Remove 'On This Page' from a lot of pages

Author: seanbright

docs/Configuration/Applications/Asterisk-Queues/Building-Queues.md

@@ -24,7 +24,6 @@ The first thing we want to do is register a couple of SIP devices to our server.
 
 In sip.conf, we add the following to the bottom of our file:
 
-On This Page
 
 ```
 sip.conf

docs/Configuration/Applications/Conferencing-Applications/ConfBridge/ConfBridge-AMI-Actions.md

@@ -32,7 +32,7 @@ ListItems: 1
 
 ```
 
-On This PageConfbridgeListRooms
+ConfbridgeListRooms
 -------------------
 
 Lists data about all active conferences. ConfbridgeListRooms will follow as separate events, followed by a final event called ConfbridgeListRoomsComplete.

docs/Configuration/Applications/Conferencing-Applications/ConfBridge/ConfBridge-AMI-Events.md

@@ -17,7 +17,7 @@ Conference: 1111
 
 ```
 
-On This PageConfbridgeJoin
+ConfbridgeJoin
 --------------
 
 This event is sent when a user joins a conference - either one already in progress or as the first user to join a newly instantiated bridge.

docs/Configuration/Applications/Conferencing-Applications/ConfBridge/ConfBridge-Functions.md

@@ -8,7 +8,7 @@ Function CONFBRIDGE
 
 The CONFBRIDGE dialplan function is used to set customized Bridge and/or User Profiles on a channel for the ConfBridge application. It uses the same options defined in confbridge.conf and allows the creation of dynamic, dialplan-driven conferences.
 
-On This PageSyntax
+Syntax
 ------
 
 ```

docs/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Asterisk-PJSIP-Troubleshooting-Guide.md

@@ -29,7 +29,6 @@ Unrecognized Endpoint
 
 All inbound SIP traffic to Asterisk must be matched to a configured endpoint. If Asterisk is unable to determine which endpoint the SIP request is coming from, then the incoming request will be rejected. If you are seeing messages like:
 
-On this Page
 
 ```
 [2014-10-13 16:12:17.349] DEBUG[27284]: res_pjsip_endpoint_identifier_user.c:106 username_identify: Could not identify endpoint by username 'eggowaffles'

docs/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Configuring-Outbound-Registrations.md

@@ -18,7 +18,6 @@ Configuration options
 
 A list of outbound registration configuration options can be found on [this page](/Latest_API/API_Documentation/Module_Configuration/res_pjsip_outbound_registration). Here is a simple example configuration for an outbound registration to a provider:
 
-On this Page
 
 pjsip.conf
 

docs/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Configuring-res_pjsip-for-Presence-Subscriptions.md

@@ -12,7 +12,6 @@ Capabilities
 ============
 
 Asterisk's PJSIP channel driver provides the same presence subscription capabilities as `chan_sip` does. This means that [RFC 3856](http://tools.ietf.org/html/rfc3856) presence and [RFC 4235](http://www.rfc-editor.org/rfc/rfc4235.txt) dialog info are supported. Presence subscriptions support [RFC 3863](http://tools.ietf.org/html/rfc3863) PIDF+XML bodies as well as [XPIDF+XML](http://tools.ietf.org/html/draft-rosenberg-impp-pidf-00). Beyond that, Asterisk also supports subscribing to [RFC 4662](http://tools.ietf.org/html/rfc4662) lists of presence resources.  
-On this Page
 
 extensions.conf
 

docs/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Migrating-from-chan_sip-to-res_pjsip.md

@@ -40,7 +40,6 @@ Writing pjsip.conf
 
 ```
 
-On this Page
 
 Side by Side Examples of sip.conf and pjsip.conf Configuration
 ==============================================================

docs/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Resource-List-Subscriptions-RLS.md

@@ -12,7 +12,6 @@ In a PBX environment, it is common for SIP devices to subscribe to many resource
 
 * In a situation where all phones boot up simultaneously, each of the phones will be sending out their SIP SUBSCRIBEs nearly simultaneously, placing a larger-than-average burden on the Asterisk server's CPU.
 * In the SIP stack, N-squared long-term SIP dialogs have to be maintained, tying up more system resources (e.g. memory).
-On this Page
 
 These limitations can drastically limit the number of devices a PBX administrator can use with an Asterisk system. Even if the hardware is capable of handling the mean traffic of, say, 200 users, it may be required to limit the number of users to 50 or fewer because of the N-squared subscriptions generating so much simultaneous traffic.
 

docs/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/res_pjsip-Configuration-Examples.md

@@ -45,7 +45,6 @@ max_contacts=1
 * auth= is used for the endpoint as opposed to outbound_auth= since we want to allow inbound registration for this endpoint
 * max_contacts= is set to something non-zero as we want to allow contacts to be created through registration
 
-On this Page
 
 A SIP trunk to your service provider, including outbound registration
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