updates, content added and images, formatting. (#121)

* Update streams.rst

* Update version_upgrade.rst

* Update password_reset.rst

* Update nat.rst

* Update bridges.rst

* Update bridges.rst

* Update call_recordings.rst

* image add

* image add

* Update conference_profiles.rst

* Update conference_controls.rst

* Update additional_information.rst

* Create regex.rst

source/additional_information.rst

@@ -22,3 +22,4 @@ Additional Information
   additional_information/nat.rst
   applications/provision.rst
   advanced/version_upgrade.rst
+  additional_information/regex.rst

source/additional_information/nat.rst

@@ -38,5 +38,13 @@ FusionPBX is behind NAT and you don't have a static ip address. You do have a fi
 * Set external_sip_ip to auto-nat
 * Restart FreeSWITCH.   service freeswitch restart
 
+Symptoms of misconfigured NAT
+^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
+
+* Call drops after 30ish seconds.
+* One way audio
+* No audio
+
+
 
 .. _SIP ALG: http://docs.fusionpbx.com/en/latest/firewall/firewall_devices.html#sip-alg

source/additional_information/password_reset.rst

@@ -18,7 +18,12 @@ The current method to changing the superadmin password is actually to make a new
  mv config.php config1.php
 
 |
-2. Go to the FusionPBX install login page.  This will put FusionPBX into a recovery mode.  Choose the language for your region and **click next.**
+2. Go to the FusionPBX install login page in the web browser.  This will put FusionPBX into a recovery mode.  Choose the language for your region and **click next.**
+
+.. note::
+
+ You will type in your web browser either the ip hxxps://xxx.xxx.xxx.xxx or  the domain name hxxps://sub.domain.tld .
+ 
 
 .. image:: ../_static/images/install_lang_new.jpg
         :scale: 85%

source/additional_information/regex.rst

@@ -0,0 +1,20 @@
+####################
+Regular Expressions
+####################
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

source/advanced/version_upgrade.rst

@@ -9,7 +9,7 @@ Version Upgrade can take several steps to perform. Below will show how to upgrad
 Version 4.2 to 4.4
 ^^^^^^^^^^^^^^^^^^
 
-1.  Switch branches
+1. Switch branches
 
 ::
 
@@ -29,7 +29,11 @@ Version 4.2 to 4.4
  
 - Then go to Advanced -> Upgrade and update the Source Code, Schema, Menu Defaults and Permission Defaults.
 
-2.  Update the following Dialplans.
+.. note::
+
+ config.lua needs to be read and write by the webserver in order for advanced > default settings to update config.lua with new path information. Make sure config.lua and config.php are in /etc/fuionpbx/ . Don't miss this step chown -R www-data:www-data /etc/fusionpbx/ 
+
+2. Update the following Dialplans.
 
 ::
 
@@ -40,9 +44,9 @@ Version 4.2 to 4.4
 
 - Update these Dialplans by first selecting and deleting their entries from within the Dialplan Manager for all domains. Then, run Advanced -> Upgrade -> App Defaults to retrieve the new versions of the diaplans.
 
-3.  In the menu go to Status then SIP Status and press 'Flush Cache'.
+3. In the menu go to Status then SIP Status and press 'Flush Cache'.
 
-4.  Update old recordings set the record_name and record_path.
+4. Update old recordings set the record_name and record_path.
 
 ::
 
@@ -50,7 +54,19 @@ Version 4.2 to 4.4
  wget https://raw.githubusercontent.com/fusionpbx/fusionpbx-scripts/master/upgrade/record_path.php
  php record_path.php
  
-5.  Resave all Call Center Queues to update each call center queue dialplan. Then restart mod call center or FreeSWITCH.
+5. Resave all Call Center Queues to update each call center queue dialplan. Then restart mod call center or FreeSWITCH.
+
+6. Advanced > Default Settings
+
+The email section in Advanced > Default settings, changes have been made.
+
+*  You will find duplicates with a blank value.  The duplicates must be updated with the existing info from the originals. These duplicates are the new and correct settings.  You'll have to update these blank ones with the existing values (like smtp server info) to the new default ones.  Then delete the original ones.
+
+*  Don't delete the blank entries.  The code behind them are for version 4.4+ and the original ones are not.
+
+.. note::
+
+ If you already deleted the blank ones, you'll have to delete the email section then run Advanced > Upgrade > App Defaults check box.  Then go back to Advanced > Default settings and set the email section back up.
 
 
 Version 4.0 to 4.2
@@ -133,22 +149,22 @@ https://domain.com/core/upgrade/index.php
 
 12. Update Time Conditions (Bug Fix)
  
- ::
+::
  
  Goto Advanced > Upgrades page.  Check box Update Source, execute. 
  Goto Advanced > Default settings > Category > delete the category: time condition presets.
  Goto Advanced > Upgrade >  check box App Defaults, execute.
  Goto Advanced > Default settings. Click "Reload" at the top right. (This will get the new presets)
 
- ::
- 
- Next steps are for existing Time Conditions
+Next steps are for existing Time Conditions
+
+::
+
  Goto Apps > Time Conditions and edit the time conditions remove all holidays and hit save.
  Select the holidays over again.
 
 
- .. note::
-
+.. note::
 
   Many of the provisioning templates were updated.  If you use custom provisioning templates you should consider updating them with the new versions. 
 

source/applications/bridges.rst

@@ -13,3 +13,68 @@ Bridge statements are used to send calls directly to other destinations like ano
 * Click the edit icon on the right to edit a bridge
 * Click the X to delete a bridge
 
+Bridge Examples
+^^^^^^^^^^^^^^^^^
+
+
+Bridges are how ring groups are made.  The code in FusionPBX simplifys that for you.  You can however manually do what ring groups do and with bridges.
+
+**Bridge Statement advanced options**
+
+* For multiple destinations. Multiple destinations are allowed as long as you use a  ,  |  or  :_:
+::
+
+ Comma , Means simultaneous
+ Pipe |  Means In a sequence
+ colon under score colon  :_: Means Enterprise
+
+**Loopback to an external number**
+
+::
+
+ loopback/12085551234
+ 
+**Loopback to multiple external numbers simultaneously**
+
+::
+
+ loopback/12085551234,loopback/12085552222,loopback/12085553333
+ 
+**To another sip server, sip gateway, or another carrier**
+
+::
+
+ sofia/internal/[email protected]:5060
+
+**To a user**
+
+::
+
+ user/1001
+
+**Using LCR**
+
+::
+
+ lcr/12085551234
+ 
+**Using variables**
+
+::
+
+ {abc=123}sofia/internal/[email protected]:5060
+ 
+**Using variables in sequence with a sip server**
+
+::
+
+ {abc=123}sofia/internal/[email protected]:5060|sofia/internal/[email protected]:5060
+ 
+
+**Using variables in sequence with a sip server**
+
+::
+
+ [server=d1]sofia/internal/[email protected]:5060|[server=d2]sofia/internal/[email protected]:5060 
+
+

source/applications/call_recordings.rst

@@ -11,3 +11,4 @@ Shows the call recordings with name, length, date and time, and call direction.
 
 * Click the eye icon on the right to view more details
 * Click the X to delete a recording
+* Click multiple check boxes to delete multiple at once.

source/applications/conference_controls.rst

@@ -13,3 +13,13 @@ Call controls enable ability to assign digits to actions. They can be used to mu
 
 * Click the edit icon on the right to adjust the control
 * Click the plus to create a new control set
+
+
+
+default Conference Control
+^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
+
+.. image:: ../_static/images/applications/fusionpbx_conference_controls1.jpg
+        :scale: 85%
+
+

source/applications/conference_profiles.rst

@@ -13,3 +13,37 @@ A group of conference parameters saved together as a profile.
 
 * Click the edit icon on the right to adjust the profile
 * Click the plus to create a new profile
+
+
+Default Profile
+^^^^^^^^^^^^^^^^
+
+.. image:: ../_static/images/applications/conference_profiles/fusionpbx_conference_profiles.jpg
+        :scale: 85%
+
+
+
+* cdr-log-dir: Set as auto.  Could be set manually and is enabled. 
+* domain: enabled.
+* rate: The rate in kHz. 8000kHz and is enabled. 
+* interval: 20 is the default.
+* energy-level: 15 is the default.
+* auto-gain-level: 0 is the default.
+* caller-controls: default is the default.
+* moderator-controls: moderator is the default.
+* muted-sound: conference/conf-muted.wav is the default.
+* unmuted-sound: conference/conf-unmuted.wav is the default.
+* alone-sound: conference/conf-alone.wav is the default.
+* moh-sound: local_stream://default is the default.
+* enter-sound: tone_stream://%(200,0,500,600,700) is the default.
+* exit-sound: tone_stream://%(500,0,300,200,100,50,25) is the default.
+* kicked-sound: conference/conf-kicked.wav is the default.
+* locked-sound: conference/conf-locked.wav is the default.
+* is-locked-sound: conference/conf-is-locked.wav is the default.
+* is-unlocked-sound: conference/conf-is-unlocked.wav is the default.
+* pin-sound: conference/conf-pin.wav is the default.
+* bad-pin-sound: conference/conf-bad-pin.wav is the default.
+* caller-id-name:
+* caller-id-number:
+* comfort-noise: true is the default.
+

source/applications/streams.rst

@@ -23,5 +23,10 @@ Define details for streaming audio.
 
 .. Note::
 
-  Please be aware of your countries copyright laws for streaming the content you are going to stream
+ Editing a stream path will result in having to update anything that is using the stream.  For example, if you have extension 500 using stream "Local Weather" and you edit the shout:// path then you will have to go back to extension 500 and reset the music on hold for extension 500.  This is by design.
+
+
+.. Warning::
+
+  Please be aware of your countries copyright laws for streaming the content you are going to stream.