - - - -
- SMS Messages - An SMS Message resource represents an inbound or outbound SMS - message. - - SMS Message Resource URI - /2012-04-24/Accounts/{AccountSid}/SMS/Messages/{SMSMessageSid} - - - Resource Properties - - - - - - Property - Description - - - - - Sid - A string that uniquely identifies this SMS Message. - - - DateCreated - The date that this SMS Message was created. - - - DateUpdated - The date that this SMS Message was last updated. - - - DateSent - The date that the SMS was sent or received by RestComm. - - - AccountSid - The unique id of the Account that sent or received this SMS - message. - - - From - The phone number or short code that initiated the message. - - - To - The phone number or short code that received the message. - - - Body - The text body of the SMS message. Up to 160 characters - long. - - - Status - The status of this SMS message. Possible values are queued, - sending, sent, failed, and received. - - - Direction - The direction of this SMS message. Possible values are - incoming, outbound-api, outbound-call. - - - ApiVersion - The API version RestComm used to handle the SMS message. - - - Uri - The URI for this account, relative to - http://localhost:port/restcomm. - - - -
-
- Supported Operations - - HTTP GET - Returns the representation of an SMS Message resource, - including the properties above. - -
-
- SMS Message List Resource - - SMS Message List Resource URI - /2012-04-24/Accounts/{AccountSid}/SMS/Messages - -
- Supported Operations - - HTTP GET - Returns the list representation of all the Call resources for - this Account, including the - properties above. - - - HTTP POST - Sends a new SMS Message and returns the representation of the - SMS Message resource, including - the properties above. Below you - will find a list of required and - optional parameters. - - - Request Parameters - - - - - - Parameter - Description - - - - - From(Required) - A phone number that is enabled for SMS. - - - To(Required) - The destination phone number in E.164 format. - - - Body(Required) - The text of the message you want to send, limited to 160 - characters. - - - Custom headers X- (optional) - Optionally you can provide as many custom headers as you - wish. The custom headers must start with X-, - for example "X-MyCustom-Header=My custom header value" - - - -
- - - - Using custom headers - - These additional headers will be part of the SIP MESSAGE that Restcomm will create and dispatch to the SMS Aggregator. - - - - - Using SMS and making DID calls - - You need to configure Restcomm to send SMS messages and DID phone - calls to a Service Provider for provisioning. In the - restcomm.xml - file, the outbound-proxy-uri and the SMS outbound-endpoint must - point to the Service Provider IP address. You may also decide to - use - Restcomm AMI. - - - - - - Send SMS Messages - - - Note the encoding used - %2B13216549878 - instead of the - +13216549878 - The + sign is encoded to to send SMS from the command line. - - - - - "From" DID number - - The "From" number should be the DID SMS enabled number from - VoIP - Innovations. - - - From the bash terminal, you can run the command below: - - - - - To send the same SMS but this time also provide some additional headers: - - - - - - Get list of SMS Messages - - This will display list of message sent - From the bash terminal, you can run the command below: - - - - -
-
-
-
- Recordings - Recordings are generated when you use the <Record> verb. - Those recordings are hosted with RestComm for you to retrieve. - The - Recordings list resource represents the set of an account's - recordings. - - Recording Resource URI - /2012-04-24/Accounts/{AccountSid}/Recordings/{RecordingSid} - - To download the audio file just append .wav after the - RecordingSid. - - Resource Properties - - - - - - Property - Description - - - - - Sid - A string that uniquely identifies this recording. - - - DateCreated - The date that this recording was created. - - - DateUpdated - The date that this recording was last updated. - - - AccountSid - The unique id of the Account that created this recording. - - - CallSid - The unique id of the call during which the recording was - made. - - - Duration - The length of the recording, in seconds. - - - ApiVersion - The API version in use during the recording. - - - Uri - The URI for this account, relative to - http://localhost:port/restcomm. - - - -
-
- Supported Operations - - HTTP GET - Returns the representation of a Recording resource, including - the properties above. - - - HTTP DELETE - Removes the recording from the account. - -
-
- Recording List Resource - - Recording List Resource URI - /2012-04-24/Accounts/{AccountSid}/Recordings - -
- Supported Operations - - HTTP GET - Returns the list representation of all the Recording - resources for this Account, including the - properties above. - - - - - How to Record a Message - - Go to the Advanced Chapter under - - section to learn how to record a message. - - - - - - Get List of Recordings - - The list of recorded wav files can be found in the directory - $RESTCOMM_HOME/standalone/deployments/restcomm.war/recordings/ - - From the bash terminal, you can run the command below: - - - - - -
-
-
-
- Transcriptions - A Transcription resource represents a transcription of a - recording. A transcription is a text version of - a recording produced - using automatic speech recognition. - - Transcription Resource URI - /2012-04-24/Accounts/{AccountSid}/Transcriptions/{TranscriptionSid} - - - Resource Properties - - - - - - Property - Description - - - - - Sid - A string that uniquely identifies this transcription. - - - DateCreated - The date that this transcription was created. - - - DateUpdated - The date that this transcription was last updated. - - - AccountSid - The unique id of the Account that created this - transcription. - - - Status - A string representing the status of the transcription. - Possible values are in-progress, completed, - and failed. - - - RecordingSid - The unique id of the Recording this Transcription was made - of. - - - Duration - The duration of the transcribed audio, in seconds. - - - TranscriptionText - The text content of the transcription. - - - Uri - The URI for this account, relative to - http://localhost:port/restcomm. - - - -
-
- Supported Operations - - HTTP GET - Returns the representation of a Transcription resource, - including the properties above. - - - HTTP DELETE - Removes the Transcription from the account. - -
-
- Transcription List Resource - - Transcription List Resource URI - /2012-04-24/Accounts/{AccountSid}/Transcriptions - -
- Supported Operations - - HTTP GET - Returns the list representation of all the Transcription - resources for this Account, including the - properties above. - -
-
-
-
- Notifications - A Notification resource represents a single log entry made by - RestComm while handling your calls or your use of the - Restful APIs. It - is very useful for debugging purposes. The Notifications - list resource - represents the set of notifications - generated for an account. - - Notification Resource URI - /2012-04-24/Accounts/{AccountSid}/Notifications/{NotificationSid} - - - Resource Properties - - - - - - Property - Description - - - - - Sid - A string that uniquely identifies this transcription. - - - DateCreated - The date that this transcription was created. - - - DateUpdated - The date that this transcription was last updated. - - - AccountSid - The unique id of the Account that created this - transcription. - - - CallSid - CallSid is the unique id of the call during which the - notification was generated. Empty if the - notification was - generated by the Restful APIs without regard to a specific - phone - call. - - - ApiVersion - The RestComm API version in use when this notification was - generated. May be empty for events - that don't have a specific API - version. - - - Log - An integer log level corresponding to the type of - notification: 0 is ERROR, 1 is WARNING. - - - ErrorCode - A unique error code for the error condition. You can lookup - errors, in our Error Dictionary. - - - MoreInfo - A URL for more information about the error condition. The - URL is a page in our Error Dictionary. - - - MessageText - The text for the notification. - - - MessageDate - The date the notification was actually generated - - - RequestUrl - The URL of the resource that caused the notification to be - generated. - - - RequestMethod - The HTTP method in use for the request that caused the - notification to be generated. - - - RequestVariables - The HTTP GET or POST variables that RestComm generated and - sent to your server. Also, if the - notification was generated by - the Restful APIs, this field will include any - HTTP POST or PUT - variables - you sent. - - - ResponseHeaders - The HTTP headers returned by your server. - - - ResponseBody - The HTTP body returned by your server. - - - Uri - The URI for this account, relative to - http://localhost:port/restcomm. - - - -
-
- Supported Operations - - HTTP GET - Returns the representation of a Notification resource, - including the properties above. - -
-
- Notification List Resource - - Notification List Resource URI - /2012-04-24/Accounts/{AccountSid}/Notifications - -
- Supported Operations - - HTTP GET - Returns the list representation of all the Notification - resources for this Account, including the - properties above. - -
-
-
-
- OutboundProxy - Using Outbound proxy endpoint you can get the details of Primary - and Backup outbound proxy, - get the current active outbound proxy and - also switch outbound proxy. - - OutboundProxy Resource URI - /2012-04-24//Accounts/{accountSid}/OutboundProxy - - -
- Supported Operations - - HTTP GET - Returns the list of outbound proxies. - - - HTTP GET - /2012-04-24//Accounts/{accountSid}/OutboundProxy/switchProxy - Switch the outbound proxy and returns the proxy in use - - - HTTP GET - /2012-04-24//Accounts/{accountSid}/OutboundProxy/getActiveProxy - Returns the currently active outbound proxy - -
-
- - - - - -
- OutboundProxy - Using Outbound proxy endpoint you can get the details of Primary - and Backup outbound proxy, - get the current active outbound proxy and - also switch outbound proxy. - - OutboundProxy Resource URI - /2012-04-24//Accounts/{accountSid}/OutboundProxy - - -
- Supported Operations - - HTTP GET - Returns the list of outbound proxies. - - - HTTP GET - /2012-04-24//Accounts/{accountSid}/OutboundProxy/switchProxy - Switch the outbound proxy and returns the proxy in use - - - HTTP GET - /2012-04-24//Accounts/{accountSid}/OutboundProxy/getActiveProxy - Returns the currently active outbound proxy - -
-
- -
- UssdPush - A UssdPush resource represents a message sent from Restcomm to a - USSD gateway. - - UssdPush Resource URI - /2012-04-24/Accounts/{AccountSid}/UssdPush - -
- Supported Operations - - HTTP POST - Sends a Ussd message to the configured USSD Gataeway - - -
- - - Example of UssdPush - - - - - - IP_USSD_GW:PORT_NUMBER - - - - - - - - See below a curl example for the USSD Push: - - curl -X POST -H "application/json" http://ACae6e420f425248d6a26948c17a9e2acf:YOUR_PWD@RESTCOMM_IP_ADDRESS:8080/restcomm/2012-04-24/Accounts/ACae6e420f425248d6a26948c17a-d "From=Restcomm" -d "To=123" -d "Url=http://RESTCOMM_IP_ADDRESS:8080/restcomm-rvd/services/apps/YOUR_USSD_APP/controller" - ]]> - - - - -
- - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Services_for_SIP_Servlets.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Services_for_SIP_Servlets.xml deleted file mode 100644 index c89b0d7600..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-Services_for_SIP_Servlets.xml +++ /dev/null @@ -1,2140 +0,0 @@ - - -%BOOK_ENTITIES; -]> - -
- Operating the Example Applications -
- The Location Service - The &PLATFORM_NAME; - Location Service contains a list of mappings of - request URIs to destination addresses. When the Location Service receives - a request, it performs a lookup on that mapping and proxies the - request - simultaneously to the destination address (or addresses) associated with - that URI. - - - The Location Service Mappings Cannot Currently Be Configured - - The Location Service currently performs a lookup on a - hard-coded - list of addresses. This model is evolving toward the eventual use of a - database. - - - Regardless of whether you are using the JBoss Application Server - or - the Tomcat Servlet Container as the Servlets Server, the application, - container and Location Service perform the following steps: - - - - - A user—let us call her Alice—makes a call to - sip:receiver@sip-servlets.com - . The - INVITE - is received by the servlet container, which - then starts the Location Service. - - - - - The Location Service, using non-SIP means, determines that the - callee (i.e. the receiver) is registered at two locations, - identified - by the two SIP URIs, - sip:receiver@127.0.0.1:5090 - and - sip:receiver@127.0.0.1:6090 - . - - - - The Location Service proxies to those two destinations in - parallel, without record-routing, and without making use of - supervised - mode. - - - - - One of the destinations returns a - 200 OK - status code; the second proxy is then canceled. - - - - - The - 200 OK - is forwarded to Alice, and call - setup is completed as usual. - - - - Here is the current list of hard-coded contacts and their - location - URIs: - - - sip:receiver@sip-servlets.com - - - sip:receiver@127.0.0.1:5090 - - - - - sip:receiver@127.0.0.1:6090 - - - - -
- The Location Service: Installing, Configuring and Running - -   - - Pre-Install Requirements and Prerequisites - The following requirements must be met before installation can - begin. - - - - Software Prerequisites - - Either - an &SHORT_PLATFORM_NAME; - for JBoss or an &SHORT_PLATFORM_NAME; - for Tomcat Installation - - - The - Location Service requires either an &SHORT_PLATFORM_NAME; - for JBoss or an - &SHORT_PLATFORM_NAME; - for Tomcat binary installation. - - You - can find detailed instructions on installing &SHORT_PLATFORM_NAME; - for - JBoss here: - - . - - You - can find detailed instructions on installing &SHORT_PLATFORM_NAME; - for - Tomcat here: - - . - - - - - - Downloading - - The Location Service is comprised of two archive files, a Web - Archive (WAR) and a Default Application Router (DAR) configuration - file, which you need to add to your installed SIP Servlets Server. - For - more information about WAR files, refer to the - JBoss Application Server Administration and Development Guide - - . For - more information about DAR files, refer to the - JSR 289 spec, Appendix C - . - - - Download - the Location Service's WAR file from here: - - . - - Download - the Location Service's DAR file from here: - - . - - - Installing - - Both the - location-service-1.4.war - WAR file - and the - locationservice-dar.properties - DAR file - that you downloaded should be placed into different directories in - your SIP Servlet Server installation hierarchy. Which directory - depends on whether you are using the Location Service with &SHORT_PLATFORM_NAME; - for - JBoss or with &SHORT_PLATFORM_NAME; - for Tomcat: - - - - - &SHORT_PLATFORM_NAME; - for JBoss - - - - Place - location-service-1.4.war - into the - - JBOSS_HOME - /server/default/deploy/ - - directory, and - locationservice-dar.properties - into the - - JBOSS_HOME - /server/default/conf/dars/ - - directory. - - - - - &SHORT_PLATFORM_NAME; - for Tomcat - - - - Place - location-service-1.4.war - into the - - CATALINA_HOME - /webapps/ - - directory, and - locationservice-dar.properties - into the - - CATALINA_HOME - /conf/dars/ - - directory. - - - - - - Configuring - - The - darConfigurationFileLocation - attribute of - the - Service - element must be set to the value - conf/dars/locationservice-dar.properties - . The - instructions are given below by SIP Servlets Server type: - - - - - &SHORT_PLATFORM_NAME; - for JBoss - - - - Open the - - JBOSS_HOME - /server/default/deploy/jboss-web.deployer/server.xml - - configuration file and find the - Service - element. Add an attribute to it called - darConfigurationFileLocation - , and set it to - conf/dars/locationservice-dar.properties - - : - - - Editing &SHORT_PLATFORM_NAME; - for JBoss's server.xml for the Location Service - - <Service - name="jboss.web" - className="org.mobicents.servlet.sip.startup.SipStandardService" - sipApplicationDispatcherClassName="org.mobicents.servlet.sip.core.SipApplicationDispatcherImpl" - darConfigurationFileLocation="conf/dars/locationservice-dar.properties" - sipStackPropertiesFile="conf/mss-sip-stack.properties"> - - - Make sure that the configuration file only contains one - darConfigurationFileLocation - attribute: your - new one. - - - - - &SHORT_PLATFORM_NAME; - for Tomcat - - - - Open the - - CATALINA_HOME - /conf/server.xml - - configuration file and find the - Service - element. Add an attribute to it called - darConfigurationFileLocation - , and set it to - conf/dars/locationservice-dar.properties - : - - - Editing &SHORT_PLATFORM_NAME; - for Tomcat's server.xml for the Location Service - - <Service - name="Sip-Servlets" - className="org.mobicents.servlet.sip.startup.SipStandardService" - sipApplicationDispatcherClassName="org.mobicents.servlet.sip.core.SipApplicationDispatcherImpl" - darConfigurationFileLocation="conf/dars/locationservice-dar.properties" - sipStackPropertiesFile="conf/mss-sip-stack.properties"> - - - Make sure that the configuration file only contains one - darConfigurationFileLocation - attribute: your - new one. - - - - - - Running - - Once the WAR and DAR files have been placed in the right - directories, and the JBoss Application Server or Tomcat Servlet - Container knows where to find them (which you specified in a - server.xml - file), then you should go ahead and - run the SIP Servlets Server. - - - - To learn how to run the SIP Servlets-enabled JBoss Application - Server, refer to - - . - - - To learn how to run the SIP Servlets-enabled Tomcat Container, - refer to - - . - - - Testing - The following procedure shows how to test the Location - Service. - - - - - <step> - <para> - Start two SIP soft-phones. The first phone should be set up as - <userinput>sip:receiver@sip-servlets.com</userinput> - at the IP address - <userinput>127.0.0.1</userinput> - on port - <userinput>5090</userinput> - . The second phone can be set up in any - way you like. Note that the SIP phones do not have to be - registered. - </para> - </step> - <step> - <para> - Using the second phone, make a call to - <literal>sip:receiver@sip-servlets.com</literal> - . If the Location - Service has been set up correctly and is running, the first phone—as - the receiver or callee—should now be ringing. - </para> - </step> - </procedure> - <!--<formalpara id="sfss-binary-Location_Service-Using"> <title>Using -   --> - - Stopping - - To learn how to stop the SIP Servlets-enabled JBoss Application - Server, refer to - - . - - - - To learn how to stop the SIP Servlets-enabled Tomcat Container, - refer to - - . - - - Uninstalling - - Unless disk space is at a premium, there is usually no need to - uninstall the Location Service. However, if you will not be using - it - again, you may want to unset or reset the - darConfigurationFileLocation - attribute of the - Service - element, which you set in the - server.xml - configuration file in - - . - - - - You may also wish to delete the WAR and DAR files for the Location - Service, which you installed in - - . - -
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- The Diameter Event-Changing Service - The Diameter Event-Changing Service is based on the Location - Service, which performs call-charging at a fixed rate. Upon the - initiation - of a call, a debit of €10.00 occurs. In the cases of a call being - rejected - or the caller disconnecting (hanging up) before an answer is received, - the - caller's account is refunded. - Note - that an &SHORT_PLATFORM_NAME; - for JBoss installation is required to run this - example; it will not work with &SHORT_PLATFORM_NAME; - for Tomcat. - - Provided here is a step-by-step description of the procedure as - performed by the application and container: - - - Diameter Event-Changing Service Step-By-Step - - - - A user, Alice, makes a call to - sip:receiver@sip-servlets.com - . The - INVITE - is received by the servlet container, which - sends a request to debit Alice's account to the Charging Server. - The - servlet container then invokes the location service. - - - - - - The Location Service determines, without using the SIP protocol - itself, where the callee—or receiver—is registered. The callee may - be - registered at two locations identified by two SIP URIs: - sip:receiver@127.0.0.1:5090 - and - sip:receiver@127.0.0.1:6090 - . - - - - - The Location Service proxies to those two destinations - simultaneously, without record-routing and without using the - supervised - mode. - - - - - - One of the destinations returns - 200 (OK) - , and - so the container cancels the other. - - - - - - The - 200 (OK) - is forwarded upstream to Alice, - and the call setup is carried out as usual. - - - - - If none of the registered destinations accepts the - call, a Diameter Accounting-Request for refund is sent to the Diameter - Charging Server in order to debit the already-credited €10.00 - - - - -
- Diameter Event-Changing Service: Installing, Configuring and - Running - Preparing - your &SHORT_PLATFORM_NAME; - for JBoss server to run the Diameter - Event-Changing example requires downloading a WAR archive, a DAR - archive, the Ericsson Charging Emulator, setting an attribute in - JBoss's - server.xml - configuration file, and then running - JBoss AS. Detailed instructions follow. - - - Pre-Install Requirements and Prerequisites - The following requirements must be met before installation can - begin. - - - - Software Prerequisites - - One &SHORT_PLATFORM_NAME; - for JBoss Installation - - - Before - proceeding, you should follow the instructions for - installing, configuring, running and testing &SHORT_PLATFORM_NAME; - for JBoss from - the binary distribution. - - - - - - Downloading - The following procedure describes how to download the required - files. - - - - - First, download the Web Application Archive - ( - WAR - ) file corresponding to this example, the - current version of which is named - diameter-event-charging-1.0.war - , from - - . - - - - - Secondly, download the corresponding Disk Archive - ( - DAR - ) configuration file here: - - . - - - - - Finally, you will need to download the Ericsson Charging - Emulator, version 1.0, from - - . - - - - - Installing - The following procedure describes how to install the - downloaded files. - - - - - Place the - diameter-event-charging-1.0.war - WAR archive into the - $JBOSS_HOME/server/<profile>/deploy - directory. - - - - - Place the - diametereventcharging-dar.properties - DAR file - in your - - $JBOSS_HOME/server/<profile>/conf/dars - - directory. - - - - - Finally, open the terminal, move into the directory to which - you downloaded the Ericsson Charging SDK (for the sake of this - example, we will call this directory - charging_sdk), - and then unzip the downloaded - zip file (you can use Java's - - jar - - - command for this: - - ~]$ cd charging_sdk -charging_sdk]$ jar -xvf ChargingSDK-1_0_D31E.zip - Alternatively, - you can use Linux's - unzip - command to do the dirty work: - - charging_sdk]$ unzip ChargingSDK-1_0_D31E.zip - - - - Configuring - - To configure the server for the Event-Changing example, simply - open the - server.xml - configuration file in your - server's - - $JBOSS_HOME/server/<profile>/deploy/jboss-web.deployer/ - - directory, and edit the value of the - darConfigurationFileLocation - attribute of the - Service - element so that it is - conf/dars/mobicents-dar.properties - . - - - - Editing the darConfigurationFileLocation Attribute of the - Service Tag - ... - -<Service name="jboss.web" - className="org.mobicents.servlet.sip.startup.SipStandardService" - sipApplicationDispatcherClassName="org.mobicents.servlet.sip.core.SipApplicationDispatcherImpl" - sipApplicationRouterClassName="org.mobicents.servlet.sip.router.DefaultApplicationRouter" - darConfigurationFileLocation="conf/dars/mobicents-dar.properties" - sipStackPropertiesFile="conf/mss-sip-stack.properties"> - -... - - - - - Running - The following procedure describes how to run the Diameter - Event-Changing Service. - - - Diameter Event-Changing Service - - First, - you should run your &SHORT_PLATFORM_NAME; - for JBoss server. For - instructions on doing so, refer to - - . - - - - - Then, run the Ericsson Charging Emulator. Open a terminal, - change the working directory to the location of the unzipped - Charging Emulator files (in - ChargingSDK-1_0_D31E - or a similarly-named - directory), and run it with the - - java -jar - PPSDiamEmul.jar - - command: - - ~]$ java -jar PPSDiamEmul.jar - - - - Using - Using the Event-Changing service means, firstly, inserting - some - parameters into the Charging Emulator, and then, by using two SIP - (soft)phones, calling one with the other. The following sequential - instructions show you how. - - - - SIP (Soft)Phone? Which? - The &PLATFORM_NAME; - team recommends one of the following SIP phones, - and has found that they work well: the 3CX Phone, the SJ Phone or the - WengoPhone. - - - - Using the Diameter Event-Changing Service - - Configure the Ericsson SDK Charging Emulator - - Once you have started the Charging Emulator, you should - configure it exactly as portrayed in - - . - -
- Configuring the Charging Emulator - - - - - -
- - - - Set the - Peer ID - to: - aaa://127.0.0.1:21812 - - - - - Set the - Realm - to: - mobicents.org - - - - - Set the - Host IP - to: - 127.0.0.1 - - - - - - - Start two SIP (soft)phones. You should set the first phone up - with the following parameters: - sip:receiver@sip-servlets - on IP address - 127.0.0.1 - on port - 5090 - . The other phone can be set up any way - you like. - - - - - Before making a call, open the - - Config - Options - - dialog window, as shown in the image. - -
- Configuring Accounts in the Charging Emulator - - - - - -
- - - In the - Account Configuration - window of - the Charging Emulator, you can see the user's balances. Select a - user to watch the balance. You can also stretch the window - lengthwise to view the user's transaction history. - - - - - Time to call! From the second, - any-configuration - phone, make a call to - sip:receiver@sip-servlets.com - . Upon doing so, the - other phone should ring or signal that it is being contacted - . - - - - - You should be able to see a request—immediately following the - invite and before the other party (i.e. you) accepts or rejects - the - call—sent to the Charging Emulator. That is when the debit of the - user's account is made. In the case that the call is - rejected, or - the caller gives up, a second, new Diameter request is sent to - refund the initial amount charged by the call. On the other hand, if - the call is accepted, nothing else related to Diameter happens, and - no second request takes place. - - Please note that this is not the correct way to do - charging, as Diameter provides other means, such as unit - reservation. However, for the purpose of a demonstration it is - sufficient to show the debit and follow-up credit working. Also, - this is a fixed-price call, regardless of the duration. Charging - can, of course, be configured so that it is time-based. - - - - -
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- The Call-Blocking Service - The &PLATFORM_NAME; - Call-Blocking Service, upon receiving an - INVITE - request, checks to see whether the sender's - address is a blocked contact. If so, it returns a - FORBIDDEN - reply; otherwise, call setup proceeds as - normal. - - - Blocked Contacts Cannot Currently Be Configured - Blocked contacts are currently hard-coded addresses. This model - is - evolving towards the eventual use of a database. - - - Here is the current hard-coded list of blocked contacts: - - - - sip:blocked-sender@sip-servlets.com - - - - - sip:blocked-sender@127.0.0.1 - - - - -
- The Call-Blocking Service: Installing, Configuring and Running - -   - - Pre-Install Requirements and Prerequisites - The following requirements must be met before installation can - begin. - - - - Software Prerequisites - - Either - an &SHORT_PLATFORM_NAME; - for JBoss or an &SHORT_PLATFORM_NAME; - for Tomcat Installation - - - The - Call-Blocking Service requires either an &SHORT_PLATFORM_NAME; - for JBoss - or an &SHORT_PLATFORM_NAME; - for Tomcat binary installation. - - You - can find detailed instructions on installing &SHORT_PLATFORM_NAME; - for - JBoss here: - - . - - You - can find detailed instructions on installing &SHORT_PLATFORM_NAME; - for - Tomcat here: - - . - - - - - - Downloading - - The Call-Blocking Service is comprised of two archive files, a - Web Archive (WAR) and a Default Application Router (DAR) configuration - file, which you need to add to your installed SIP Servlets Server. - For - more information about WAR files, refer to the - JBoss Application Server Administration and Development Guide - - . For - more information about DAR files, refer to the - JSR 289 spec, Appendix C - . - - - Download - the Call-Blocking Service's WAR file from here: - - . - - Download - the Call-Blocking Service's DAR file from here: - - . - - - Installing - - Both the - call-blocking-1.4.war - WAR file and - the - call-blocking-servlet-dar.properties - DAR file - that you downloaded should be placed into different directories in - your SIP Servlet Server installation hierarchy. Which directory depends - on whether you are using the Call-Blocking Service with &SHORT_PLATFORM_NAME; - for JBoss - or with &SHORT_PLATFORM_NAME; - for Tomcat: - - - - - &SHORT_PLATFORM_NAME; - for JBoss - - - - Place - call-blocking-1.4.war - into the - - JBOSS_HOME - /server/default/deploy/ - - directory, and - call-blocking-servlet-dar.properties - into the - - JBOSS_HOME - /server/default/conf/dars/ - - directory. - - - - - &SHORT_PLATFORM_NAME; - for Tomcat - - - - Place - call-blocking-servlet-dar.properties - into the - - CATALINA_HOME - /webapps/ - - directory, and - call-blocking-servlet-dar.properties - into the - - CATALINA_HOME - /conf/dars/ - - directory. - - - - - - Configuring - - The - darConfigurationFileLocation - attribute of - the - Service - element must be set to the value - conf/dars/call-blocking-servlet-dar.properties - - . - The instructions for doing so are given below by SIP Servlets Server - type: - - - - - &SHORT_PLATFORM_NAME; - for JBoss - - - - Open the - - JBOSS_HOME - /server/default/deploy/jboss-web.deployer/server.xml - - configuration file and find the - Service - element. Add an attribute to it called - darConfigurationFileLocation - , and set it to - conf/dars/call-blocking-servlet-dar.properties - - : - - - Editing &SHORT_PLATFORM_NAME; - for JBoss's server.xml for the Call-Blocking Service - - <Service - name="jboss.web" - className="org.mobicents.servlet.sip.startup.SipStandardService" - sipApplicationDispatcherClassName="org.mobicents.servlet.sip.core.SipApplicationDispatcherImpl" - darConfigurationFileLocation="conf/dars/call-blocking-servlet-dar.properties" - sipStackPropertiesFile="conf/mss-sip-stack.properties"> - - - Make sure that the configuration file only contains one - darConfigurationFileLocation - attribute: your - new one. - - - - - &SHORT_PLATFORM_NAME; - for Tomcat - - - - Open the - - CATALINA_HOME - /conf/server.xml - - configuration file and find the - Service - element. Add an attribute to it called - darConfigurationFileLocation - , and set it to - conf/dars/call-blocking-servlet-dar.properties - - : - - - Editing &SHORT_PLATFORM_NAME; - for Tomcat's server.xml for the Call-Blocking Service - - <Service - name="Sip-Servlets" - className="org.mobicents.servlet.sip.startup.SipStandardService" - sipApplicationDispatcherClassName="org.mobicents.servlet.sip.core.SipApplicationDispatcherImpl" - darConfigurationFileLocation="conf/dars/call-blocking-servlet-dar.properties" - sipStackPropertiesFile="conf/mss-sip-stack.properties"> - - - Make sure that the configuration file only contains one - darConfigurationFileLocation - attribute: your - new one. - - - - - - Running - - Once the WAR and DAR files have been placed in the right - directories, and the JBoss Application Server or Tomcat Servlet - Container knows where to find them (which you specified in a - server.xml - file), then you should go ahead and - run the SIP Servlets Server. - - - - To learn how to run the SIP Servlets-enabled JBoss Application - Server, refer to - - . - - - To learn how to run the SIP Servlets-enabled Tomcat Container, - refer to - - . - - - Testing - The following procedure shows how to test the Call-Blocking - Service. - - - - Testing the Call Blocking Service - - - Start a SIP softphone of your choice. The account name should - be - blocked-sender - . The - From Header - should list one of the following addresses: - sip:blocked-sender@sip-servlets.com - or - sip:blocked-sender@127.0.0.1 - . The SIP - softphone does not need to be registered. - - - - - Make a call to any address, and you should receive a - FORBIDDEN - response. - - - - - - Stopping - - To learn how to stop the SIP Servlets-enabled JBoss Application - Server, refer to - - . - - - - To learn how to stop the SIP Servlets-enabled Tomcat Container, - refer to - - . - - - Uninstalling - - Unless disk space is at a premium, there is usually no need to - uninstall the Call-Blocking Service. However, if you will not be - using - it again, you may want to unset or reset the - darConfigurationFileLocation - attribute of the - Service - element, which you set in the - server.xml - configuration file in - - . - - - - You may also wish to delete the WAR and DAR files for the - Call-Blocking Service, which you installed in - - . - -
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- The Call-Forwarding Service - The &PLATFORM_NAME; - Call-Forwarding Service, upon receiving an - INVITE - request, checks to see whether the sender's - address is among those in a list of addresses which need to be forwarded. - If so, then the Call-Forwarding Service acts as a Back-to-Back User - Agent - (B2BUA), and creates a new call leg to the destination. When the response - is received from the new call leg, it sends it an acknowledgment - ( - ACK - ) and then responds to the original caller. If, on - the other hand, the server does not receive an - ACK - , - then it tears down the new call leg with a - BYE - . Once - the - BYE - is received, then it answers - OK - directly and sends the - BYE - to - the new call leg. - - - Contacts to Forward Cannot Currently Be Configured - Contacts to forward are currently hard-coded addresses. This - model - is evolving toward the eventual use of a database. - - - Here is the current hard-coded list of contacts to forward: - - - - - sip:receiver@sip-servlets.com - - - - - sip:receiver@127.0.0.1 - - - - -
- The Call-Forwarding Service: Installing, Configuring and - Running -   - - Pre-Install Requirements and Prerequisites - The following requirements must be met before installation can - begin. - - - - Software Prerequisites - - Either - an &SHORT_PLATFORM_NAME; - for JBoss or an &SHORT_PLATFORM_NAME; - for Tomcat Installation - - - The - Call-Forwarding Service requires either an &SHORT_PLATFORM_NAME; - for JBoss - or an &SHORT_PLATFORM_NAME; - for Tomcat binary installation. - - You - can find detailed instructions on installing &SHORT_PLATFORM_NAME; - for - JBoss here: - - . - - You - can find detailed instructions on installing &SHORT_PLATFORM_NAME; - for - Tomcat here: - - . - - - - - - Downloading - - The Call-Forwarding Service is comprised of two archive files, a - Web Archive (WAR) and a Data Archive (DAR), which you need to add - to - your installed SIP Servlets Server. For more information about WAR and - DAR files, refer to the - JBoss Application Server Administration and Development Guide - - . - - - Download - the Call-Forwarding Service's WAR file from here: - - . - - Download - the Call-Forwarding Service's DAR file from here: - - . - - - Installing - - Both the - call-forwarding-1.4.war - WAR file - and the - call-forwarding-servlet-dar.properties - DAR file that you downloaded should be placed into different - directories in your SIP Servlet Server installation hierarchy. - Which - directory depends on whether you are using the Call-Forwarding Service - with &SHORT_PLATFORM_NAME; - for JBoss or with &SHORT_PLATFORM_NAME; - for Tomcat: - - - - - &SHORT_PLATFORM_NAME; - for JBoss - - - - Place - call-forwarding-1.4.war - into the - - JBOSS_HOME - /server/default/deploy/ - - directory, and - call-forwarding-servlet-dar.properties - into - the - - JBOSS_HOME - /server/default/conf/dars/ - - directory. - - - - - &SHORT_PLATFORM_NAME; - for Tomcat - - - - Place - call-forwarding-1.4.war - into the - - CATALINA_HOME - /webapps/ - - directory, and - call-forwarding-servlet-dar.properties - into - the - - CATALINA_HOME - /conf/dars/ - - directory. - - - - - - Configuring - - The - darConfigurationFileLocation - attribute of - the - Service - element must be set to the value - conf/dars/call-forwarding-b2bua-servlet-dar.properties - - . - The instructions for doing so are given below by SIP Servlets Server - type: - - - - - &SHORT_PLATFORM_NAME; - for JBoss - - - - Open the - - JBOSS_HOME - /server/default/deploy/jboss-web.deployer/server.xml - - configuration file and find the - Service - element. Add an attribute to it called - darConfigurationFileLocation - , and set it to - conf/dars/call-forwarding-b2bua-servlet-dar.properties - - : - - - Editing &SHORT_PLATFORM_NAME; - for JBoss's server.xml for the Call-Forwarding Service - - <Service - name="jboss.web" - className="org.mobicents.servlet.sip.startup.SipStandardService" - sipApplicationDispatcherClassName="org.mobicents.servlet.sip.core.SipApplicationDispatcherImpl" - darConfigurationFileLocation="conf/dars/call-forwarding-b2bua-servlet-dar.properties" - sipStackPropertiesFile="conf/mss-sip-stack.properties"> - - - Make sure that the configuration file only contains one - darConfigurationFileLocation - attribute: your - new one. - - - - - &SHORT_PLATFORM_NAME; - for Tomcat - - - - Open the - - CATALINA_HOME - /conf/server.xml - - configuration file and find the - Service - element. Add an attribute to it called - darConfigurationFileLocation - , and set it to - conf/dars/call-forwarding-b2bua-servlet-dar.properties - - : - - - Editing &SHORT_PLATFORM_NAME; - for Tomcat's server.xml for the Call-Forwarding Service - - <Service - name="Sip-Servlets" - className="org.mobicents.servlet.sip.startup.SipStandardService" - sipApplicationDispatcherClassName="org.mobicents.servlet.sip.core.SipApplicationDispatcherImpl" - darConfigurationFileLocation="conf/dars/call-forwarding-b2bua-servlet-dar.properties" - sipStackPropertiesFile="conf/mss-sip-stack.properties"> - - - Make sure that the configuration file only contains one - darConfigurationFileLocation - attribute: your - new one. - - - - - - Running - - Once the WAR and DAR files have been placed in the right - directories, and the JBoss Application Server or Tomcat Servlet - Container knows where to find them (which you specified in a - server.xml - file), then you should go ahead and - run the SIP Servlets Server. - - - - To learn how to run the SIP Servlets-enabled JBoss Application - Server, refer to - - . - - - To learn how to run the SIP Servlets-enabled Tomcat Container, - refer to - - . - - - Testing - The following procedure shows how to test the Call-Forwarding - Service. - - - - - <step> - <para>Start two SIP soft-phones of your choice. Set the account - settings of the first SIP softphone to: - </para> - <itemizedlist> - <listitem> - <para> - Account name: - <userinput>forward-receiver</userinput> - </para> - </listitem> - <listitem> - <para> - IP address: - <userinput>127.0.0.1</userinput> - </para> - </listitem> - <listitem> - <para> - Port: - <userinput>5090</userinput> - </para> - </listitem> - </itemizedlist> - <para>Neither of the SIP soft-phones needs to be registered.</para> - </step> - <step> - <para> - From the second phone, make a call to - <literal>sip:receiver@sip-servlets.com</literal>. - The first phone, - "forward-receiver", should now be ringing. - </para> - </step> - </procedure> - <!--<formalpara id="sfss-binary-Call-Forwarding_Service-Using"> <title>Using -   --> - - Stopping - - To learn how to stop the SIP Servlets-enabled JBoss Application - Server, refer to - - . - - - - To learn how to stop the SIP Servlets-enabled Tomcat Container, - refer to - - . - - - Uninstalling - - Unless disk space is at a premium, there is usually no need to - uninstall the Call-Forwarding Service. However, if you will not be - using it again, you may want to unset or reset the - darConfigurationFileLocation - attribute of the - Service - element, which you set in the - server.xml - configuration file in - - . - - - - You may also wish to delete the WAR and DAR files for the - Call-Forwarding Service, which you installed in - - . - -
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- The Call-Controller Service - The Call-Controller service is a composition of two other - services: - Call-Blocking and Call-Forwarding. Essentially, it performs the services - of both call-forwarding and call-blocking. - - - - - To learn about how the Call-Blocking service works, refer to - - . - - - - - To learn about how the Call-Forwarding service works, refer to - - . - - - - - Blocked Contacts and Contacts to Forward Cannot Currently Be - Configured - Both the list of blocked contacts and the list of contacts to - forward are currently both hard-coded. However, both of those models - are - evolving toward the eventual use of databases. - - - -
- The Call-Controller Service: Installing, Configuring and - Running - The - Call-Controller service requires the two WAR files for the - Call-Blocking and Call-Forwarding services to be placed in the - correct - directory inside your &PLATFORM_NAME; - SIP Servlets Server binary installation. - However, the Call-Controller service does - not - require their corresponding DAR files: you need only to download and - install a DAR file customized for the Call-Controller service. The - instructions below show you how to do precisely this; there is no - need, - therefore, to first install either the Call-Blocking or the - Call-Forwarding services, though it is helpful to at least be familiar - with them. - - - Pre-Install Requirements and Prerequisites - The following requirements must be met before installation can - begin. - - - - Software Prerequisites - - Either - an &SHORT_PLATFORM_NAME; - for JBoss or an &SHORT_PLATFORM_NAME; - for Tomcat Installation - - - The - Call-Controller Service requires either an &SHORT_PLATFORM_NAME; - for JBoss - or an &SHORT_PLATFORM_NAME; - for Tomcat binary installation. - - You - can find detailed instructions on installing &SHORT_PLATFORM_NAME; - for - JBoss here: - - . - - You - can find detailed instructions on installing &SHORT_PLATFORM_NAME; - for - Tomcat here: - - . - - - - - - Downloading - - The Call-Controller Service is comprised of two WAR files, one - for the Call-Forwarding service and one for Call-Blocking, and a - customized Call-Controller DAR file. You do not need to install the - DAR files for the Call-Forwarding or the Call-Blocking services. - For - more information about WAR files, refer to the - JBoss Application Server Administration and Development Guide - - . For - more information about DAR files, refer to the - JSR 289 spec, Appendix C - - - Download - the Call-Blocking Service's WAR file from here: - - . - - Download - the Call-Forwarding Service's WAR file from here: - - . - - Download - the Call-Controller Service's DAR file from here: - - . - - - Installing - - The - call-blocking-1.4.war - , - call-forwarding-1.4.war - and - call-controller-servlet-dar.properties - archive - files that you downloaded should be placed into different directories - in your SIP Servlet Server installation hierarchy. Which directory - depends on whether you are using the Call-Controller Service with &SHORT_PLATFORM_NAME; - for JBoss or with &SHORT_PLATFORM_NAME; - for Tomcat: - - - - - &SHORT_PLATFORM_NAME; - for JBoss - - - - Place - call-blocking-1.4.war - and - call-forwarding-1.4.war - into the - - JBOSS_HOME - /server/default/deploy/ - - directory, and - call-controller-servlet-dar.properties - into - the - - JBOSS_HOME - /server/default/conf/dars/ - - directory. - - - - - &SHORT_PLATFORM_NAME; - for Tomcat - - - - Place - call-blocking-1.4.war - and - call-forwarding-1.4.war - into the - - CATALINA_HOME - /webapps/ - - directory, and - call-controller-servlet-dar.properties - into - the - - CATALINA_HOME - /conf/dars/ - - directory. - - - - - - Configuring - - The - darConfigurationFileLocation - attribute of - the - Service - element must be set to the value - conf/dars/call-controller-servlet-dar.properties - - . - Instructions for doing so are given below by SIP Servlets Server - type: - - - - - &SHORT_PLATFORM_NAME; - for JBoss - - - - Open the - - JBOSS_HOME - /server/default/deploy/jboss-web.deployer/server.xml - - configuration file and find the - Service - element. Add an attribute to it called - darConfigurationFileLocation - , and set it to - conf/dars/call-controller-servlet-dar.properties - - : - - - Editing &SHORT_PLATFORM_NAME; - for JBoss's server.xml for the Call-Controller Service - - <Service - name="jboss.web" - className="org.mobicents.servlet.sip.startup.SipStandardService" - sipApplicationDispatcherClassName="org.mobicents.servlet.sip.core.SipApplicationDispatcherImpl" - darConfigurationFileLocation="conf/dars/call-controller-servlet-dar.properties " - sipStackPropertiesFile="conf/mss-sip-stack.properties"> - - - Make sure that the configuration file only contains one - darConfigurationFileLocation - attribute: your - new one. - - - - - &SHORT_PLATFORM_NAME; - for Tomcat - - - - Open the - - CATALINA_HOME - /conf/server.xml - - configuration file and find the - Service - element. Add an attribute to it called - darConfigurationFileLocation - , and set it to - conf/dars/call-controller-servlet-dar.properties - - : - - - Editing &SHORT_PLATFORM_NAME; - for Tomcat's server.xml for the Call-Controller Service - - <Service - name="Sip-Servlets" - className="org.mobicents.servlet.sip.startup.SipStandardService" - sipApplicationDispatcherClassName="org.mobicents.servlet.sip.core.SipApplicationDispatcherImpl" - darConfigurationFileLocation="conf/dars/call-controller-servlet-dar.properties " - sipStackPropertiesFile="conf/mss-sip-stack.properties"> - - - Make sure that the configuration file only contains one - darConfigurationFileLocation - attribute: your - new one. - - - - - - Running - - Once the WAR and DAR files have been placed in the right - directories, and the JBoss Application Server or Tomcat Servlet - Container knows where to find them (which you specified in a - server.xml - file), then you should go ahead and - run the SIP Servlets Server. - - - - To learn how to run the SIP Servlets-enabled JBoss Application - Server, refer to - - . - - - To learn how to run the SIP Servlets-enabled Tomcat Container, - refer to - - . - - - Testing - Two use-cases can be distinguished for the Call-Controller - service: one in which a call is blocked, and another in which a - call - is forwarded. Therefore, we have two cases for which we can test the - Call-Controller. - - - - Blocking a Call with Call-Controller - - Start two SIP soft-phones of your choice. Set the account - settings of the SIP soft-phones to: - - - Relevant First Softphone Settings - - - Account name: - forward-receiver - - - - - IP address: - 127.0.0.1 - - - - - Port: - 5090 - - - - - Relevant Second Softphone Settings - - - Account name: - blocked-sender - - - - Neither of the SIP soft-phones needs to be registered. - - - - From the second phone, - blocked-sender - , make - a call to - sip:receiver@sip-servlets.com - . You - should receive a - FORBIDDEN - response. - - - - - Forwarding a Call with Call-Controller - - Start two SIP soft-phones of your choice. Set the account - settings of the SIP soft-phones to: - - - Relevant First Softphone Settings - - - Account name: - forward-receiver - - - - - IP address: - 127.0.0.1 - - - - - Port: - 5090 - - - - - Relevant Second Softphone Settings - - - Account name: - forward-sender - - - - Neither of the SIP soft-phones needs to be registered. - - - - From the second softphone, - forward-sender - , - make a call to - sip:receiver@sip-servlets.com - . The - first phone, - forward-receiver - , should now be - ringing. - - - - - - Stopping - - To learn how to stop the SIP Servlets-enabled JBoss Application - Server, refer to - - . - - - - To learn how to stop the SIP Servlets-enabled Tomcat Container, - refer to - - . - - - Uninstalling - - Unless disk space is at a premium, there is usually no need to - uninstall the Call-Controller Service. However, if you will not be - using it again, you may want to unset or reset the - darConfigurationFileLocation - attribute of the - Service - element, which you set in the - server.xml - configuration file in - - . - - - - You may also wish to delete the WAR and DAR files for the - Call-Controller Service, which you installed in - - . - - - - - provides more information about other service examples - available. - - -
-
- -
diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-scopes_and_contexts.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-scopes_and_contexts.xml deleted file mode 100644 index 1f5c67e08c..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/concept-chapter-scopes_and_contexts.xml +++ /dev/null @@ -1,10 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - Scopes and Contexts of the RestComm - - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/images/RestComm-ekiga-sip-phone-demo.png b/restcomm/restcomm.docs/sources/src/main/resources/en-US/images/RestComm-ekiga-sip-phone-demo.png deleted file mode 100644 index 6b2a4f5174..0000000000 Binary files a/restcomm/restcomm.docs/sources/src/main/resources/en-US/images/RestComm-ekiga-sip-phone-demo.png and /dev/null differ diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/images/RestComm-ekiga-sip-phone-demo_5555.png 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- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - 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a/restcomm/restcomm.docs/sources/src/main/resources/en-US/images/sflphone-config-basic.png and /dev/null differ diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/task-chapter-examples.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/task-chapter-examples.xml deleted file mode 100644 index a163bbbcf1..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/task-chapter-examples.xml +++ /dev/null @@ -1,94 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - Click2Call Example - With the advent of MSS 2.x, the Mobicents team made available - asynchronous processing of requests feature in the world of - SipServlets and demonstrated this feature with a new version of the - traditional Click2Call example. - Now its time to get Click2Call one step further and demonstrate - the CTF features with it. - In this new version of the Click2Call, the CDI programming model - makes the application significant simpler by using beans as services. - Redundant is the use of the ServletContext to store attributes such as - registered users or calls. This is now being served by the - CallStatusContainer and the SipRegistar modules. Several functions of - the application are being delegated to beans also. Methods that need - to fire an event are using the CDI event producer/observer model - instead of a queue that was placed in the ServletContext and the - SimpleSipServlet class is using the SipRegistar module to register - users. - If you check out the Click2Call example you will notice the - following major changes: - - - - SimpleSipServlet is nothing more than a simple class. No more extend - SipServlet and - @Override - doRequest(SipServletRequest req) and such. All the methods of the - class that needs to get notified for a SIP request or response have - @Observe @SipEvent - and that enough to handle SIP messages. - - - - - Everywhere in the project, when there is a need for the SipFactory - we don’t take it using JNDI anymore. A simple - @Inject - SipFactory sipFactory will do the job since CTF makes available - SipFactory for injection. - - - - Several of the application’s functionality have been moved to - stand alone modules. One of them is the SipRegistar module that will - take care of SIP Registration requests. SimpleSipServlet injects - SipRegistar module and delegates registration requests to it. Please - note that these modules could reside in an external archive jar file - placed in the classpath of the application, thus the beginning of - reusable SIP modules. - - - - You will also notice that SipSession or SipApplicationSession in the - SimpleSipServlet wont be retrieved by the request or the response. - If you include the following in the application - @Inject - SipSession sipSession and - @Inject - SipApplicationSession sipAppSession CTF will make sure you will get - the SipSession and SipApplicationSession from the request or the - response accordingly, depending on the SIP event you observe. - - - - - The drawing below will give you the general idea of the - application’s interactions with the serveral modules and the CTF. - -
- Click2Call using CTF - - - - - -
- - - MSS 2.x comes with Click2CallAsync predeployed, make sure you - undeploy this application and apply the correct dar configuration - before you deploy the new version of Click2Call. - - In the examples of the framework you will find a version of - Click2Call with CTF, suitable for servlet 2.5 containers such as - Tomcat 6.0.29 and JBoss AS5. - - diff --git a/restcomm/restcomm.docs/sources/src/main/resources/en-US/task-chapter-getting_started.xml b/restcomm/restcomm.docs/sources/src/main/resources/en-US/task-chapter-getting_started.xml deleted file mode 100644 index f710d0f8b0..0000000000 --- a/restcomm/restcomm.docs/sources/src/main/resources/en-US/task-chapter-getting_started.xml +++ /dev/null @@ -1,284 +0,0 @@ - - -%BOOK_ENTITIES; -]> - - - Getting started with CTF - In order to get started with CTF, the only thing you have to do - is to define few maven dependencies in your project and register the - CTF listener in the sip.xml or web.xml or your application. - First dependency is Weld, the CDI reference implementation by - JBoss and more specific the weld-servlet module. - - - org.jboss.weld.servlet - weld-servlet - 1.1.2.Final - - ]]> - - Second is the framework itself: - - - org.mobicents.servlet.sip.ctf - ctf-core - ]]>&CTF_VERSION; - - ]]> - - - - Make sure you use the JBoss public maven repository: - - - jboss-public-repository-group - JBoss Public Maven Repository Group - https://repository.jboss.org/nexus/content/groups/public - default - - true - never - - - true - never - - - ]]> - - - - - The CTF listener that you have to register in your application - is: - - - - org.mobicents.servlet.sip.ctf.environment.servlet.SipServletsListener - - - ]]> - - - To get the source code along with the examples: - - - checkout project from svn repository: svn co - https://mobicents.googlecode.com/svn/trunk/frameworks/sip-servlets/cdi-telco-framework - - - - In the folder core/ctf-core/ you will find the project - responsible to bootstrap the framework, in order to build it: mvn - clean compile install - - - In the folder core/ctf-core-extension/ you will find the Sip - Servlets extensions to CDI, such as CTF annotations, in order to - build it: mvn clean compile install - - - - In the folder examples/ you will find the examples of the framework, - build according to the notes of every example. The examples contain - the following: - - - click2call-servlet3.0-cdi . The Click2Call application with - asynchronous processing of request feature and CTF. Deploy to MSS - 2.x Tomcat 7.0.8 only - - - click2call-servlet25-cdi. The Click2Call application with - CTF. Deploy to Servlet 2.5 container such as MSS 1.x Tomcat - 6.0.29. The example can be deployed to JBoss AS5 also, build - using : mvn clean compile package -Pjboss5 - - - - SipRegistration. - - - SipRegistarClient - SIP registration example for java - servlets 3.0. Deploy to MSS 2.x Tomcat 7.0.8 only. - - - SipRegistarClient25 - SIP registration example for java - servlets 2.5. Deploy to MSS 1.x Tomcat 6.0.29. The example can - be deployed to JBoss AS5 also, build using : mvn clean compile - package -Pjboss5 - - - - - - - - - - To build issue mvn clean compile package for each project and - deploy accordingly - - - - For JBoss AS5 special maven dependency needed for MSS 1.x JBoss AS5 - version for the JBoss Weld dependency, that will be: - - - org.jboss.weld.servlet - weld-servlet - runtime - jboss5 - 1.1.2.Final - - - org.mobicents.servlet.sip.ctf - ctf-core - ]]>&CTF_VERSION; - jboss5 - - ]]> - - - - - CTF have been tested and verified working in the following MSS - containers: - - - MSS 1.6 JBoss AS5 - - - MSS 1.6 Tomcat 6.0.29 - - - MSS 2.0-Branch Tomcat 7.0.8 - - - - - Summary of CTF dependencies to use - - - MSS 1.x on Tomcat 6.x - - - CDI - - - org.jboss.weld.servlet - weld-servlet - 1.1.2.Final - - ]]> - - - CTF - - - org.mobicents.servlet.sip.ctf - ctf-core - ]]>&CTF_VERSION; - - ]]> - - - - - - - MSS 1.x on JBoss AS5 - - - CDI - - - org.jboss.weld.servlet - weld-servlet - runtime - jboss5 - 1.1.2.Final - - ]]> - - - CTF - - - org.mobicents.servlet.sip.ctf - ctf-core - ]]>&CTF_VERSION; - jboss5 - - ]]> - - - - - - - MSS 2.x on Tomcat 7.x - - - CDI - - - org.jboss.weld.servlet - weld-servlet - 1.1.2.Final - - ]]> - - - CTF - - - org.mobicents.servlet.sip.ctf - ctf-core - ]]>&CTF_VERSION; - - ]]> - - - - - - -